diff options
| author | Linus Torvalds <torvalds@linux-foundation.org> | 2018-08-15 00:10:30 +0300 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2018-08-15 00:10:30 +0300 |
| commit | 747f62305dfb8a592835c7401069bfdbc06acbae (patch) | |
| tree | 5123b38238c489be1407202b138cdbbb31198f51 | |
| parent | 2c20443ec221dcb76484b30933593e8ecd836bbd (diff) | |
| parent | f5b6c1fcb42fe7d6f2f6eb2220512e2a5f875133 (diff) | |
| download | linux-747f62305dfb8a592835c7401069bfdbc06acbae.tar.xz | |
Merge tag 'sound-4.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"It's been busy summer weeks and hence lots of changes, partly for a
few new drivers and partly for a wide range of fixes.
Here are highlights:
ALSA Core:
- Fix rawmidi buffer management, code cleanup / refactoring
- Fix the SG-buffer page handling with incorrect fallback size
- Fix the stall at virmidi trigger callback with a large buffer; also
offloading and code-refactoring along with it
- Various ALSA sequencer code cleanups
ASoC:
- Deploy the standard snd_pcm_stop_xrun() helper in several drivers
- Support for providing name prefixes to generic component nodes
- Quite a few fixes for DPCM as it gains a bit wider use and more
robust testing
- Generalization of the DIO2125 support to a simple amplifier driver
- Accessory detection support for the audio graph card
- DT support for PXA AC'97 devices
- Quirks for a number of new x86 systems
- Support for AM Logic Meson, Everest ES7154, Intel systems with
RT5682, Qualcomm QDSP6 and WCD9335, Realtek RT5682 and TI TAS5707
HD-audio:
- Code refactoring in HD-audio ext codec codes to drop own classes;
preliminary works for the upcoming legacy codec support
- Generalized DRM audio component for the upcoming radeon / amdgpu
support
- Unification of mic mute-LED and GPIO support for various codecs
- Further improvement of CA0132 codec support including Recon3D
- Proper vga_switcheroo handling for AMD i-GPU
- Update of model list in documentation
- Fixups for another HP Spectre x360, Conexant codecs, power-save
blacklist update
USB-audio:
- Fix the invalid sample rate setup with external clock
- Support of UAC3 selector units and processing units
- Basic UAC3 power-domain support
- Support for Encore mDSD and Thesycon-based DSD devices
- Preparation for future complete callback changes
Firewire:
- Add support for MOTU Traveler
Misc:
- The endianess notation fixes in various drivers
- Add fall-through comment in lots of drivers
- Various sparse warning fixes, e.g. about PCM format types"
* tag 'sound-4.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (529 commits)
ASoC: adav80x: mark expected switch fall-through
ASoC: da7219: Add delays to capture path to remove DC offset noise
ALSA: usb-audio: Mark expected switch fall-through
ALSA: mixart: Mark expected switch fall-through
ALSA: opl3: Mark expected switch fall-through
ALSA: hda/ca0132 - Add exit commands for Recon3D
ALSA: hda/ca0132 - Change mixer controls for Recon3D
ALSA: hda/ca0132 - Add Recon3D input and output select commands
ALSA: hda/ca0132 - Add DSP setup defaults for Recon3D
ALSA: hda/ca0132 - Add Recon3D startup functions and setup
ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio
ALSA: hda/ca0132 - Add Recon3D pincfg
ALSA: hda/ca0132 - Add quirk ID and enum for Recon3D
ALSA: hda/ca0132 - Add alt_functions unsolicited response
ALSA: hda/ca0132 - Clean up ca0132_init function.
ALSA: hda/ca0132 - Create mmio gpio function to make code clearer
ASoC: wm_adsp: Make DSP name configurable by codec driver
ASoC: wm_adsp: Declare firmware controls from codec driver
ASoC: max98373: Added software reset register to readable registers
ASoC: wm_adsp: Correct DSP pointer for preloader control
...
498 files changed, 19417 insertions, 6533 deletions
diff --git a/Documentation/devicetree/bindings/sound/ac97-bus.txt b/Documentation/devicetree/bindings/sound/ac97-bus.txt new file mode 100644 index 000000000000..103c428f2595 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ac97-bus.txt @@ -0,0 +1,32 @@ +Generic AC97 Device Properties + +This documents describes the devicetree bindings for an ac97 controller child +node describing ac97 codecs. + +Required properties: +-compatible : Must be "ac97,vendor_id1,vendor_id2 + The ids shall be the 4 characters hexadecimal encoding, such as + given by "%04x" formatting of printf +-reg : Must be the ac97 codec number, between 0 and 3 + +Example: +ac97: sound@40500000 { + compatible = "marvell,pxa270-ac97"; + reg = < 0x40500000 0x1000 >; + interrupts = <14>; + reset-gpios = <&gpio 95 GPIO_ACTIVE_HIGH>; + #sound-dai-cells = <1>; + pinctrl-names = "default"; + pinctrl-0 = < &pinctrl_ac97_default >; + clocks = <&clks CLK_AC97>, <&clks CLK_AC97CONF>; + clock-names = "AC97CLK", "AC97CONFCLK"; + + #address-cells = <1>; + #size-cells = <0>; + audio-codec@0 { + reg = <0>; + compatible = "ac97,574d,4c13"; + clocks = <&fixed_wm9713_clock>; + clock-names = "ac97_clk"; + } +}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt new file mode 100644 index 000000000000..3dfc2515e5c6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt @@ -0,0 +1,23 @@ +* Amlogic Audio FIFO controllers + +Required properties: +- compatible: 'amlogic,axg-toddr' or + 'amlogic,axg-frddr' +- reg: physical base address of the controller and length of memory + mapped region. +- interrupts: interrupt specifier for the fifo. +- clocks: phandle to the fifo peripheral clock provided by the audio + clock controller. +- resets: phandle to memory ARB line provided by the arb reset controller. +- #sound-dai-cells: must be 0. + +Example of FRDDR A on the A113 SoC: + +frddr_a: audio-controller@1c0 { + compatible = "amlogic,axg-frddr"; + reg = <0x0 0x1c0 0x0 0x1c>; + #sound-dai-cells = <0>; + interrupts = <GIC_SPI 88 IRQ_TYPE_EDGE_RISING>; + clocks = <&clkc_audio AUD_CLKID_FRDDR_A>; + resets = <&arb AXG_ARB_FRDDR_A>; +}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt new file mode 100644 index 000000000000..80b411296480 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt @@ -0,0 +1,124 @@ +Amlogic AXG sound card: + +Required properties: + +- compatible: "amlogic,axg-sound-card" +- model : User specified audio sound card name, one string + +Optional properties: + +- audio-aux-devs : List of phandles pointing to auxiliary devices +- audio-widgets : Please refer to widgets.txt. +- audio-routing : A list of the connections between audio components. + +Subnodes: + +- dai-link: Container for dai-link level properties and the CODEC + sub-nodes. There should be at least one (and probably more) + subnode of this type. + +Required dai-link properties: + +- sound-dai: phandle and port of the CPU DAI. + +Required TDM Backend dai-link properties: +- dai-format : CPU/CODEC common audio format + +Optional TDM Backend dai-link properties: +- dai-tdm-slot-rx-mask-{0,1,2,3}: Receive direction slot masks +- dai-tdm-slot-tx-mask-{0,1,2,3}: Transmit direction slot masks + When omitted, mask is assumed to have to no + slots. A valid must have at one slot, so at + least one these mask should be provided with + an enabled slot. +- dai-tdm-slot-num : Please refer to tdm-slot.txt. + If omitted, slot number is set to accommodate the largest + mask provided. +- dai-tdm-slot-width : Please refer to tdm-slot.txt. default to 32 if omitted. +- mclk-fs : Multiplication factor between stream rate and mclk + +Backend dai-link subnodes: + +- codec: dai-link representing backend links should have at least one subnode. + One subnode for each codec of the dai-link. + dai-link representing frontend links have no codec, therefore have no + subnodes + +Required codec subnodes properties: + +- sound-dai: phandle and port of the CODEC DAI. + +Optional codec subnodes properties: + +- dai-tdm-slot-tx-mask : Please refer to tdm-slot.txt. +- dai-tdm-slot-rx-mask : Please refer to tdm-slot.txt. + +Example: + +sound { + compatible = "amlogic,axg-sound-card"; + model = "AXG-S420"; + audio-aux-devs = <&tdmin_a>, <&tdmout_c>; + audio-widgets = "Line", "Lineout", + "Line", "Linein", + "Speaker", "Speaker1 Left", + "Speaker", "Speaker1 Right"; + "Speaker", "Speaker2 Left", + "Speaker", "Speaker2 Right"; + audio-routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2", + "SPDIFOUT IN 0", "FRDDR_A OUT 3", + "TDM_C Playback", "TDMOUT_C OUT", + "TDMIN_A IN 2", "TDM_C Capture", + "TDMIN_A IN 5", "TDM_C Loopback", + "TODDR_A IN 0", "TDMIN_A OUT", + "Lineout", "Lineout AOUTL", + "Lineout", "Lineout AOUTR", + "Speaker1 Left", "SPK1 OUT_A", + "Speaker2 Left", "SPK2 OUT_A", + "Speaker1 Right", "SPK1 OUT_B", + "Speaker2 Right", "SPK2 OUT_B", + "Linein AINL", "Linein", + "Linein AINR", "Linein"; + + dai-link@0 { + sound-dai = <&frddr_a>; + }; + + dai-link@1 { + sound-dai = <&toddr_a>; + }; + + dai-link@2 { + sound-dai = <&tdmif_c>; + dai-format = "i2s"; + dai-tdm-slot-tx-mask-2 = <1 1>; + dai-tdm-slot-tx-mask-3 = <1 1>; + dai-tdm-slot-rx-mask-1 = <1 1>; + mclk-fs = <256>; + + codec@0 { + sound-dai = <&lineout>; + }; + + codec@1 { + sound-dai = <&speaker_amp1>; + }; + + codec@2 { + sound-dai = <&speaker_amp2>; + }; + + codec@3 { + sound-dai = <&linein>; + }; + + }; + + dai-link@3 { + sound-dai = <&spdifout>; + + codec { + sound-dai = <&spdif_dit>; + }; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt new file mode 100644 index 000000000000..521c38ad89e7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt @@ -0,0 +1,20 @@ +* Amlogic Audio SPDIF Output + +Required properties: +- compatible: 'amlogic,axg-spdifout' +- clocks: list of clock phandle, one for each entry clock-names. +- clock-names: should contain the following: + * "pclk" : peripheral clock. + * "mclk" : master clock +- #sound-dai-cells: must be 0. + +Example on the A113 SoC: + +spdifout: audio-controller@480 { + compatible = "amlogic,axg-spdifout"; + reg = <0x0 0x480 0x0 0x50>; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_SPDIFOUT>, + <&clkc_audio AUD_CLKID_SPDIFOUT_CLK>; + clock-names = "pclk", "mclk"; +}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt new file mode 100644 index 000000000000..1c1b7490554e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt @@ -0,0 +1,28 @@ +* Amlogic Audio TDM formatters + +Required properties: +- compatible: 'amlogic,axg-tdmin' or + 'amlogic,axg-tdmout' +- reg: physical base address of the controller and length of memory + mapped region. +- clocks: list of clock phandle, one for each entry clock-names. +- clock-names: should contain the following: + * "pclk" : peripheral clock. + * "sclk" : bit clock. + * "sclk_sel" : bit clock input multiplexer. + * "lrclk" : sample clock + * "lrclk_sel": sample clock input multiplexer + +Example of TDMOUT_A on the A113 SoC: + +tdmout_a: audio-controller@500 { + compatible = "amlogic,axg-tdmout"; + reg = <0x0 0x500 0x0 0x40>; + clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>, + <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>, + <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>, + <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>, + <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>; + clock-names = "pclk", "sclk", "sclk_sel", + "lrclk", "lrclk_sel"; +}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt new file mode 100644 index 000000000000..cabfb26a5f22 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt @@ -0,0 +1,22 @@ +* Amlogic Audio TDM Interfaces + +Required properties: +- compatible: 'amlogic,axg-tdm-iface' +- clocks: list of clock phandle, one for each entry clock-names. +- clock-names: should contain the following: + * "sclk" : bit clock. + * "lrclk": sample clock + * "mclk" : master clock + -> optional if the interface is in clock slave mode. +- #sound-dai-cells: must be 0. + +Example of TDM_A on the A113 SoC: + +tdmif_a: audio-controller@0 { + compatible = "amlogic,axg-tdm-iface"; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_MST_A_MCLK>, + <&clkc_audio AUD_CLKID_MST_A_SCLK>, + <&clkc_audio AUD_CLKID_MST_A_LRCLK>; + clock-names = "mclk", "sclk", "lrclk"; +}; diff --git a/Documentation/devicetree/bindings/sound/atmel-i2s.txt b/Documentation/devicetree/bindings/sound/atmel-i2s.txt index 735368b8a73f..40549f496a81 100644 --- a/Documentation/devicetree/bindings/sound/atmel-i2s.txt +++ b/Documentation/devicetree/bindings/sound/atmel-i2s.txt @@ -15,7 +15,6 @@ Required properties: - clock-names: Should be one of each entry matching the clocks phandles list: - "pclk" (peripheral clock) Required. - "gclk" (generated clock) Optional (1). - - "aclk" (Audio PLL clock) Optional (1). - "muxclk" (I2S mux clock) Optional (1). Optional properties: @@ -23,9 +22,9 @@ Optional properties: - princtrl-names: Should contain only one value - "default". -(1) : Only the peripheral clock is required. The generated clock, the Audio - PLL clock adn the I2S mux clock are optional and should only be set - together, when Master Mode is required. +(1) : Only the peripheral clock is required. The generated clock and the I2S + mux clock are optional and should only be set together, when Master Mode + is required. Example: @@ -40,8 +39,8 @@ Example: (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | AT91_XDMAC_DT_PERID(32))>; dma-names = "tx", "rx"; - clocks = <&i2s0_clk>, <&i2s0_gclk>, <&audio_pll_pmc>, <&i2s0muxck>; - clock-names = "pclk", "gclk", "aclk", "muxclk"; + clocks = <&i2s0_clk>, <&i2s0_gclk>, <&i2s0muxck>; + clock-names = "pclk", "gclk", "muxclk"; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2s0_default>; }; diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-card.txt index d04ea3b1a1dd..7e63e53a901c 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-card.txt +++ b/Documentation/devicetree/bindings/sound/audio-graph-card.txt @@ -18,6 +18,8 @@ Below are same as Simple-Card. - bitclock-inversion - frame-inversion - mclk-fs +- hp-det-gpio +- mic-det-gpio - dai-tdm-slot-num - dai-tdm-slot-width - clocks / system-clock-frequency diff --git a/Documentation/devicetree/bindings/sound/dioo,dio2125.txt b/Documentation/devicetree/bindings/sound/dioo,dio2125.txt deleted file mode 100644 index 63dbfe0f11d0..000000000000 --- a/Documentation/devicetree/bindings/sound/dioo,dio2125.txt +++ /dev/null @@ -1,12 +0,0 @@ -DIO2125 Audio Driver - -Required properties: -- compatible : "dioo,dio2125" -- enable-gpios : the gpio connected to the enable pin of the dio2125 - -Example: - -amp: analog-amplifier { - compatible = "dioo,dio2125"; - enable-gpios = <&gpio GPIOH_3 0>; -}; diff --git a/Documentation/devicetree/bindings/sound/everest,es7134.txt b/Documentation/devicetree/bindings/sound/everest,es7134.txt index 5495a3cb8b7b..091666069bde 100644 --- a/Documentation/devicetree/bindings/sound/everest,es7134.txt +++ b/Documentation/devicetree/bindings/sound/everest,es7134.txt @@ -1,10 +1,15 @@ ES7134 i2s DA converter Required properties: -- compatible : "everest,es7134" or "everest,es7144" +- compatible : "everest,es7134" or + "everest,es7144" or + "everest,es7154" +- VDD-supply : regulator phandle for the VDD supply +- PVDD-supply: regulator phandle for the PVDD supply for the es7154 Example: i2s_codec: external-codec { compatible = "everest,es7134"; + VDD-supply = <&vcc_5v>; }; diff --git a/Documentation/devicetree/bindings/sound/everest,es7241.txt b/Documentation/devicetree/bindings/sound/everest,es7241.txt new file mode 100644 index 000000000000..28f82cf4959f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es7241.txt @@ -0,0 +1,28 @@ +ES7241 i2s AD converter + +Required properties: +- compatible : "everest,es7241" +- VDDP-supply: regulator phandle for the VDDA supply +- VDDA-supply: regulator phandle for the VDDP supply +- VDDD-supply: regulator phandle for the VDDD supply + +Optional properties: +- reset-gpios: gpio connected to the reset pin +- m0-gpios : gpio connected to the m0 pin +- m1-gpios : gpio connected to the m1 pin +- everest,sdout-pull-down: + Format used by the serial interface is controlled by pulling + the sdout. If the sdout is pulled down, leftj format is used. + If this property is not provided, sdout is assumed to pulled + up and i2s format is used + +Example: + +linein: audio-codec@2 { + #sound-dai-cells = <0>; + compatible = "everest,es7241"; + VDDA-supply = <&vcc_3v3>; + VDDP-supply = <&vcc_3v3>; + VDDD-supply = <&vcc_3v3>; + reset-gpios = <&gpio GPIOH_42>; +}; diff --git a/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt b/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt new file mode 100644 index 000000000000..2ea85d5be6a4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt @@ -0,0 +1,27 @@ +Marvell PXA2xx audio complex + +This descriptions matches the AC97 controller found in pxa2xx and pxa3xx series. + +Required properties: + - compatible: should be one of the following: + "marvell,pxa250-ac97" + "marvell,pxa270-ac97" + "marvell,pxa300-ac97" + - reg: device MMIO address space + - interrupts: single interrupt generated by AC97 IP + - clocks: input clock of the AC97 IP, refer to clock-bindings.txt + +Optional properties: + - pinctrl-names, pinctrl-0: refer to pinctrl-bindings.txt + - reset-gpios: gpio used for AC97 reset, refer to gpio.txt + +Example: + ac97: sound@40500000 { + compatible = "marvell,pxa250-ac97"; + reg = < 0x40500000 0x1000 >; + interrupts = <14>; + reset-gpios = <&gpio 113 GPIO_ACTIVE_HIGH>; + #sound-dai-cells = <1>; + pinctrl-names = "default"; + pinctrl-0 = < &pmux_ac97_default >; + }; diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt index 74c9ba6c2823..93b982e9419f 100644 --- a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt +++ b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt @@ -5,6 +5,14 @@ Required properties: compatible Must be "mrvl,pxa-ssp-dai" port A phandle reference to a PXA ssp upstream device +Optional properties: + + clock-names + clocks Through "clock-names" and "clocks", external clocks + can be configured. If a clock names "extclk" exists, + it will be set to the mclk rate of the audio stream + and be used as clock provider of the DAI. + Example: /* upstream device */ diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt deleted file mode 100644 index 551fbb8348c2..000000000000 --- a/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt +++ /dev/null @@ -1,15 +0,0 @@ -DT bindings for ARM PXA2xx PCM platform driver - -This is just a dummy driver that registers the PXA ASoC platform driver. -It does not have any resources assigned. - -Required properties: - - - compatible 'mrvl,pxa-pcm-audio' - -Example: - - pxa_pcm_audio: snd_soc_pxa_audio { - compatible = "mrvl,pxa-pcm-audio"; - }; - diff --git a/Documentation/devicetree/bindings/sound/name-prefix.txt b/Documentation/devicetree/bindings/sound/name-prefix.txt new file mode 100644 index 000000000000..645775908657 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/name-prefix.txt @@ -0,0 +1,24 @@ +Name prefix: + +Card implementing the routing property define the connection between +audio components as list of string pair. Component using the same +sink/source names may use the name prefix property to prepend the +name of their sinks/sources with the provided string. + +Optional name prefix property: +- sound-name-prefix : string using as prefix for the sink/source names of + the component. + +Example: Two instances of the same component. + +amp0: analog-amplifier@0 { + compatible = "simple-audio-amplifier"; + enable-gpios = <&gpio GPIOH_3 0>; + sound-name-prefix = "FRONT"; +}; + +amp1: analog-amplifier@1 { + compatible = "simple-audio-amplifier"; + enable-gpios = <&gpio GPIOH_4 0>; + sound-name-prefix = "BACK"; +}; diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8096.txt b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt index c7600a93ab39..c814e867850f 100644 --- a/Documentation/devicetree/bindings/sound/qcom,apq8096.txt +++ b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt @@ -7,7 +7,7 @@ This binding describes the APQ8096 sound card, which uses qdsp for audio. Value type: <stringlist> Definition: must be "qcom,apq8096-sndcard" -- qcom,audio-routing: +- audio-routing: Usage: Optional Value type: <stringlist> Definition: A list of the connections between audio components. @@ -49,6 +49,12 @@ This binding describes the APQ8096 sound card, which uses qdsp for audio. "DMIC1" "DMIC2" "DMIC3" + +- model: + Usage: required + Value type: <stringlist> + Definition: The user-visible name of this sound card. + = dailinks Each subnode of sndcard represents either a dailink, and subnodes of each dailinks would be cpu/codec/platform dais. @@ -79,11 +85,16 @@ dailinks would be cpu/codec/platform dais. Value type: <phandle with arguments> Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node. +Obsolete: + qcom,model: String for soundcard name (Use model instead) + qcom,audio-routing: A list of the connections between audio components. + (Use audio-routing instead) + Example: audio { compatible = "qcom,apq8096-sndcard"; - qcom,model = "DB820c"; + model = "DB820c"; mm1-dai-link { link-name = "MultiMedia1"; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt index cb709e5dbc44..bbae426cdfb1 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt @@ -18,6 +18,11 @@ used by the apr service device. = ADM routing "routing" subnode of the ADM node represents adm routing specific configuration +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,q6adm-routing". + - #sound-dai-cells Usage: required Value type: <u32> @@ -28,6 +33,7 @@ q6adm@8 { compatible = "qcom,q6adm"; reg = <APR_SVC_ADM>; q6routing: routing { + compatible = "qcom,q6adm-routing"; #sound-dai-cells = <0>; }; }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt index bdbf87df8c0b..a8179409c194 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt @@ -17,6 +17,11 @@ used by all apr services. Must contain the following properties. subnode of "dais" representing board specific dai setup. "dais" node should have following properties followed by dai children. +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,q6afe-dais" + - #sound-dai-cells Usage: required Value type: <u32> @@ -100,6 +105,7 @@ q6afe@4 { reg = <APR_SVC_AFE>; dais { + compatible = "qcom,q6afe-dais"; #sound-dai-cells = <1>; #address-cells = <1>; #size-cells = <0>; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt index 2178eb91146f..f9c7bd8c1bc0 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt @@ -17,6 +17,11 @@ used by the apr service device. = ASM DAIs (Digial Audio Interface) "dais" subnode of the ASM node represents dai specific configuration +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,q6asm-dais". + - #sound-dai-cells Usage: required Value type: <u32> @@ -28,6 +33,7 @@ q6asm@7 { compatible = "qcom,q6asm"; reg = <APR_SVC_ASM>; q6asmdai: dais { + compatible = "qcom,q6asm-dais"; #sound-dai-cells = <1>; }; }; diff --git a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt new file mode 100644 index 000000000000..408c4837e6d5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt @@ -0,0 +1,80 @@ +* Qualcomm Technologies Inc. SDM845 ASoC sound card driver + +This binding describes the SDM845 sound card, which uses qdsp for audio. + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,sdm845-sndcard" + +- audio-routing: + Usage: Optional + Value type: <stringlist> + Definition: A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names could be power supplies, MicBias + of codec and the jacks on the board. + +- model: + Usage: required + Value type: <stringlist> + Definition: The user-visible name of this sound card. + += dailinks +Each subnode of sndcard represents either a dailink, and subnodes of each +dailinks would be cpu/codec/platform dais. + +- link-name: + Usage: required + Value type: <string> + Definition: User friendly name for dai link + += CPU, PLATFORM, CODEC dais subnodes +- cpu: + Usage: required + Value type: <subnode> + Definition: cpu dai sub-node + +- codec: + Usage: required + Value type: <subnode> + Definition: codec dai sub-node + +- platform: + Usage: Optional + Value type: <subnode> + Definition: platform dai sub-node + +- sound-dai: + Usage: required + Value type: <phandle> + Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node. + +Example: + +audio { + compatible = "qcom,sdm845-sndcard"; + model = "sdm845-snd-card"; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&pri_mi2s_active &pri_mi2s_ws_active>; + pinctrl-1 = <&pri_mi2s_sleep &pri_mi2s_ws_sleep>; + + mm1-dai-link { + link-name = "MultiMedia1"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + pri-mi2s-dai-link { + link-name = "PRI MI2S Playback"; + cpu { + sound-dai = <&q6afedai PRIMARY_MI2S_RX>; + }; + + platform { + sound-dai = <&q6routing>; + }; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt b/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt new file mode 100644 index 000000000000..1d8d49e30af7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wcd9335.txt @@ -0,0 +1,123 @@ +QCOM WCD9335 Codec + +Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC, supports +Qualcomm Technologies, Inc. (QTI) multimedia solutions, including +the MSM8996, MSM8976, and MSM8956 chipsets. It has in-built +Soundwire controller, interrupt mux. It supports both I2S/I2C and +SLIMbus audio interfaces. + +Required properties with SLIMbus Interface: + +- compatible: + Usage: required + Value type: <stringlist> + Definition: For SLIMbus interface it should be "slimMID,PID", + textual representation of Manufacturer ID, Product Code, + shall be in lower case hexadecimal with leading zeroes + suppressed. Refer to slimbus/bus.txt for details. + Should be: + "slim217,1a0" for MSM8996 and APQ8096 SoCs with SLIMbus. + +- reg + Usage: required + Value type: <u32 u32> + Definition: Should be ('Device index', 'Instance ID') + +- interrupts + Usage: required + Value type: <prop-encoded-array> + Definition: Interrupts via WCD INTR1 and INTR2 pins + +- interrupt-names: + Usage: required + Value type: <String array> + Definition: Interrupt names of WCD INTR1 and INTR2 + Should be: "intr1", "intr2" + +- reset-gpio: + Usage: required + Value type: <String Array> + Definition: Reset gpio line + +- qcom,ifd: + Usage: required + Value type: <phandle> + Definition: SLIM interface device + +- clocks: + Usage: required + Value type: <prop-encoded-array> + Definition: See clock-bindings.txt section "consumers". List of + three clock specifiers for mclk, mclk2 and slimbus clock. + +- clock-names: + Usage: required + Value type: <string> + Definition: Must contain "mclk", "mclk2" and "slimbus" strings. + +- vdd-buck-supply: + Usage: required + Value type: <phandle> + Definition: Should contain a reference to the 1.8V buck supply + +- vdd-buck-sido-supply: + Usage: required + Value type: <phandle> + Definition: Should contain a reference to the 1.8V SIDO buck supply + +- vdd-rx-supply: + Usage: required + Value type: <phandle> + Definition: Should contain a reference to the 1.8V rx supply + +- vdd-tx-supply: + Usage: required + Value type: <phandle> + Definition: Should contain a reference to the 1.8V tx supply + +- vdd-vbat-supply: + Usage: Optional + Value type: <phandle> + Definition: Should contain a reference to the vbat supply + +- vdd-micbias-supply: + Usage: required + Value type: <phandle> + Definition: Should contain a reference to the micbias supply + +- vdd-io-supply: + Usage: required + Value type: <phandle> + Definition: Should contain a reference to the 1.8V io supply + +- interrupt-controller: + Usage: required + Definition: Indicating that this is a interrupt controller + +- #interrupt-cells: + Usage: required + Value type: <int> + Definition: should be 1 + +#sound-dai-cells + Usage: required + Value type: <u32> + Definition: Must be 1 + +codec@1{ + compatible = "slim217,1a0"; + reg = <1 0>; + interrupts = <&msmgpio 54 IRQ_TYPE_LEVEL_HIGH>; + interrupt-names = "intr2" + reset-gpio = <&msmgpio 64 0>; + qcom,ifd = <&wc9335_ifd>; + clock-names = "mclk", "native"; + clocks = <&rpmcc RPM_SMD_DIV_CLK1>, + <&rpmcc RPM_SMD_BB_CLK1>; + vdd-buck-supply = <&pm8994_s4>; + vdd-rx-supply = <&pm8994_s4>; + vdd-buck-sido-supply = <&pm8994_s4>; + vdd-tx-supply = <&pm8994_s4>; + vdd-io-supply = <&pm8994_s4>; + #sound-dai-cells = <1>; +} diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index b86d790f630f..9e764270c36b 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -352,6 +352,7 @@ Required properties: - "renesas,rcar_sound-r8a7794" (R-Car E2) - "renesas,rcar_sound-r8a7795" (R-Car H3) - "renesas,rcar_sound-r8a7796" (R-Car M3-W) + - "renesas,rcar_sound-r8a77965" (R-Car M3-N) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index b208a752576c..54aefab71f2c 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -7,6 +7,7 @@ Required properties: - compatible: should be one of the following: - "rockchip,rk3066-i2s": for rk3066 + - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30 - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036 - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228 diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt new file mode 100644 index 000000000000..312e9a129530 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -0,0 +1,50 @@ +RT5682 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5682" or "realtek,rt5682i" + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using GPIO5 pin as dmic1 data pin + +- realtek,dmic1-clk-pin + 0: using GPIO1 pin as dmic1 clock pin + 1: using GPIO3 pin as dmic1 clock pin + +- realtek,jd-src + 0: No JD is used + 1: using JD1 as JD source + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Pins on the device (for linking into audio routes) for RT5682: + + * DMIC L1 + * DMIC R1 + * IN1P + * HPOL + * HPOR + +Example: + +rt5682 { + compatible = "realtek,rt5682i"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; + realtek,dmic1-data-pin = <1>; + realtek,dmic1-clk-pin = <1>; + realtek,jd-src = <1>; +}; diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 0f214457476f..9c58f724396a 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -17,7 +17,7 @@ Optional properties: - VDDD-supply : the regulator provider of VDDD -- micbias-resistor-k-ohms : the bias resistor to be used in kOmhs +- micbias-resistor-k-ohms : the bias resistor to be used in kOhms The resistor can take values of 2k, 4k or 8k. If set to 0 it will be off. If this node is not mentioned or if the value is unknown, then diff --git a/Documentation/devicetree/bindings/sound/simple-amplifier.txt b/Documentation/devicetree/bindings/sound/simple-amplifier.txt new file mode 100644 index 000000000000..8647edae7af0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-amplifier.txt @@ -0,0 +1,12 @@ +Simple Amplifier Audio Driver + +Required properties: +- compatible : "dioo,dio2125" or "simple-audio-amplifier" +- enable-gpios : the gpio connected to the enable pin of the simple amplifier + +Example: + +amp: analog-amplifier { + compatible = "simple-audio-amplifier"; + enable-gpios = <&gpio GPIOH_3 0>; +}; diff --git a/Documentation/devicetree/bindings/sound/tas571x.txt b/Documentation/devicetree/bindings/sound/tas571x.txt index b4959f10b74b..7c8fd37c2f9e 100644 --- a/Documentation/devicetree/bindings/sound/tas571x.txt +++ b/Documentation/devicetree/bindings/sound/tas571x.txt @@ -7,6 +7,7 @@ powerdown (optional). Required properties: - compatible: should be one of the following: + - "ti,tas5707" - "ti,tas5711", - "ti,tas5717", - "ti,tas5719", diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 4d83c1c0ca04..4a3cecc8ad38 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -1568,7 +1568,7 @@ joystick_io The driver requires firmware files ``turtlebeach/msndinit.bin`` and ``turtlebeach/msndperm.bin`` in the proper firmware directory. -See Documentation/sound/oss/MultiSound for important information +See Documentation/sound/cards/multisound.sh for important information about this driver. Note that it has been discontinued, but the Voyetra Turtle Beach knowledge base entry for it is still available at diff --git a/Documentation/sound/cards/multisound.sh b/Documentation/sound/cards/multisound.sh new file mode 100755 index 000000000000..a915a1affcde --- /dev/null +++ b/Documentation/sound/cards/multisound.sh @@ -0,0 +1,1139 @@ +#! /bin/sh +# +# Turtle Beach MultiSound Driver Notes +# -- Andrew Veliath <andrewtv@usa.net> +# +# Last update: September 10, 1998 +# Corresponding msnd driver: 0.8.3 +# +# ** This file is a README (top part) and shell archive (bottom part). +# The corresponding archived utility sources can be unpacked by +# running `sh MultiSound' (the utilities are only needed for the +# Pinnacle and Fiji cards). ** +# +# +# -=-=- Getting Firmware -=-=- +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# See the section `Obtaining and Creating Firmware Files' in this +# document for instructions on obtaining the necessary firmware +# files. +# +# +# Supported Features +# ~~~~~~~~~~~~~~~~~~ +# +# Currently, full-duplex digital audio (/dev/dsp only, /dev/audio is +# not currently available) and mixer functionality (/dev/mixer) are +# supported (memory mapped digital audio is not yet supported). +# Digital transfers and monitoring can be done as well if you have +# the digital daughterboard (see the section on using the S/PDIF port +# for more information). +# +# Support for the Turtle Beach MultiSound Hurricane architecture is +# composed of the following modules (these can also operate compiled +# into the kernel): +# +# snd-msnd-lib - MultiSound base (requires snd) +# +# snd-msnd-classic - Base audio/mixer support for Classic, Monetery and +# Tahiti cards +# +# snd-msnd-pinnacle - Base audio/mixer support for Pinnacle and Fiji cards +# +# +# Important Notes - Read Before Using +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# The firmware files are not included (may change in future). You +# must obtain these images from Turtle Beach (they are included in +# the MultiSound Development Kits), and place them in /etc/sound for +# example, and give the full paths in the Linux configuration. If +# you are compiling in support for the MultiSound driver rather than +# using it as a module, these firmware files must be accessible +# during kernel compilation. +# +# Please note these files must be binary files, not assembler. See +# the section later in this document for instructions to obtain these +# files. +# +# +# Configuring Card Resources +# ~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# ** This section is very important, as your card may not work at all +# or your machine may crash if you do not do this correctly. ** +# +# * Classic/Monterey/Tahiti +# +# These cards are configured through the driver snd-msnd-classic. You must +# know the io port, then the driver will select the irq and memory resources +# on the card. It is up to you to know if these are free locations or now, +# a conflict can lock the machine up. +# +# * Pinnacle/Fiji +# +# The Pinnacle and Fiji cards have an extra config port, either +# 0x250, 0x260 or 0x270. This port can be disabled to have the card +# configured strictly through PnP, however you lose the ability to +# access the IDE controller and joystick devices on this card when +# using PnP. The included pinnaclecfg program in this shell archive +# can be used to configure the card in non-PnP mode, and in PnP mode +# you can use isapnptools. These are described briefly here. +# +# pinnaclecfg is not required; you can use the snd-msnd-pinnacle module +# to fully configure the card as well. However, pinnaclecfg can be +# used to change the resource values of a particular device after the +# snd-msnd-pinnacle module has been loaded. If you are compiling the +# driver into the kernel, you must set these values during compile +# time, however other peripheral resource values can be changed with +# the pinnaclecfg program after the kernel is loaded. +# +# +# *** PnP mode +# +# Use pnpdump to obtain a sample configuration if you can; I was able +# to obtain one with the command `pnpdump 1 0x203' -- this may vary +# for you (running pnpdump by itself did not work for me). Then, +# edit this file and use isapnp to uncomment and set the card values. +# Use these values when inserting the snd-msnd-pinnacle module. Using +# this method, you can set the resources for the DSP and the Kurzweil +# synth (Pinnacle). Since Linux does not directly support PnP +# devices, you may have difficulty when using the card in PnP mode +# when it the driver is compiled into the kernel. Using non-PnP mode +# is preferable in this case. +# +# Here is an example mypinnacle.conf for isapnp that sets the card to +# io base 0x210, irq 5 and mem 0xd8000, and also sets the Kurzweil +# synth to 0x330 and irq 9 (may need editing for your system): +# +# (READPORT 0x0203) +# (CSN 2) +# (IDENTIFY *) +# +# # DSP +# (CONFIGURE BVJ0440/-1 (LD 0 +# (INT 0 (IRQ 5 (MODE +E))) (IO 0 (BASE 0x0210)) (MEM 0 (BASE 0x0d8000)) +# (ACT Y))) +# +# # Kurzweil Synth (Pinnacle Only) +# (CONFIGURE BVJ0440/-1 (LD 1 +# (IO 0 (BASE 0x0330)) (INT 0 (IRQ 9 (MODE +E))) +# (ACT Y))) +# +# (WAITFORKEY) +# +# +# *** Non-PnP mode +# +# The second way is by running the card in non-PnP mode. This +# actually has some advantages in that you can access some other +# devices on the card, such as the joystick and IDE controller. To +# configure the card, unpack this shell archive and build the +# pinnaclecfg program. Using this program, you can assign the +# resource values to the card's devices, or disable the devices. As +# an alternative to using pinnaclecfg, you can specify many of the +# configuration values when loading the snd-msnd-pinnacle module (or +# during kernel configuration when compiling the driver into the +# kernel). +# +# If you specify cfg=0x250 for the snd-msnd-pinnacle module, it +# automatically configure the card to the given io, irq and memory +# values using that config port (the config port is jumper selectable +# on the card to 0x250, 0x260 or 0x270). +# +# See the `snd-msnd-pinnacle Additional Options' section below for more +# information on these parameters (also, if you compile the driver +# directly into the kernel, these extra parameters can be useful +# here). +# +# +# ** It is very easy to cause problems in your machine if you choose a +# resource value which is incorrect. ** +# +# +# Examples +# ~~~~~~~~ +# +# * MultiSound Classic/Monterey/Tahiti: +# +# modprobe snd +# insmod snd-msnd-lib +# insmod snd-msnd-classic io=0x290 irq=7 mem=0xd0000 +# +# * MultiSound Pinnacle in PnP mode: +# +# modprobe snd +# insmod snd-msnd-lib +# isapnp mypinnacle.conf +# insmod snd-msnd-pinnacle io=0x210 irq=5 mem=0xd8000 <-- match mypinnacle.conf values +# +# * MultiSound Pinnacle in non-PnP mode (replace 0x250 with your configuration port, +# one of 0x250, 0x260 or 0x270): +# +# modprobe snd +# insmod snd-msnd-lib +# insmod snd-msnd-pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 +# +# * To use the MPU-compatible Kurzweil synth on the Pinnacle in PnP +# mode, add the following (assumes you did `isapnp mypinnacle.conf'): +# +# insmod snd +# insmod mpu401 io=0x330 irq=9 <-- match mypinnacle.conf values +# +# * To use the MPU-compatible Kurzweil synth on the Pinnacle in non-PnP +# mode, add the following. Note how we first configure the peripheral's +# resources, _then_ install a Linux driver for it: +# +# insmod snd +# pinnaclecfg 0x250 mpu 0x330 9 +# insmod mpu401 io=0x330 irq=9 +# +# -- OR you can use the following sequence without pinnaclecfg in non-PnP mode: +# +# modprobe snd +# insmod snd-msnd-lib +# insmod snd-msnd-pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 mpu_io=0x330 mpu_irq=9 +# insmod snd +# insmod mpu401 io=0x330 irq=9 +# +# * To setup the joystick port on the Pinnacle in non-PnP mode (though +# you have to find the actual Linux joystick driver elsewhere), you +# can use pinnaclecfg: +# +# pinnaclecfg 0x250 joystick 0x200 +# +# -- OR you can configure this using snd-msnd-pinnacle with the following: +# +# modprobe snd +# insmod snd-msnd-lib +# insmod snd-msnd-pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 joystick_io=0x200 +# +# +# snd-msnd-classic, snd-msnd-pinnacle Required Options +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# If the following options are not given, the module will not load. +# Examine the kernel message log for informative error messages. +# WARNING--probing isn't supported so try to make sure you have the +# correct shared memory area, otherwise you may experience problems. +# +# io I/O base of DSP, e.g. io=0x210 +# irq IRQ number, e.g. irq=5 +# mem Shared memory area, e.g. mem=0xd8000 +# +# +# snd-msnd-classic, snd-msnd-pinnacle Additional Options +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# fifosize The digital audio FIFOs, in kilobytes. If not +# specified, the default will be used. Increasing +# this value will reduce the chance of a FIFO +# underflow at the expense of increasing overall +# latency. For example, fifosize=512 will +# allocate 512kB read and write FIFOs (1MB total). +# While this may reduce dropouts, a heavy machine +# load will undoubtedly starve the FIFO of data +# and you will eventually get dropouts. One +# option is to alter the scheduling priority of +# the playback process, using `nice' or some form +# of POSIX soft real-time scheduling. +# +# calibrate_signal Setting this to one calibrates the ADCs to the +# signal, zero calibrates to the card (defaults +# to zero). +# +# +# snd-msnd-pinnacle Additional Options +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# digital Specify digital=1 to enable the S/PDIF input +# if you have the digital daughterboard +# adapter. This will enable access to the +# DIGITAL1 input for the soundcard in the mixer. +# Some mixer programs might have trouble setting +# the DIGITAL1 source as an input. If you have +# trouble, you can try the setdigital.c program +# at the bottom of this document. +# +# cfg Non-PnP configuration port for the Pinnacle +# and Fiji (typically 0x250, 0x260 or 0x270, +# depending on the jumper configuration). If +# this option is omitted, then it is assumed +# that the card is in PnP mode, and that the +# specified DSP resource values are already +# configured with PnP (i.e. it won't attempt to +# do any sort of configuration). +# +# When the Pinnacle is in non-PnP mode, you can use the following +# options to configure particular devices. If a full specification +# for a device is not given, then the device is not configured. Note +# that you still must use a Linux driver for any of these devices +# once their resources are setup (such as the Linux joystick driver, +# or the MPU401 driver from OSS for the Kurzweil synth). +# +# mpu_io I/O port of MPU (on-board Kurzweil synth) +# mpu_irq IRQ of MPU (on-board Kurzweil synth) +# ide_io0 First I/O port of IDE controller +# ide_io1 Second I/O port of IDE controller +# ide_irq IRQ IDE controller +# joystick_io I/O port of joystick +# +# +# Obtaining and Creating Firmware Files +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# For the Classic/Tahiti/Monterey +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# Download to /tmp and unzip the following file from Turtle Beach: +# +# ftp://ftp.voyetra.com/pub/tbs/msndcl/msndvkit.zip +# +# When unzipped, unzip the file named MsndFiles.zip. Then copy the +# following firmware files to /etc/sound (note the file renaming): +# +# cp DSPCODE/MSNDINIT.BIN /etc/sound/msndinit.bin +# cp DSPCODE/MSNDPERM.REB /etc/sound/msndperm.bin +# +# When configuring the Linux kernel, specify /etc/sound/msndinit.bin and +# /etc/sound/msndperm.bin for the two firmware files (Linux kernel +# versions older than 2.2 do not ask for firmware paths, and are +# hardcoded to /etc/sound). +# +# If you are compiling the driver into the kernel, these files must +# be accessible during compilation, but will not be needed later. +# The files must remain, however, if the driver is used as a module. +# +# +# For the Pinnacle/Fiji +# ~~~~~~~~~~~~~~~~~~~~~ +# +# Download to /tmp and unzip the following file from Turtle Beach (be +# sure to use the entire URL; some have had trouble navigating to the +# URL): +# +# ftp://ftp.voyetra.com/pub/tbs/pinn/pnddk100.zip +# +# Unpack this shell archive, and run make in the created directory +# (you need a C compiler and flex to build the utilities). This +# should give you the executables conv, pinnaclecfg and setdigital. +# conv is only used temporarily here to create the firmware files, +# while pinnaclecfg is used to configure the Pinnacle or Fiji card in +# non-PnP mode, and setdigital can be used to set the S/PDIF input on +# the mixer (pinnaclecfg and setdigital should be copied to a +# convenient place, possibly run during system initialization). +# +# To generating the firmware files with the `conv' program, we create +# the binary firmware files by doing the following conversion +# (assuming the archive unpacked into a directory named PINNDDK): +# +# ./conv < PINNDDK/dspcode/pndspini.asm > /etc/sound/pndspini.bin +# ./conv < PINNDDK/dspcode/pndsperm.asm > /etc/sound/pndsperm.bin +# +# The conv (and conv.l) program is not needed after conversion and can +# be safely deleted. Then, when configuring the Linux kernel, specify +# /etc/sound/pndspini.bin and /etc/sound/pndsperm.bin for the two +# firmware files (Linux kernel versions older than 2.2 do not ask for +# firmware paths, and are hardcoded to /etc/sound). +# +# If you are compiling the driver into the kernel, these files must +# be accessible during compilation, but will not be needed later. +# The files must remain, however, if the driver is used as a module. +# +# +# Using Digital I/O with the S/PDIF Port +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# If you have a Pinnacle or Fiji with the digital daughterboard and +# want to set it as the input source, you can use this program if you +# have trouble trying to do it with a mixer program (be sure to +# insert the module with the digital=1 option, or say Y to the option +# during compiled-in kernel operation). Upon selection of the S/PDIF +# port, you should be able monitor and record from it. +# +# There is something to note about using the S/PDIF port. Digital +# timing is taken from the digital signal, so if a signal is not +# connected to the port and it is selected as recording input, you +# will find PCM playback to be distorted in playback rate. Also, +# attempting to record at a sampling rate other than the DAT rate may +# be problematic (i.e. trying to record at 8000Hz when the DAT signal +# is 44100Hz). If you have a problem with this, set the recording +# input to analog if you need to record at a rate other than that of +# the DAT rate. +# +# +# -- Shell archive attached below, just run `sh MultiSound' to extract. +# Contains Pinnacle/Fiji utilities to convert firmware, configure +# in non-PnP mode, and select the DIGITAL1 input for the mixer. +# +# +#!/bin/sh +# This is a shell archive (produced by GNU sharutils 4.2). +# To extract the files from this archive, save it to some FILE, remove +# everything before the `!/bin/sh' line above, then type `sh FILE'. +# +# Made on 1998-12-04 10:07 EST by <andrewtv@ztransform.velsoft.com>. +# Source directory was `/home/andrewtv/programming/pinnacle/pinnacle'. +# +# Existing files will *not* be overwritten unless `-c' is specified. +# +# This shar contains: +# length mode name +# ------ ---------- ------------------------------------------ +# 2064 -rw-rw-r-- MultiSound.d/setdigital.c +# 10224 -rw-rw-r-- MultiSound.d/pinnaclecfg.c +# 106 -rw-rw-r-- MultiSound.d/Makefile +# 146 -rw-rw-r-- MultiSound.d/conv.l +# 1491 -rw-rw-r-- MultiSound.d/msndreset.c +# +save_IFS="${IFS}" +IFS="${IFS}:" +gettext_dir=FAILED +locale_dir=FAILED +first_param="$1" +for dir in $PATH +do + if test "$gettext_dir" = FAILED && test -f $dir/gettext \ + && ($dir/gettext --version >/dev/null 2>&1) + then + set `$dir/gettext --version 2>&1` + if test "$3" = GNU + then + gettext_dir=$dir + fi + fi + if test "$locale_dir" = FAILED && test -f $dir/shar \ + && ($dir/shar --print-text-domain-dir >/dev/null 2>&1) + then + locale_dir=`$dir/shar --print-text-domain-dir` + fi +done +IFS="$save_IFS" +if test "$locale_dir" = FAILED || test "$gettext_dir" = FAILED +then + echo=echo +else + TEXTDOMAINDIR=$locale_dir + export TEXTDOMAINDIR + TEXTDOMAIN=sharutils + export TEXTDOMAIN + echo="$gettext_dir/gettext -s" +fi +touch -am 1231235999 $$.touch >/dev/null 2>&1 +if test ! -f 1231235999 && test -f $$.touch; then + shar_touch=touch +else + shar_touch=: + echo + $echo 'WARNING: not restoring timestamps. Consider getting and' + $echo "installing GNU \`touch', distributed in GNU File Utilities..." + echo +fi +rm -f 1231235999 $$.touch +# +if mkdir _sh01426; then + $echo 'x -' 'creating lock directory' +else + $echo 'failed to create lock directory' + exit 1 +fi +# ============= MultiSound.d/setdigital.c ============== +if test ! -d 'MultiSound.d'; then + $echo 'x -' 'creating directory' 'MultiSound.d' + mkdir 'MultiSound.d' +fi +if test -f 'MultiSound.d/setdigital.c' && test "$first_param" != -c; then + $echo 'x -' SKIPPING 'MultiSound.d/setdigital.c' '(file already exists)' +else + $echo 'x -' extracting 'MultiSound.d/setdigital.c' '(text)' + sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/setdigital.c' && +/********************************************************************* +X * +X * setdigital.c - sets the DIGITAL1 input for a mixer +X * +X * Copyright (C) 1998 Andrew Veliath +X * +X * This program is free software; you can redistribute it and/or modify +X * it under the terms of the GNU General Public License as published by +X * the Free Software Foundation; either version 2 of the License, or +X * (at your option) any later version. +X * +X * This program is distributed in the hope that it will be useful, +X * but WITHOUT ANY WARRANTY; without even the implied warranty of +X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +X * GNU General Public License for more details. +X * +X * You should have received a copy of the GNU General Public License +X * along with this program; if not, write to the Free Software +X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. +X * +X ********************************************************************/ +X +#include <stdio.h> +#include <stdlib.h> +#include <unistd.h> +#include <fcntl.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <sys/ioctl.h> +#include <sys/soundcard.h> +X +int main(int argc, char *argv[]) +{ +X int fd; +X unsigned long recmask, recsrc; +X +X if (argc != 2) { +X fprintf(stderr, "usage: setdigital <mixer device>\n"); +X exit(1); +X } +X +X if ((fd = open(argv[1], O_RDWR)) < 0) { +X perror(argv[1]); +X exit(1); +X } +X +X if (ioctl(fd, SOUND_MIXER_READ_RECMASK, &recmask) < 0) { +X fprintf(stderr, "error: ioctl read recording mask failed\n"); +X perror("ioctl"); +X close(fd); +X exit(1); +X } +X +X if (!(recmask & SOUND_MASK_DIGITAL1)) { +X fprintf(stderr, "error: cannot find DIGITAL1 device in mixer\n"); +X close(fd); +X exit(1); +X } +X +X if (ioctl(fd, SOUND_MIXER_READ_RECSRC, &recsrc) < 0) { +X fprintf(stderr, "error: ioctl read recording source failed\n"); +X perror("ioctl"); +X close(fd); +X exit(1); +X } +X +X recsrc |= SOUND_MASK_DIGITAL1; +X +X if (ioctl(fd, SOUND_MIXER_WRITE_RECSRC, &recsrc) < 0) { +X fprintf(stderr, "error: ioctl write recording source failed\n"); +X perror("ioctl"); +X close(fd); +X exit(1); +X } +X +X close(fd); +X +X return 0; +} +SHAR_EOF + $shar_touch -am 1204092598 'MultiSound.d/setdigital.c' && + chmod 0664 'MultiSound.d/setdigital.c' || + $echo 'restore of' 'MultiSound.d/setdigital.c' 'failed' + if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ + && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then + md5sum -c << SHAR_EOF >/dev/null 2>&1 \ + || $echo 'MultiSound.d/setdigital.c:' 'MD5 check failed' +e87217fc3e71288102ba41fd81f71ec4 MultiSound.d/setdigital.c +SHAR_EOF + else + shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/setdigital.c'`" + test 2064 -eq "$shar_count" || + $echo 'MultiSound.d/setdigital.c:' 'original size' '2064,' 'current size' "$shar_count!" + fi +fi +# ============= MultiSound.d/pinnaclecfg.c ============== +if test -f 'MultiSound.d/pinnaclecfg.c' && test "$first_param" != -c; then + $echo 'x -' SKIPPING 'MultiSound.d/pinnaclecfg.c' '(file already exists)' +else + $echo 'x -' extracting 'MultiSound.d/pinnaclecfg.c' '(text)' + sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/pinnaclecfg.c' && +/********************************************************************* +X * +X * pinnaclecfg.c - Pinnacle/Fiji Device Configuration Program +X * +X * This is for NON-PnP mode only. For PnP mode, use isapnptools. +X * +X * This is Linux-specific, and must be run with root permissions. +X * +X * Part of the Turtle Beach MultiSound Sound Card Driver for Linux +X * +X * Copyright (C) 1998 Andrew Veliath +X * +X * This program is free software; you can redistribute it and/or modify +X * it under the terms of the GNU General Public License as published by +X * the Free Software Foundation; either version 2 of the License, or +X * (at your option) any later version. +X * +X * This program is distributed in the hope that it will be useful, +X * but WITHOUT ANY WARRANTY; without even the implied warranty of +X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +X * GNU General Public License for more details. +X * +X * You should have received a copy of the GNU General Public License +X * along with this program; if not, write to the Free Software +X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. +X * +X ********************************************************************/ +X +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <errno.h> +#include <unistd.h> +#include <asm/types.h> +#include <sys/io.h> +X +#define IREG_LOGDEVICE 0x07 +#define IREG_ACTIVATE 0x30 +#define LD_ACTIVATE 0x01 +#define LD_DISACTIVATE 0x00 +#define IREG_EECONTROL 0x3F +#define IREG_MEMBASEHI 0x40 +#define IREG_MEMBASELO 0x41 +#define IREG_MEMCONTROL 0x42 +#define IREG_MEMRANGEHI 0x43 +#define IREG_MEMRANGELO 0x44 +#define MEMTYPE_8BIT 0x00 +#define MEMTYPE_16BIT 0x02 +#define MEMTYPE_RANGE 0x00 +#define MEMTYPE_HIADDR 0x01 +#define IREG_IO0_BASEHI 0x60 +#define IREG_IO0_BASELO 0x61 +#define IREG_IO1_BASEHI 0x62 +#define IREG_IO1_BASELO 0x63 +#define IREG_IRQ_NUMBER 0x70 +#define IREG_IRQ_TYPE 0x71 +#define IRQTYPE_HIGH 0x02 +#define IRQTYPE_LOW 0x00 +#define IRQTYPE_LEVEL 0x01 +#define IRQTYPE_EDGE 0x00 +X +#define HIBYTE(w) ((BYTE)(((WORD)(w) >> 8) & 0xFF)) +#define LOBYTE(w) ((BYTE)(w)) +#define MAKEWORD(low,hi) ((WORD)(((BYTE)(low))|(((WORD)((BYTE)(hi)))<<8))) +X +typedef __u8 BYTE; +typedef __u16 USHORT; +typedef __u16 WORD; +X +static int config_port = -1; +X +static int msnd_write_cfg(int cfg, int reg, int value) +{ +X outb(reg, cfg); +X outb(value, cfg + 1); +X if (value != inb(cfg + 1)) { +X fprintf(stderr, "error: msnd_write_cfg: I/O error\n"); +X return -EIO; +X } +X return 0; +} +X +static int msnd_read_cfg(int cfg, int reg) +{ +X outb(reg, cfg); +X return inb(cfg + 1); +} +X +static int msnd_write_cfg_io0(int cfg, int num, WORD io) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io))) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io))) +X return -EIO; +X return 0; +} +X +static int msnd_read_cfg_io0(int cfg, int num, WORD *io) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X +X *io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO0_BASELO), +X msnd_read_cfg(cfg, IREG_IO0_BASEHI)); +X +X return 0; +} +X +static int msnd_write_cfg_io1(int cfg, int num, WORD io) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io))) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io))) +X return -EIO; +X return 0; +} +X +static int msnd_read_cfg_io1(int cfg, int num, WORD *io) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X +X *io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO1_BASELO), +X msnd_read_cfg(cfg, IREG_IO1_BASEHI)); +X +X return 0; +} +X +static int msnd_write_cfg_irq(int cfg, int num, WORD irq) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq))) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE)) +X return -EIO; +X return 0; +} +X +static int msnd_read_cfg_irq(int cfg, int num, WORD *irq) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X +X *irq = msnd_read_cfg(cfg, IREG_IRQ_NUMBER); +X +X return 0; +} +X +static int msnd_write_cfg_mem(int cfg, int num, int mem) +{ +X WORD wmem; +X +X mem >>= 8; +X mem &= 0xfff; +X wmem = (WORD)mem; +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem))) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem))) +X return -EIO; +X if (wmem && msnd_write_cfg(cfg, IREG_MEMCONTROL, (MEMTYPE_HIADDR | MEMTYPE_16BIT))) +X return -EIO; +X return 0; +} +X +static int msnd_read_cfg_mem(int cfg, int num, int *mem) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X +X *mem = MAKEWORD(msnd_read_cfg(cfg, IREG_MEMBASELO), +X msnd_read_cfg(cfg, IREG_MEMBASEHI)); +X *mem <<= 8; +X +X return 0; +} +X +static int msnd_activate_logical(int cfg, int num) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE)) +X return -EIO; +X return 0; +} +X +static int msnd_write_cfg_logical(int cfg, int num, WORD io0, WORD io1, WORD irq, int mem) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg_io0(cfg, num, io0)) +X return -EIO; +X if (msnd_write_cfg_io1(cfg, num, io1)) +X return -EIO; +X if (msnd_write_cfg_irq(cfg, num, irq)) +X return -EIO; +X if (msnd_write_cfg_mem(cfg, num, mem)) +X return -EIO; +X if (msnd_activate_logical(cfg, num)) +X return -EIO; +X return 0; +} +X +static int msnd_read_cfg_logical(int cfg, int num, WORD *io0, WORD *io1, WORD *irq, int *mem) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_read_cfg_io0(cfg, num, io0)) +X return -EIO; +X if (msnd_read_cfg_io1(cfg, num, io1)) +X return -EIO; +X if (msnd_read_cfg_irq(cfg, num, irq)) +X return -EIO; +X if (msnd_read_cfg_mem(cfg, num, mem)) +X return -EIO; +X return 0; +} +X +static void usage(void) +{ +X fprintf(stderr, +X "\n" +X "pinnaclecfg 1.0\n" +X "\n" +X "usage: pinnaclecfg <config port> [device config]\n" +X "\n" +X "This is for use with the card in NON-PnP mode only.\n" +X "\n" +X "Available devices (not all available for Fiji):\n" +X "\n" +X " Device Description\n" +X " -------------------------------------------------------------------\n" +X " reset Reset all devices (i.e. disable)\n" +X " show Display current device configurations\n" +X "\n" +X " dsp <io> <irq> <mem> Audio device\n" +X " mpu <io> <irq> Internal Kurzweil synth\n" +X " ide <io0> <io1> <irq> On-board IDE controller\n" +X " joystick <io> Joystick port\n" +X "\n"); +X exit(1); +} +X +static int cfg_reset(void) +{ +X int i; +X +X for (i = 0; i < 4; ++i) +X msnd_write_cfg_logical(config_port, i, 0, 0, 0, 0); +X +X return 0; +} +X +static int cfg_show(void) +{ +X int i; +X int count = 0; +X +X for (i = 0; i < 4; ++i) { +X WORD io0, io1, irq; +X int mem; +X msnd_read_cfg_logical(config_port, i, &io0, &io1, &irq, &mem); +X switch (i) { +X case 0: +X if (io0 || irq || mem) { +X printf("dsp 0x%x %d 0x%x\n", io0, irq, mem); +X ++count; +X } +X break; +X case 1: +X if (io0 || irq) { +X printf("mpu 0x%x %d\n", io0, irq); +X ++count; +X } +X break; +X case 2: +X if (io0 || io1 || irq) { +X printf("ide 0x%x 0x%x %d\n", io0, io1, irq); +X ++count; +X } +X break; +X case 3: +X if (io0) { +X printf("joystick 0x%x\n", io0); +X ++count; +X } +X break; +X } +X } +X +X if (count == 0) +X fprintf(stderr, "no devices configured\n"); +X +X return 0; +} +X +static int cfg_dsp(int argc, char *argv[]) +{ +X int io, irq, mem; +X +X if (argc < 3 || +X sscanf(argv[0], "0x%x", &io) != 1 || +X sscanf(argv[1], "%d", &irq) != 1 || +X sscanf(argv[2], "0x%x", &mem) != 1) +X usage(); +X +X if (!(io == 0x290 || +X io == 0x260 || +X io == 0x250 || +X io == 0x240 || +X io == 0x230 || +X io == 0x220 || +X io == 0x210 || +X io == 0x3e0)) { +X fprintf(stderr, "error: io must be one of " +X "210, 220, 230, 240, 250, 260, 290, or 3E0\n"); +X usage(); +X } +X +X if (!(irq == 5 || +X irq == 7 || +X irq == 9 || +X irq == 10 || +X irq == 11 || +X irq == 12)) { +X fprintf(stderr, "error: irq must be one of " +X "5, 7, 9, 10, 11 or 12\n"); +X usage(); +X } +X +X if (!(mem == 0xb0000 || +X mem == 0xc8000 || +X mem == 0xd0000 || +X mem == 0xd8000 || +X mem == 0xe0000 || +X mem == 0xe8000)) { +X fprintf(stderr, "error: mem must be one of " +X "0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000\n"); +X usage(); +X } +X +X return msnd_write_cfg_logical(config_port, 0, io, 0, irq, mem); +} +X +static int cfg_mpu(int argc, char *argv[]) +{ +X int io, irq; +X +X if (argc < 2 || +X sscanf(argv[0], "0x%x", &io) != 1 || +X sscanf(argv[1], "%d", &irq) != 1) +X usage(); +X +X return msnd_write_cfg_logical(config_port, 1, io, 0, irq, 0); +} +X +static int cfg_ide(int argc, char *argv[]) +{ +X int io0, io1, irq; +X +X if (argc < 3 || +X sscanf(argv[0], "0x%x", &io0) != 1 || +X sscanf(argv[0], "0x%x", &io1) != 1 || +X sscanf(argv[1], "%d", &irq) != 1) +X usage(); +X +X return msnd_write_cfg_logical(config_port, 2, io0, io1, irq, 0); +} +X +static int cfg_joystick(int argc, char *argv[]) +{ +X int io; +X +X if (argc < 1 || +X sscanf(argv[0], "0x%x", &io) != 1) +X usage(); +X +X return msnd_write_cfg_logical(config_port, 3, io, 0, 0, 0); +} +X +int main(int argc, char *argv[]) +{ +X char *device; +X int rv = 0; +X +X --argc; ++argv; +X +X if (argc < 2) +X usage(); +X +X sscanf(argv[0], "0x%x", &config_port); +X if (config_port != 0x250 && config_port != 0x260 && config_port != 0x270) { +X fprintf(stderr, "error: <config port> must be 0x250, 0x260 or 0x270\n"); +X exit(1); +X } +X if (ioperm(config_port, 2, 1)) { +X perror("ioperm"); +X fprintf(stderr, "note: pinnaclecfg must be run as root\n"); +X exit(1); +X } +X device = argv[1]; +X +X argc -= 2; argv += 2; +X +X if (strcmp(device, "reset") == 0) +X rv = cfg_reset(); +X else if (strcmp(device, "show") == 0) +X rv = cfg_show(); +X else if (strcmp(device, "dsp") == 0) +X rv = cfg_dsp(argc, argv); +X else if (strcmp(device, "mpu") == 0) +X rv = cfg_mpu(argc, argv); +X else if (strcmp(device, "ide") == 0) +X rv = cfg_ide(argc, argv); +X else if (strcmp(device, "joystick") == 0) +X rv = cfg_joystick(argc, argv); +X else { +X fprintf(stderr, "error: unknown device %s\n", device); +X usage(); +X } +X +X if (rv) +X fprintf(stderr, "error: device configuration failed\n"); +X +X return 0; +} +SHAR_EOF + $shar_touch -am 1204092598 'MultiSound.d/pinnaclecfg.c' && + chmod 0664 'MultiSound.d/pinnaclecfg.c' || + $echo 'restore of' 'MultiSound.d/pinnaclecfg.c' 'failed' + if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ + && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then + md5sum -c << SHAR_EOF >/dev/null 2>&1 \ + || $echo 'MultiSound.d/pinnaclecfg.c:' 'MD5 check failed' +366bdf27f0db767a3c7921d0a6db20fe MultiSound.d/pinnaclecfg.c +SHAR_EOF + else + shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/pinnaclecfg.c'`" + test 10224 -eq "$shar_count" || + $echo 'MultiSound.d/pinnaclecfg.c:' 'original size' '10224,' 'current size' "$shar_count!" + fi +fi +# ============= MultiSound.d/Makefile ============== +if test -f 'MultiSound.d/Makefile' && test "$first_param" != -c; then + $echo 'x -' SKIPPING 'MultiSound.d/Makefile' '(file already exists)' +else + $echo 'x -' extracting 'MultiSound.d/Makefile' '(text)' + sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/Makefile' && +CC = gcc +CFLAGS = -O +PROGS = setdigital msndreset pinnaclecfg conv +X +all: $(PROGS) +X +clean: +X rm -f $(PROGS) +SHAR_EOF + $shar_touch -am 1204092398 'MultiSound.d/Makefile' && + chmod 0664 'MultiSound.d/Makefile' || + $echo 'restore of' 'MultiSound.d/Makefile' 'failed' + if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ + && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then + md5sum -c << SHAR_EOF >/dev/null 2>&1 \ + || $echo 'MultiSound.d/Makefile:' 'MD5 check failed' +76ca8bb44e3882edcf79c97df6c81845 MultiSound.d/Makefile +SHAR_EOF + else + shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/Makefile'`" + test 106 -eq "$shar_count" || + $echo 'MultiSound.d/Makefile:' 'original size' '106,' 'current size' "$shar_count!" + fi +fi +# ============= MultiSound.d/conv.l ============== +if test -f 'MultiSound.d/conv.l' && test "$first_param" != -c; then + $echo 'x -' SKIPPING 'MultiSound.d/conv.l' '(file already exists)' +else + $echo 'x -' extracting 'MultiSound.d/conv.l' '(text)' + sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/conv.l' && +%% +[ \n\t,\r] +\;.* +DB +[0-9A-Fa-f]+H { int n; sscanf(yytext, "%xH", &n); printf("%c", n); } +%% +int yywrap() { return 1; } +void main() { yylex(); } +SHAR_EOF + $shar_touch -am 0828231798 'MultiSound.d/conv.l' && + chmod 0664 'MultiSound.d/conv.l' || + $echo 'restore of' 'MultiSound.d/conv.l' 'failed' + if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ + && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then + md5sum -c << SHAR_EOF >/dev/null 2>&1 \ + || $echo 'MultiSound.d/conv.l:' 'MD5 check failed' +d2411fc32cd71a00dcdc1f009e858dd2 MultiSound.d/conv.l +SHAR_EOF + else + shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/conv.l'`" + test 146 -eq "$shar_count" || + $echo 'MultiSound.d/conv.l:' 'original size' '146,' 'current size' "$shar_count!" + fi +fi +# ============= MultiSound.d/msndreset.c ============== +if test -f 'MultiSound.d/msndreset.c' && test "$first_param" != -c; then + $echo 'x -' SKIPPING 'MultiSound.d/msndreset.c' '(file already exists)' +else + $echo 'x -' extracting 'MultiSound.d/msndreset.c' '(text)' + sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/msndreset.c' && +/********************************************************************* +X * +X * msndreset.c - resets the MultiSound card +X * +X * Copyright (C) 1998 Andrew Veliath +X * +X * This program is free software; you can redistribute it and/or modify +X * it under the terms of the GNU General Public License as published by +X * the Free Software Foundation; either version 2 of the License, or +X * (at your option) any later version. +X * +X * This program is distributed in the hope that it will be useful, +X * but WITHOUT ANY WARRANTY; without even the implied warranty of +X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +X * GNU General Public License for more details. +X * +X * You should have received a copy of the GNU General Public License +X * along with this program; if not, write to the Free Software +X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. +X * +X ********************************************************************/ +X +#include <stdio.h> +#include <stdlib.h> +#include <unistd.h> +#include <fcntl.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <sys/ioctl.h> +#include <sys/soundcard.h> +X +int main(int argc, char *argv[]) +{ +X int fd; +X +X if (argc != 2) { +X fprintf(stderr, "usage: msndreset <mixer device>\n"); +X exit(1); +X } +X +X if ((fd = open(argv[1], O_RDWR)) < 0) { +X perror(argv[1]); +X exit(1); +X } +X +X if (ioctl(fd, SOUND_MIXER_PRIVATE1, 0) < 0) { +X fprintf(stderr, "error: msnd ioctl reset failed\n"); +X perror("ioctl"); +X close(fd); +X exit(1); +X } +X +X close(fd); +X +X return 0; +} +SHAR_EOF + $shar_touch -am 1204100698 'MultiSound.d/msndreset.c' && + chmod 0664 'MultiSound.d/msndreset.c' || + $echo 'restore of' 'MultiSound.d/msndreset.c' 'failed' + if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ + && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then + md5sum -c << SHAR_EOF >/dev/null 2>&1 \ + || $echo 'MultiSound.d/msndreset.c:' 'MD5 check failed' +c52f876521084e8eb25e12e01dcccb8a MultiSound.d/msndreset.c +SHAR_EOF + else + shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/msndreset.c'`" + test 1491 -eq "$shar_count" || + $echo 'MultiSound.d/msndreset.c:' 'original size' '1491,' 'current size' "$shar_count!" + fi +fi +rm -fr _sh01426 +exit 0 diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index 7c2d37571af0..e06238131f77 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -34,6 +34,22 @@ ALC262 ====== inv-dmic Inverted internal mic workaround +fsc-h270 + Fixups for Fujitsu-Siemens Celsius H270 +fsc-s7110 + Fixups for Fujitsu-Siemens Lifebook S7110 +hp-z200 + Fixups for HP Z200 +tyan + Fixups for Tyan Thunder n6650W +lenovo-3000 + Fixups for Lenovo 3000 +benq + Fixups for Benq ED8 +benq-t31 + Fixups for Benq T31 +bayleybay + Fixups for Intel BayleyBay ALC267/268 ========== @@ -41,6 +57,8 @@ inv-dmic Inverted internal mic workaround hp-eapd Disable HP EAPD on NID 0x15 +spdif + Enable SPDIF output on NID 0x1e ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models) =================================================================== @@ -70,6 +88,10 @@ dell-headset-multi Headset jack, which can also be used as mic-in dell-headset-dock Headset jack (without mic-in), and also dock I/O +dell-headset3 + Headset jack (without mic-in), and also dock I/O, variant 3 +dell-headset4 + Headset jack (without mic-in), and also dock I/O, variant 4 alc283-dac-wcaps Fixups for Chromebook with ALC283 alc283-sense-combo @@ -80,15 +102,173 @@ tpt440 Lenovo Thinkpad T440s setup tpt460 Lenovo Thinkpad T460/560 setup +tpt470-dock + Lenovo Thinkpad T470 dock setup dual-codecs Lenovo laptops with dual codecs alc700-ref Intel reference board with ALC700 codec +vaio + Pin fixups for Sony VAIO laptops +dell-m101z + COEF setup for Dell M101z +asus-g73jw + Subwoofer pin fixup for ASUS G73JW +lenovo-eapd + Inversed EAPD setup for Lenovo laptops +sony-hweq + H/W EQ COEF setup for Sony laptops +pcm44k + Fixed PCM 44kHz constraints (for buggy devices) +lifebook + Dock pin fixups for Fujitsu Lifebook +lifebook-extmic + Headset mic fixup for Fujitsu Lifebook +lifebook-hp-pin + Headphone pin fixup for Fujitsu Lifebook +lifebook-u7x7 + Lifebook U7x7 fixups +alc269vb-amic + ALC269VB analog mic pin fixups +alc269vb-dmic + ALC269VB digital mic pin fixups +hp-mute-led-mic1 + Mute LED via Mic1 pin on HP +hp-mute-led-mic2 + Mute LED via Mic2 pin on HP +hp-mute-led-mic3 + Mute LED via Mic3 pin on HP +hp-gpio-mic1 + GPIO + Mic1 pin LED on HP +hp-line1-mic1 + Mute LED via Line1 + Mic1 pins on HP +noshutup + Skip shutup callback +sony-nomic + Headset mic fixup for Sony laptops +aspire-headset-mic + Headset pin fixup for Acer Aspire +asus-x101 + ASUS X101 fixups +acer-ao7xx + Acer AO7xx fixups +acer-aspire-e1 + Acer Aspire E1 fixups +acer-ac700 + Acer AC700 fixups +limit-mic-boost + Limit internal mic boost on Lenovo machines +asus-zenbook + ASUS Zenbook fixups +asus-zenbook-ux31a + ASUS Zenbook UX31A fixups +ordissimo + Ordissimo EVE2 (or Malata PC-B1303) fixups +asus-tx300 + ASUS TX300 fixups +alc283-int-mic + ALC283 COEF setup for Lenovo machines +mono-speakers + Subwoofer and headset fixupes for Dell Inspiron +alc290-subwoofer + Subwoofer fixups for Dell Vostro +thinkpad + Binding with thinkpad_acpi driver for Lenovo machines +dmic-thinkpad + thinkpad_acpi binding + digital mic support +alc255-acer + ALC255 fixups on Acer machines +alc255-asus + ALC255 fixups on ASUS machines +alc255-dell1 + ALC255 fixups on Dell machines +alc255-dell2 + ALC255 fixups on Dell machines, variant 2 +alc293-dell1 + ALC293 fixups on Dell machines +alc283-headset + Headset pin fixups on ALC283 +aspire-v5 + Acer Aspire V5 fixups +hp-gpio4 + GPIO and Mic1 pin mute LED fixups for HP +hp-gpio-led + GPIO mute LEDs on HP +hp-gpio2-hotkey + GPIO mute LED with hot key handling on HP +hp-dock-pins + GPIO mute LEDs and dock support on HP +hp-dock-gpio-mic + GPIO, Mic mute LED and dock support on HP +hp-9480m + HP 9480m fixups +alc288-dell1 + ALC288 fixups on Dell machines +alc288-dell-xps13 + ALC288 fixups on Dell XPS13 +dell-e7x + Dell E7x fixups +alc293-dell + ALC293 fixups on Dell machines +alc298-dell1 + ALC298 fixups on Dell machines +alc298-dell-aio + ALC298 fixups on Dell AIO machines +alc275-dell-xps + ALC275 fixups on Dell XPS models +alc256-dell-xps13 + ALC256 fixups on Dell XPS13 +lenovo-spk-noise + Workaround for speaker noise on Lenovo machines +lenovo-hotkey + Hot-key support via Mic2 pin on Lenovo machines +dell-spk-noise + Workaround for speaker noise on Dell machines +alc255-dell1 + ALC255 fixups on Dell machines +alc295-disable-dac3 + Disable DAC3 routing on ALC295 +alc280-hp-headset + HP Elitebook fixups +alc221-hp-mic + Front mic pin fixup on HP machines +alc298-spk-volume + Speaker pin routing workaround on ALC298 +dell-inspiron-7559 + Dell Inspiron 7559 fixups +ativ-book + Samsung Ativ book 8 fixups +alc221-hp-mic + ALC221 headset fixups on HP machines +alc256-asus-mic + ALC256 fixups on ASUS machines +alc256-asus-aio + ALC256 fixups on ASUS AIO machines +alc233-eapd + ALC233 fixups on ASUS machines +alc294-lenovo-mic + ALC294 Mic pin fixup for Lenovo AIO machines +alc225-wyse + Dell Wyse fixups +alc274-dell-aio + ALC274 fixups on Dell AIO machines +alc255-dummy-lineout + Dell Precision 3930 fixups +alc255-dell-headset"}, + Dell Precision 3630 fixups +alc295-hp-x360 + HP Spectre X360 fixups ALC66x/67x/892 ============== +aspire + Subwoofer pin fixup for Aspire laptops +ideapad + Subwoofer pin fixup for Ideapad laptops mario Chromebook mario model fixup +hp-rp5800 + Headphone pin fixup for HP RP5800 asus-mode1 ASUS asus-mode2 @@ -105,10 +285,40 @@ asus-mode7 ASUS asus-mode8 ASUS +zotac-z68 + Front HP fixup for Zotac Z68 inv-dmic Inverted internal mic workaround +alc662-headset-multi + Dell headset jack, which can also be used as mic-in (ALC662) dell-headset-multi Headset jack, which can also be used as mic-in +alc662-headset + Headset mode support on ALC662 +alc668-headset + Headset mode support on ALC668 +bass16 + Bass speaker fixup on pin 0x16 +bass1a + Bass speaker fixup on pin 0x1a +automute + Auto-mute fixups for ALC668 +dell-xps13 + Dell XPS13 fixups +asus-nx50 + ASUS Nx50 fixups +asus-nx51 + ASUS Nx51 fixups +alc891-headset + Headset mode support on ALC891 +alc891-headset-multi + Dell headset jack, which can also be used as mic-in (ALC891) +acer-veriton + Acer Veriton speaker pin fixup +asrock-mobo + Fix invalid 0x15 / 0x16 pins +usi-headset + Headset support on USI machines dual-codecs Lenovo laptops with dual codecs @@ -116,20 +326,70 @@ ALC680 ====== N/A -ALC88x/898/1150 -====================== +ALC88x/898/1150/1220 +==================== +abit-aw9d + Pin fixups for Abit AW9D-MAX +lenovo-y530 + Pin fixups for Lenovo Y530 +acer-aspire-7736 + Fixup for Acer Aspire 7736 +asus-w90v + Pin fixup for ASUS W90V +cd + Enable audio CD pin NID 0x1c +no-front-hp + Disable front HP pin NID 0x1b +vaio-tt + Pin fixup for VAIO TT +eee1601 + COEF setups for ASUS Eee 1601 +alc882-eapd + Change EAPD COEF mode on ALC882 +alc883-eapd + Change EAPD COEF mode on ALC883 +gpio1 + Enable GPIO1 +gpio2 + Enable GPIO2 +gpio3 + Enable GPIO3 +alc889-coef + Setup ALC889 COEF +asus-w2jc + Fixups for ASUS W2JC acer-aspire-4930g Acer Aspire 4930G/5930G/6530G/6930G/7730G acer-aspire-8930g Acer Aspire 8330G/6935G acer-aspire Acer Aspire others +macpro-gpio + GPIO setup for Mac Pro +dac-route + Workaround for DAC routing on Acer Aspire +mbp-vref + Vref setup for Macbook Pro +imac91-vref + Vref setup for iMac 9,1 +mba11-vref + Vref setup for MacBook Air 1,1 +mba21-vref + Vref setup for MacBook Air 2,1 +mp11-vref + Vref setup for Mac Pro 1,1 +mp41-vref + Vref setup for Mac Pro 4,1 inv-dmic Inverted internal mic workaround no-primary-hp VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC) +asus-bass + Bass speaker setup for ASUS ET2700 dual-codecs ALC1220 dual codecs for Gaming mobos +clevo-p950 + Fixups for Clevo P950 ALC861/660 ========== diff --git a/Documentation/sound/soc/dpcm.rst b/Documentation/sound/soc/dpcm.rst index 395e5a516282..fe61e02277f8 100644 --- a/Documentation/sound/soc/dpcm.rst +++ b/Documentation/sound/soc/dpcm.rst @@ -254,9 +254,7 @@ configuration. channels->min = channels->max = 2; /* set DAI0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/MAINTAINERS b/MAINTAINERS index 3d6e7e7b30d5..634039e7f083 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13585,6 +13585,13 @@ L: linux-block@vger.kernel.org S: Maintained F: drivers/block/skd*[ch] +STI AUDIO (ASoC) DRIVERS +M: Arnaud Pouliquen <arnaud.pouliquen@st.com> +L: alsa-devel@alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +F: sound/soc/sti/ + STI CEC DRIVER M: Benjamin Gaignard <benjamin.gaignard@linaro.org> S: Maintained @@ -13598,6 +13605,14 @@ T: git git://linuxtv.org/media_tree.git S: Maintained F: drivers/media/usb/stk1160/ +STM32 AUDIO (ASoC) DRIVERS +M: Olivier Moysan <olivier.moysan@st.com> +M: Arnaud Pouliquen <arnaud.pouliquen@st.com> +L: alsa-devel@alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/st,stm32-*.txt +F: sound/soc/stm/ + STM32 TIMER/LPTIMER DRIVERS M: Fabrice Gasnier <fabrice.gasnier@st.com> S: Maintained diff --git a/arch/arm/mach-pxa/devices.c b/arch/arm/mach-pxa/devices.c index d7c9a8476d57..5a16ea74e28a 100644 --- a/arch/arm/mach-pxa/devices.c +++ b/arch/arm/mach-pxa/devices.c @@ -4,6 +4,7 @@ #include <linux/init.h> #include <linux/platform_device.h> #include <linux/dma-mapping.h> +#include <linux/dmaengine.h> #include <linux/spi/pxa2xx_spi.h> #include <linux/platform_data/i2c-pxa.h> @@ -59,16 +60,6 @@ static struct resource pxamci_resources[] = { .end = IRQ_MMC, .flags = IORESOURCE_IRQ, }, - [2] = { - .start = 21, - .end = 21, - .flags = IORESOURCE_DMA, - }, - [3] = { - .start = 22, - .end = 22, - .flags = IORESOURCE_DMA, - }, }; static u64 pxamci_dmamask = 0xffffffffUL; @@ -406,16 +397,6 @@ static struct resource pxa_ir_resources[] = { .end = 0x40700023, .flags = IORESOURCE_MEM, }, - [5] = { - .start = 17, - .end = 17, - .flags = IORESOURCE_DMA, - }, - [6] = { - .start = 18, - .end = 18, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa_device_ficp = { @@ -544,18 +525,6 @@ static struct resource pxa25x_resource_ssp[] = { .end = IRQ_SSP, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for RX */ - .start = 13, - .end = 13, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for TX */ - .start = 14, - .end = 14, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa25x_device_ssp = { @@ -582,18 +551,6 @@ static struct resource pxa25x_resource_nssp[] = { .end = IRQ_NSSP, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for RX */ - .start = 15, - .end = 15, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for TX */ - .start = 16, - .end = 16, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa25x_device_nssp = { @@ -620,18 +577,6 @@ static struct resource pxa25x_resource_assp[] = { .end = IRQ_ASSP, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for RX */ - .start = 23, - .end = 23, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for TX */ - .start = 24, - .end = 24, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa25x_device_assp = { @@ -750,18 +695,6 @@ static struct resource pxa27x_resource_ssp1[] = { .end = IRQ_SSP, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for RX */ - .start = 13, - .end = 13, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for TX */ - .start = 14, - .end = 14, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa27x_device_ssp1 = { @@ -788,18 +721,6 @@ static struct resource pxa27x_resource_ssp2[] = { .end = IRQ_SSP2, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for RX */ - .start = 15, - .end = 15, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for TX */ - .start = 16, - .end = 16, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa27x_device_ssp2 = { @@ -826,18 +747,6 @@ static struct resource pxa27x_resource_ssp3[] = { .end = IRQ_SSP3, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for RX */ - .start = 66, - .end = 66, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for TX */ - .start = 67, - .end = 67, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa27x_device_ssp3 = { @@ -894,16 +803,6 @@ static struct resource pxa3xx_resources_mci2[] = { .end = IRQ_MMC2, .flags = IORESOURCE_IRQ, }, - [2] = { - .start = 93, - .end = 93, - .flags = IORESOURCE_DMA, - }, - [3] = { - .start = 94, - .end = 94, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa3xx_device_mci2 = { @@ -933,16 +832,6 @@ static struct resource pxa3xx_resources_mci3[] = { .end = IRQ_MMC3, .flags = IORESOURCE_IRQ, }, - [2] = { - .start = 100, - .end = 100, - .flags = IORESOURCE_DMA, - }, - [3] = { - .start = 101, - .end = 101, - .flags = IORESOURCE_DMA, - }, }; struct platform_device pxa3xx_device_mci3 = { @@ -1020,18 +909,6 @@ static struct resource pxa3xx_resources_nand[] = { .end = IRQ_NAND, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for Data DMA */ - .start = 97, - .end = 97, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for Command DMA */ - .start = 99, - .end = 99, - .flags = IORESOURCE_DMA, - }, }; static u64 pxa3xx_nand_dma_mask = DMA_BIT_MASK(32); @@ -1065,18 +942,6 @@ static struct resource pxa3xx_resource_ssp4[] = { .end = IRQ_SSP4, .flags = IORESOURCE_IRQ, }, - [2] = { - /* DRCMR for RX */ - .start = 2, - .end = 2, - .flags = IORESOURCE_DMA, - }, - [3] = { - /* DRCMR for TX */ - .start = 3, - .end = 3, - .flags = IORESOURCE_DMA, - }, }; /* @@ -1202,11 +1067,6 @@ void __init pxa2xx_set_spi_info(unsigned id, struct pxa2xx_spi_master *info) platform_device_add(pd); } -static struct mmp_dma_platdata pxa_dma_pdata = { - .dma_channels = 0, - .nb_requestors = 0, -}; - static struct resource pxa_dma_resource[] = { [0] = { .start = 0x40000000, @@ -1233,9 +1093,7 @@ static struct platform_device pxa2xx_pxa_dma = { .resource = pxa_dma_resource, }; -void __init pxa2xx_set_dmac_info(int nb_channels, int nb_requestors) +void __init pxa2xx_set_dmac_info(struct mmp_dma_platdata *dma_pdata) { - pxa_dma_pdata.dma_channels = nb_channels; - pxa_dma_pdata.nb_requestors = nb_requestors; - pxa_register_device(&pxa2xx_pxa_dma, &pxa_dma_pdata); + pxa_register_device(&pxa2xx_pxa_dma, dma_pdata); } diff --git a/arch/arm/mach-pxa/devices.h b/arch/arm/mach-pxa/devices.h index 11263f7c455b..498b07bc6a3e 100644 --- a/arch/arm/mach-pxa/devices.h +++ b/arch/arm/mach-pxa/devices.h @@ -1,4 +1,8 @@ /* SPDX-License-Identifier: GPL-2.0 */ +#define PDMA_FILTER_PARAM(_prio, _requestor) (&(struct pxad_param) { \ + .prio = PXAD_PRIO_##_prio, .drcmr = _requestor }) +struct mmp_dma_platdata; + extern struct platform_device pxa_device_pmu; extern struct platform_device pxa_device_mci; extern struct platform_device pxa3xx_device_mci2; @@ -55,7 +59,7 @@ extern struct platform_device pxa3xx_device_gpio; extern struct platform_device pxa93x_device_gpio; void __init pxa_register_device(struct platform_device *dev, void *data); -void __init pxa2xx_set_dmac_info(int nb_channels, int nb_requestors); +void __init pxa2xx_set_dmac_info(struct mmp_dma_platdata *dma_pdata); struct i2c_pxa_platform_data; extern void pxa_set_i2c_info(struct i2c_pxa_platform_data *info); diff --git a/arch/arm/mach-pxa/pxa25x.c b/arch/arm/mach-pxa/pxa25x.c index ba431fad5c47..ab8808ce7e21 100644 --- a/arch/arm/mach-pxa/pxa25x.c +++ b/arch/arm/mach-pxa/pxa25x.c @@ -16,6 +16,8 @@ * initialization stuff for PXA machines which can be overridden later if * need be. */ +#include <linux/dmaengine.h> +#include <linux/dma/pxa-dma.h> #include <linux/gpio.h> #include <linux/gpio-pxa.h> #include <linux/module.h> @@ -26,6 +28,7 @@ #include <linux/syscore_ops.h> #include <linux/irq.h> #include <linux/irqchip.h> +#include <linux/platform_data/mmp_dma.h> #include <asm/mach/map.h> #include <asm/suspend.h> @@ -201,6 +204,39 @@ static struct platform_device *pxa25x_devices[] __initdata = { &pxa_device_asoc_platform, }; +static const struct dma_slave_map pxa25x_slave_map[] = { + /* PXA25x, PXA27x and PXA3xx common entries */ + { "pxa2xx-ac97", "pcm_pcm_mic_mono", PDMA_FILTER_PARAM(LOWEST, 8) }, + { "pxa2xx-ac97", "pcm_pcm_aux_mono_in", PDMA_FILTER_PARAM(LOWEST, 9) }, + { "pxa2xx-ac97", "pcm_pcm_aux_mono_out", + PDMA_FILTER_PARAM(LOWEST, 10) }, + { "pxa2xx-ac97", "pcm_pcm_stereo_in", PDMA_FILTER_PARAM(LOWEST, 11) }, + { "pxa2xx-ac97", "pcm_pcm_stereo_out", PDMA_FILTER_PARAM(LOWEST, 12) }, + { "pxa-ssp-dai.1", "rx", PDMA_FILTER_PARAM(LOWEST, 13) }, + { "pxa-ssp-dai.1", "tx", PDMA_FILTER_PARAM(LOWEST, 14) }, + { "pxa-ssp-dai.2", "rx", PDMA_FILTER_PARAM(LOWEST, 15) }, + { "pxa-ssp-dai.2", "tx", PDMA_FILTER_PARAM(LOWEST, 16) }, + { "pxa2xx-ir", "rx", PDMA_FILTER_PARAM(LOWEST, 17) }, + { "pxa2xx-ir", "tx", PDMA_FILTER_PARAM(LOWEST, 18) }, + { "pxa2xx-mci.0", "rx", PDMA_FILTER_PARAM(LOWEST, 21) }, + { "pxa2xx-mci.0", "tx", PDMA_FILTER_PARAM(LOWEST, 22) }, + + /* PXA25x specific map */ + { "pxa25x-ssp.0", "rx", PDMA_FILTER_PARAM(LOWEST, 13) }, + { "pxa25x-ssp.0", "tx", PDMA_FILTER_PARAM(LOWEST, 14) }, + { "pxa25x-nssp.1", "rx", PDMA_FILTER_PARAM(LOWEST, 15) }, + { "pxa25x-nssp.1", "tx", PDMA_FILTER_PARAM(LOWEST, 16) }, + { "pxa25x-nssp.2", "rx", PDMA_FILTER_PARAM(LOWEST, 23) }, + { "pxa25x-nssp.2", "tx", PDMA_FILTER_PARAM(LOWEST, 24) }, +}; + +static struct mmp_dma_platdata pxa25x_dma_pdata = { + .dma_channels = 16, + .nb_requestors = 40, + .slave_map = pxa25x_slave_map, + .slave_map_cnt = ARRAY_SIZE(pxa25x_slave_map), +}; + static int __init pxa25x_init(void) { int ret = 0; @@ -215,7 +251,7 @@ static int __init pxa25x_init(void) register_syscore_ops(&pxa2xx_mfp_syscore_ops); if (!of_have_populated_dt()) { - pxa2xx_set_dmac_info(16, 40); + pxa2xx_set_dmac_info(&pxa25x_dma_pdata); pxa_register_device(&pxa25x_device_gpio, &pxa25x_gpio_info); ret = platform_add_devices(pxa25x_devices, ARRAY_SIZE(pxa25x_devices)); diff --git a/arch/arm/mach-pxa/pxa27x.c b/arch/arm/mach-pxa/pxa27x.c index 0c06f383ad52..5a8990a9313d 100644 --- a/arch/arm/mach-pxa/pxa27x.c +++ b/arch/arm/mach-pxa/pxa27x.c @@ -11,6 +11,8 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ +#include <linux/dmaengine.h> +#include <linux/dma/pxa-dma.h> #include <linux/gpio.h> #include <linux/gpio-pxa.h> #include <linux/module.h> @@ -23,6 +25,7 @@ #include <linux/io.h> #include <linux/irq.h> #include <linux/platform_data/i2c-pxa.h> +#include <linux/platform_data/mmp_dma.h> #include <asm/mach/map.h> #include <mach/hardware.h> @@ -297,6 +300,40 @@ static struct platform_device *devices[] __initdata = { &pxa27x_device_pwm1, }; +static const struct dma_slave_map pxa27x_slave_map[] = { + /* PXA25x, PXA27x and PXA3xx common entries */ + { "pxa2xx-ac97", "pcm_pcm_mic_mono", PDMA_FILTER_PARAM(LOWEST, 8) }, + { "pxa2xx-ac97", "pcm_pcm_aux_mono_in", PDMA_FILTER_PARAM(LOWEST, 9) }, + { "pxa2xx-ac97", "pcm_pcm_aux_mono_out", + PDMA_FILTER_PARAM(LOWEST, 10) }, + { "pxa2xx-ac97", "pcm_pcm_stereo_in", PDMA_FILTER_PARAM(LOWEST, 11) }, + { "pxa2xx-ac97", "pcm_pcm_stereo_out", PDMA_FILTER_PARAM(LOWEST, 12) }, + { "pxa-ssp-dai.0", "rx", PDMA_FILTER_PARAM(LOWEST, 13) }, + { "pxa-ssp-dai.0", "tx", PDMA_FILTER_PARAM(LOWEST, 14) }, + { "pxa-ssp-dai.1", "rx", PDMA_FILTER_PARAM(LOWEST, 15) }, + { "pxa-ssp-dai.1", "tx", PDMA_FILTER_PARAM(LOWEST, 16) }, + { "pxa2xx-ir", "rx", PDMA_FILTER_PARAM(LOWEST, 17) }, + { "pxa2xx-ir", "tx", PDMA_FILTER_PARAM(LOWEST, 18) }, + { "pxa2xx-mci.0", "rx", PDMA_FILTER_PARAM(LOWEST, 21) }, + { "pxa2xx-mci.0", "tx", PDMA_FILTER_PARAM(LOWEST, 22) }, + { "pxa-ssp-dai.2", "rx", PDMA_FILTER_PARAM(LOWEST, 66) }, + { "pxa-ssp-dai.2", "tx", PDMA_FILTER_PARAM(LOWEST, 67) }, + + /* PXA27x specific map */ + { "pxa2xx-i2s", "rx", PDMA_FILTER_PARAM(LOWEST, 2) }, + { "pxa2xx-i2s", "tx", PDMA_FILTER_PARAM(LOWEST, 3) }, + { "pxa27x-camera.0", "CI_Y", PDMA_FILTER_PARAM(HIGHEST, 68) }, + { "pxa27x-camera.0", "CI_U", PDMA_FILTER_PARAM(HIGHEST, 69) }, + { "pxa27x-camera.0", "CI_V", PDMA_FILTER_PARAM(HIGHEST, 70) }, +}; + +static struct mmp_dma_platdata pxa27x_dma_pdata = { + .dma_channels = 32, + .nb_requestors = 75, + .slave_map = pxa27x_slave_map, + .slave_map_cnt = ARRAY_SIZE(pxa27x_slave_map), +}; + static int __init pxa27x_init(void) { int ret = 0; @@ -313,7 +350,7 @@ static int __init pxa27x_init(void) if (!of_have_populated_dt()) { pxa_register_device(&pxa27x_device_gpio, &pxa27x_gpio_info); - pxa2xx_set_dmac_info(32, 75); + pxa2xx_set_dmac_info(&pxa27x_dma_pdata); ret = platform_add_devices(devices, ARRAY_SIZE(devices)); } diff --git a/arch/arm/mach-pxa/pxa3xx.c b/arch/arm/mach-pxa/pxa3xx.c index 8c64f93b669b..df9c8970adcf 100644 --- a/arch/arm/mach-pxa/pxa3xx.c +++ b/arch/arm/mach-pxa/pxa3xx.c @@ -12,6 +12,8 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ +#include <linux/dmaengine.h> +#include <linux/dma/pxa-dma.h> #include <linux/module.h> #include <linux/kernel.h> #include <linux/init.h> @@ -24,6 +26,7 @@ #include <linux/of.h> #include <linux/syscore_ops.h> #include <linux/platform_data/i2c-pxa.h> +#include <linux/platform_data/mmp_dma.h> #include <asm/mach/map.h> #include <asm/suspend.h> @@ -421,6 +424,42 @@ static struct platform_device *devices[] __initdata = { &pxa27x_device_pwm1, }; +static const struct dma_slave_map pxa3xx_slave_map[] = { + /* PXA25x, PXA27x and PXA3xx common entries */ + { "pxa2xx-ac97", "pcm_pcm_mic_mono", PDMA_FILTER_PARAM(LOWEST, 8) }, + { "pxa2xx-ac97", "pcm_pcm_aux_mono_in", PDMA_FILTER_PARAM(LOWEST, 9) }, + { "pxa2xx-ac97", "pcm_pcm_aux_mono_out", + PDMA_FILTER_PARAM(LOWEST, 10) }, + { "pxa2xx-ac97", "pcm_pcm_stereo_in", PDMA_FILTER_PARAM(LOWEST, 11) }, + { "pxa2xx-ac97", "pcm_pcm_stereo_out", PDMA_FILTER_PARAM(LOWEST, 12) }, + { "pxa-ssp-dai.0", "rx", PDMA_FILTER_PARAM(LOWEST, 13) }, + { "pxa-ssp-dai.0", "tx", PDMA_FILTER_PARAM(LOWEST, 14) }, + { "pxa-ssp-dai.1", "rx", PDMA_FILTER_PARAM(LOWEST, 15) }, + { "pxa-ssp-dai.1", "tx", PDMA_FILTER_PARAM(LOWEST, 16) }, + { "pxa2xx-ir", "rx", PDMA_FILTER_PARAM(LOWEST, 17) }, + { "pxa2xx-ir", "tx", PDMA_FILTER_PARAM(LOWEST, 18) }, + { "pxa2xx-mci.0", "rx", PDMA_FILTER_PARAM(LOWEST, 21) }, + { "pxa2xx-mci.0", "tx", PDMA_FILTER_PARAM(LOWEST, 22) }, + { "pxa-ssp-dai.2", "rx", PDMA_FILTER_PARAM(LOWEST, 66) }, + { "pxa-ssp-dai.2", "tx", PDMA_FILTER_PARAM(LOWEST, 67) }, + + /* PXA3xx specific map */ + { "pxa-ssp-dai.3", "rx", PDMA_FILTER_PARAM(LOWEST, 2) }, + { "pxa-ssp-dai.3", "tx", PDMA_FILTER_PARAM(LOWEST, 3) }, + { "pxa2xx-mci.1", "rx", PDMA_FILTER_PARAM(LOWEST, 93) }, + { "pxa2xx-mci.1", "tx", PDMA_FILTER_PARAM(LOWEST, 94) }, + { "pxa3xx-nand", "data", PDMA_FILTER_PARAM(LOWEST, 97) }, + { "pxa2xx-mci.2", "rx", PDMA_FILTER_PARAM(LOWEST, 100) }, + { "pxa2xx-mci.2", "tx", PDMA_FILTER_PARAM(LOWEST, 101) }, +}; + +static struct mmp_dma_platdata pxa3xx_dma_pdata = { + .dma_channels = 32, + .nb_requestors = 100, + .slave_map = pxa3xx_slave_map, + .slave_map_cnt = ARRAY_SIZE(pxa3xx_slave_map), +}; + static int __init pxa3xx_init(void) { int ret = 0; @@ -456,7 +495,7 @@ static int __init pxa3xx_init(void) if (of_have_populated_dt()) return 0; - pxa2xx_set_dmac_info(32, 100); + pxa2xx_set_dmac_info(&pxa3xx_dma_pdata); ret = platform_add_devices(devices, ARRAY_SIZE(devices)); if (ret) return ret; diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c index ba13f793fbce..ed36dcab80f1 100644 --- a/arch/arm/plat-pxa/ssp.c +++ b/arch/arm/plat-pxa/ssp.c @@ -127,53 +127,6 @@ static int pxa_ssp_probe(struct platform_device *pdev) if (IS_ERR(ssp->clk)) return PTR_ERR(ssp->clk); - if (dev->of_node) { - struct of_phandle_args dma_spec; - struct device_node *np = dev->of_node; - int ret; - - /* - * FIXME: we should allocate the DMA channel from this - * context and pass the channel down to the ssp users. - * For now, we lookup the rx and tx indices manually - */ - - /* rx */ - ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", - 0, &dma_spec); - - if (ret) { - dev_err(dev, "Can't parse dmas property\n"); - return -ENODEV; - } - ssp->drcmr_rx = dma_spec.args[0]; - of_node_put(dma_spec.np); - - /* tx */ - ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", - 1, &dma_spec); - if (ret) { - dev_err(dev, "Can't parse dmas property\n"); - return -ENODEV; - } - ssp->drcmr_tx = dma_spec.args[0]; - of_node_put(dma_spec.np); - } else { - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (res == NULL) { - dev_err(dev, "no SSP RX DRCMR defined\n"); - return -ENODEV; - } - ssp->drcmr_rx = res->start; - - res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (res == NULL) { - dev_err(dev, "no SSP TX DRCMR defined\n"); - return -ENODEV; - } - ssp->drcmr_tx = res->start; - } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (res == NULL) { dev_err(dev, "no memory resource defined\n"); diff --git a/drivers/ata/pata_pxa.c b/drivers/ata/pata_pxa.c index f6c46e9a4dc0..e8b6a2e464c9 100644 --- a/drivers/ata/pata_pxa.c +++ b/drivers/ata/pata_pxa.c @@ -25,7 +25,6 @@ #include <linux/libata.h> #include <linux/platform_device.h> #include <linux/dmaengine.h> -#include <linux/dma/pxa-dma.h> #include <linux/gpio.h> #include <linux/slab.h> #include <linux/completion.h> @@ -180,8 +179,6 @@ static int pxa_ata_probe(struct platform_device *pdev) struct resource *irq_res; struct pata_pxa_pdata *pdata = dev_get_platdata(&pdev->dev); struct dma_slave_config config; - dma_cap_mask_t mask; - struct pxad_param param; int ret = 0; /* @@ -278,10 +275,6 @@ static int pxa_ata_probe(struct platform_device *pdev) ap->private_data = data; - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - param.prio = PXAD_PRIO_LOWEST; - param.drcmr = pdata->dma_dreq; memset(&config, 0, sizeof(config)); config.src_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; config.dst_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; @@ -294,8 +287,7 @@ static int pxa_ata_probe(struct platform_device *pdev) * Request the DMA channel */ data->dma_chan = - dma_request_slave_channel_compat(mask, pxad_filter_fn, - ¶m, &pdev->dev, "data"); + dma_request_slave_channel(&pdev->dev, "data"); if (!data->dma_chan) return -EBUSY; ret = dmaengine_slave_config(data->dma_chan, &config); diff --git a/drivers/clk/clk.c b/drivers/clk/clk.c index e2ed078abd90..976f59e11f9a 100644 --- a/drivers/clk/clk.c +++ b/drivers/clk/clk.c @@ -67,6 +67,7 @@ struct clk_core { unsigned long max_rate; unsigned long accuracy; int phase; + struct clk_duty duty; struct hlist_head children; struct hlist_node child_node; struct hlist_head clks; @@ -2401,6 +2402,172 @@ int clk_get_phase(struct clk *clk) } EXPORT_SYMBOL_GPL(clk_get_phase); +static void clk_core_reset_duty_cycle_nolock(struct clk_core *core) +{ + /* Assume a default value of 50% */ + core->duty.num = 1; + core->duty.den = 2; +} + +static int clk_core_update_duty_cycle_parent_nolock(struct clk_core *core); + +static int clk_core_update_duty_cycle_nolock(struct clk_core *core) +{ + struct clk_duty *duty = &core->duty; + int ret = 0; + + if (!core->ops->get_duty_cycle) + return clk_core_update_duty_cycle_parent_nolock(core); + + ret = core->ops->get_duty_cycle(core->hw, duty); + if (ret) + goto reset; + + /* Don't trust the clock provider too much */ + if (duty->den == 0 || duty->num > duty->den) { + ret = -EINVAL; + goto reset; + } + + return 0; + +reset: + clk_core_reset_duty_cycle_nolock(core); + return ret; +} + +static int clk_core_update_duty_cycle_parent_nolock(struct clk_core *core) +{ + int ret = 0; + + if (core->parent && + core->flags & CLK_DUTY_CYCLE_PARENT) { + ret = clk_core_update_duty_cycle_nolock(core->parent); + memcpy(&core->duty, &core->parent->duty, sizeof(core->duty)); + } else { + clk_core_reset_duty_cycle_nolock(core); + } + + return ret; +} + +static int clk_core_set_duty_cycle_parent_nolock(struct clk_core *core, + struct clk_duty *duty); + +static int clk_core_set_duty_cycle_nolock(struct clk_core *core, + struct clk_duty *duty) +{ + int ret; + + lockdep_assert_held(&prepare_lock); + + if (clk_core_rate_is_protected(core)) + return -EBUSY; + + trace_clk_set_duty_cycle(core, duty); + + if (!core->ops->set_duty_cycle) + return clk_core_set_duty_cycle_parent_nolock(core, duty); + + ret = core->ops->set_duty_cycle(core->hw, duty); + if (!ret) + memcpy(&core->duty, duty, sizeof(*duty)); + + trace_clk_set_duty_cycle_complete(core, duty); + + return ret; +} + +static int clk_core_set_duty_cycle_parent_nolock(struct clk_core *core, + struct clk_duty *duty) +{ + int ret = 0; + + if (core->parent && + core->flags & (CLK_DUTY_CYCLE_PARENT | CLK_SET_RATE_PARENT)) { + ret = clk_core_set_duty_cycle_nolock(core->parent, duty); + memcpy(&core->duty, &core->parent->duty, sizeof(core->duty)); + } + + return ret; +} + +/** + * clk_set_duty_cycle - adjust the duty cycle ratio of a clock signal + * @clk: clock signal source + * @num: numerator of the duty cycle ratio to be applied + * @den: denominator of the duty cycle ratio to be applied + * + * Apply the duty cycle ratio if the ratio is valid and the clock can + * perform this operation + * + * Returns (0) on success, a negative errno otherwise. + */ +int clk_set_duty_cycle(struct clk *clk, unsigned int num, unsigned int den) +{ + int ret; + struct clk_duty duty; + + if (!clk) + return 0; + + /* sanity check the ratio */ + if (den == 0 || num > den) + return -EINVAL; + + duty.num = num; + duty.den = den; + + clk_prepare_lock(); + + if (clk->exclusive_count) + clk_core_rate_unprotect(clk->core); + + ret = clk_core_set_duty_cycle_nolock(clk->core, &duty); + + if (clk->exclusive_count) + clk_core_rate_protect(clk->core); + + clk_prepare_unlock(); + + return ret; +} +EXPORT_SYMBOL_GPL(clk_set_duty_cycle); + +static int clk_core_get_scaled_duty_cycle(struct clk_core *core, + unsigned int scale) +{ + struct clk_duty *duty = &core->duty; + int ret; + + clk_prepare_lock(); + + ret = clk_core_update_duty_cycle_nolock(core); + if (!ret) + ret = mult_frac(scale, duty->num, duty->den); + + clk_prepare_unlock(); + + return ret; +} + +/** + * clk_get_scaled_duty_cycle - return the duty cycle ratio of a clock signal + * @clk: clock signal source + * @scale: scaling factor to be applied to represent the ratio as an integer + * + * Returns the duty cycle ratio of a clock node multiplied by the provided + * scaling factor, or negative errno on error. + */ +int clk_get_scaled_duty_cycle(struct clk *clk, unsigned int scale) +{ + if (!clk) + return 0; + + return clk_core_get_scaled_duty_cycle(clk->core, scale); +} +EXPORT_SYMBOL_GPL(clk_get_scaled_duty_cycle); + /** * clk_is_match - check if two clk's point to the same hardware clock * @p: clk compared against q @@ -2454,12 +2621,13 @@ static void clk_summary_show_one(struct seq_file *s, struct clk_core *c, if (!c) return; - seq_printf(s, "%*s%-*s %7d %8d %8d %11lu %10lu %-3d\n", + seq_printf(s, "%*s%-*s %7d %8d %8d %11lu %10lu %5d %6d\n", level * 3 + 1, "", 30 - level * 3, c->name, c->enable_count, c->prepare_count, c->protect_count, clk_core_get_rate(c), clk_core_get_accuracy(c), - clk_core_get_phase(c)); + clk_core_get_phase(c), + clk_core_get_scaled_duty_cycle(c, 100000)); } static void clk_summary_show_subtree(struct seq_file *s, struct clk_core *c, @@ -2481,9 +2649,9 @@ static int clk_summary_show(struct seq_file *s, void *data) struct clk_core *c; struct hlist_head **lists = (struct hlist_head **)s->private; - seq_puts(s, " enable prepare protect \n"); - seq_puts(s, " clock count count count rate accuracy phase\n"); - seq_puts(s, "----------------------------------------------------------------------------------------\n"); + seq_puts(s, " enable prepare protect duty\n"); + seq_puts(s, " clock count count count rate accuracy phase cycle\n"); + seq_puts(s, "---------------------------------------------------------------------------------------------\n"); clk_prepare_lock(); @@ -2510,6 +2678,8 @@ static void clk_dump_one(struct seq_file *s, struct clk_core *c, int level) seq_printf(s, "\"rate\": %lu,", clk_core_get_rate(c)); seq_printf(s, "\"accuracy\": %lu,", clk_core_get_accuracy(c)); seq_printf(s, "\"phase\": %d", clk_core_get_phase(c)); + seq_printf(s, "\"duty_cycle\": %u", + clk_core_get_scaled_duty_cycle(c, 100000)); } static void clk_dump_subtree(struct seq_file *s, struct clk_core *c, int level) @@ -2571,6 +2741,7 @@ static const struct { ENTRY(CLK_SET_RATE_UNGATE), ENTRY(CLK_IS_CRITICAL), ENTRY(CLK_OPS_PARENT_ENABLE), + ENTRY(CLK_DUTY_CYCLE_PARENT), #undef ENTRY }; @@ -2609,6 +2780,17 @@ static int possible_parents_show(struct seq_file *s, void *data) } DEFINE_SHOW_ATTRIBUTE(possible_parents); +static int clk_duty_cycle_show(struct seq_file *s, void *data) +{ + struct clk_core *core = s->private; + struct clk_duty *duty = &core->duty; + + seq_printf(s, "%u/%u\n", duty->num, duty->den); + + return 0; +} +DEFINE_SHOW_ATTRIBUTE(clk_duty_cycle); + static void clk_debug_create_one(struct clk_core *core, struct dentry *pdentry) { struct dentry *root; @@ -2627,6 +2809,8 @@ static void clk_debug_create_one(struct clk_core *core, struct dentry *pdentry) debugfs_create_u32("clk_enable_count", 0444, root, &core->enable_count); debugfs_create_u32("clk_protect_count", 0444, root, &core->protect_count); debugfs_create_u32("clk_notifier_count", 0444, root, &core->notifier_count); + debugfs_create_file("clk_duty_cycle", 0444, root, core, + &clk_duty_cycle_fops); if (core->num_parents > 1) debugfs_create_file("clk_possible_parents", 0444, root, core, @@ -2845,6 +3029,11 @@ static int __clk_core_init(struct clk_core *core) core->phase = 0; /* + * Set clk's duty cycle. + */ + clk_core_update_duty_cycle_nolock(core); + + /* * Set clk's rate. The preferred method is to use .recalc_rate. For * simple clocks and lazy developers the default fallback is to use the * parent's rate. If a clock doesn't have a parent (or is orphaned) diff --git a/drivers/dma/pxa_dma.c b/drivers/dma/pxa_dma.c index b53fb618bbf6..b31c28b67ad3 100644 --- a/drivers/dma/pxa_dma.c +++ b/drivers/dma/pxa_dma.c @@ -179,6 +179,8 @@ static unsigned int pxad_drcmr(unsigned int line) return 0x1000 + line * 4; } +bool pxad_filter_fn(struct dma_chan *chan, void *param); + /* * Debug fs */ @@ -760,6 +762,8 @@ static void pxad_free_chan_resources(struct dma_chan *dchan) dma_pool_destroy(chan->desc_pool); chan->desc_pool = NULL; + chan->drcmr = U32_MAX; + chan->prio = PXAD_PRIO_LOWEST; } static void pxad_free_desc(struct virt_dma_desc *vd) @@ -1384,6 +1388,9 @@ static int pxad_init_dmadev(struct platform_device *op, c = devm_kzalloc(&op->dev, sizeof(*c), GFP_KERNEL); if (!c) return -ENOMEM; + + c->drcmr = U32_MAX; + c->prio = PXAD_PRIO_LOWEST; c->vc.desc_free = pxad_free_desc; vchan_init(&c->vc, &pdev->slave); init_waitqueue_head(&c->wq_state); @@ -1396,9 +1403,10 @@ static int pxad_probe(struct platform_device *op) { struct pxad_device *pdev; const struct of_device_id *of_id; + const struct dma_slave_map *slave_map = NULL; struct mmp_dma_platdata *pdata = dev_get_platdata(&op->dev); struct resource *iores; - int ret, dma_channels = 0, nb_requestors = 0; + int ret, dma_channels = 0, nb_requestors = 0, slave_map_cnt = 0; const enum dma_slave_buswidth widths = DMA_SLAVE_BUSWIDTH_1_BYTE | DMA_SLAVE_BUSWIDTH_2_BYTES | DMA_SLAVE_BUSWIDTH_4_BYTES; @@ -1429,6 +1437,8 @@ static int pxad_probe(struct platform_device *op) } else if (pdata && pdata->dma_channels) { dma_channels = pdata->dma_channels; nb_requestors = pdata->nb_requestors; + slave_map = pdata->slave_map; + slave_map_cnt = pdata->slave_map_cnt; } else { dma_channels = 32; /* default 32 channel */ } @@ -1440,6 +1450,9 @@ static int pxad_probe(struct platform_device *op) pdev->slave.device_prep_dma_memcpy = pxad_prep_memcpy; pdev->slave.device_prep_slave_sg = pxad_prep_slave_sg; pdev->slave.device_prep_dma_cyclic = pxad_prep_dma_cyclic; + pdev->slave.filter.map = slave_map; + pdev->slave.filter.mapcnt = slave_map_cnt; + pdev->slave.filter.fn = pxad_filter_fn; pdev->slave.copy_align = PDMA_ALIGNMENT; pdev->slave.src_addr_widths = widths; diff --git a/drivers/gpu/drm/i915/Kconfig b/drivers/gpu/drm/i915/Kconfig index dfd95889f4b7..5c607f2c707b 100644 --- a/drivers/gpu/drm/i915/Kconfig +++ b/drivers/gpu/drm/i915/Kconfig @@ -23,6 +23,7 @@ config DRM_I915 select SYNC_FILE select IOSF_MBI select CRC32 + select SND_HDA_I915 if SND_HDA_CORE help Choose this option if you have a system that has "Intel Graphics Media Accelerator" or "HD Graphics" integrated graphics, diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index 3ea566f99450..7dd5605d94ae 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -639,11 +639,12 @@ void intel_audio_codec_enable(struct intel_encoder *encoder, dev_priv->av_enc_map[pipe] = encoder; mutex_unlock(&dev_priv->av_mutex); - if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) { + if (acomp && acomp->base.audio_ops && + acomp->base.audio_ops->pin_eld_notify) { /* audio drivers expect pipe = -1 to indicate Non-MST cases */ if (!intel_crtc_has_type(crtc_state, INTEL_OUTPUT_DP_MST)) pipe = -1; - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, + acomp->base.audio_ops->pin_eld_notify(acomp->base.audio_ops->audio_ptr, (int) port, (int) pipe); } @@ -681,11 +682,12 @@ void intel_audio_codec_disable(struct intel_encoder *encoder, dev_priv->av_enc_map[pipe] = NULL; mutex_unlock(&dev_priv->av_mutex); - if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) { + if (acomp && acomp->base.audio_ops && + acomp->base.audio_ops->pin_eld_notify) { /* audio drivers expect pipe = -1 to indicate Non-MST cases */ if (!intel_crtc_has_type(old_crtc_state, INTEL_OUTPUT_DP_MST)) pipe = -1; - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, + acomp->base.audio_ops->pin_eld_notify(acomp->base.audio_ops->audio_ptr, (int) port, (int) pipe); } @@ -880,7 +882,7 @@ static int i915_audio_component_get_eld(struct device *kdev, int port, return ret; } -static const struct i915_audio_component_ops i915_audio_component_ops = { +static const struct drm_audio_component_ops i915_audio_component_ops = { .owner = THIS_MODULE, .get_power = i915_audio_component_get_power, .put_power = i915_audio_component_put_power, @@ -897,12 +899,12 @@ static int i915_audio_component_bind(struct device *i915_kdev, struct drm_i915_private *dev_priv = kdev_to_i915(i915_kdev); int i; - if (WARN_ON(acomp->ops || acomp->dev)) + if (WARN_ON(acomp->base.ops || acomp->base.dev)) return -EEXIST; drm_modeset_lock_all(&dev_priv->drm); - acomp->ops = &i915_audio_component_ops; - acomp->dev = i915_kdev; + acomp->base.ops = &i915_audio_component_ops; + acomp->base.dev = i915_kdev; BUILD_BUG_ON(MAX_PORTS != I915_MAX_PORTS); for (i = 0; i < ARRAY_SIZE(acomp->aud_sample_rate); i++) acomp->aud_sample_rate[i] = 0; @@ -919,8 +921,8 @@ static void i915_audio_component_unbind(struct device *i915_kdev, struct drm_i915_private *dev_priv = kdev_to_i915(i915_kdev); drm_modeset_lock_all(&dev_priv->drm); - acomp->ops = NULL; - acomp->dev = NULL; + acomp->base.ops = NULL; + acomp->base.dev = NULL; dev_priv->audio_component = NULL; drm_modeset_unlock_all(&dev_priv->drm); } diff --git a/drivers/gpu/vga/vga_switcheroo.c b/drivers/gpu/vga/vga_switcheroo.c index fc4adf3d34e8..a96bf46bc483 100644 --- a/drivers/gpu/vga/vga_switcheroo.c +++ b/drivers/gpu/vga/vga_switcheroo.c @@ -103,9 +103,11 @@ * runtime pm. If true, writing ON and OFF to the vga_switcheroo debugfs * interface is a no-op so as not to interfere with runtime pm * @list: client list + * @vga_dev: pci device, indicate which GPU is bound to current audio client * * Registered client. A client can be either a GPU or an audio device on a GPU. - * For audio clients, the @fb_info and @active members are bogus. + * For audio clients, the @fb_info and @active members are bogus. For GPU + * clients, the @vga_dev is bogus. */ struct vga_switcheroo_client { struct pci_dev *pdev; @@ -116,6 +118,7 @@ struct vga_switcheroo_client { bool active; bool driver_power_control; struct list_head list; + struct pci_dev *vga_dev; }; /* @@ -161,9 +164,8 @@ struct vgasr_priv { }; #define ID_BIT_AUDIO 0x100 -#define client_is_audio(c) ((c)->id & ID_BIT_AUDIO) -#define client_is_vga(c) ((c)->id == VGA_SWITCHEROO_UNKNOWN_ID || \ - !client_is_audio(c)) +#define client_is_audio(c) ((c)->id & ID_BIT_AUDIO) +#define client_is_vga(c) (!client_is_audio(c)) #define client_id(c) ((c)->id & ~ID_BIT_AUDIO) static int vga_switcheroo_debugfs_init(struct vgasr_priv *priv); @@ -192,14 +194,29 @@ static void vga_switcheroo_enable(void) vgasr_priv.handler->init(); list_for_each_entry(client, &vgasr_priv.clients, list) { - if (client->id != VGA_SWITCHEROO_UNKNOWN_ID) + if (!client_is_vga(client) || + client_id(client) != VGA_SWITCHEROO_UNKNOWN_ID) continue; + ret = vgasr_priv.handler->get_client_id(client->pdev); if (ret < 0) return; client->id = ret; } + + list_for_each_entry(client, &vgasr_priv.clients, list) { + if (!client_is_audio(client) || + client_id(client) != VGA_SWITCHEROO_UNKNOWN_ID) + continue; + + ret = vgasr_priv.handler->get_client_id(client->vga_dev); + if (ret < 0) + return; + + client->id = ret | ID_BIT_AUDIO; + } + vga_switcheroo_debugfs_init(&vgasr_priv); vgasr_priv.active = true; } @@ -272,7 +289,9 @@ EXPORT_SYMBOL(vga_switcheroo_handler_flags); static int register_client(struct pci_dev *pdev, const struct vga_switcheroo_client_ops *ops, - enum vga_switcheroo_client_id id, bool active, + enum vga_switcheroo_client_id id, + struct pci_dev *vga_dev, + bool active, bool driver_power_control) { struct vga_switcheroo_client *client; @@ -287,6 +306,7 @@ static int register_client(struct pci_dev *pdev, client->id = id; client->active = active; client->driver_power_control = driver_power_control; + client->vga_dev = vga_dev; mutex_lock(&vgasr_mutex); list_add_tail(&client->list, &vgasr_priv.clients); @@ -319,7 +339,7 @@ int vga_switcheroo_register_client(struct pci_dev *pdev, const struct vga_switcheroo_client_ops *ops, bool driver_power_control) { - return register_client(pdev, ops, VGA_SWITCHEROO_UNKNOWN_ID, + return register_client(pdev, ops, VGA_SWITCHEROO_UNKNOWN_ID, NULL, pdev == vga_default_device(), driver_power_control); } @@ -329,19 +349,40 @@ EXPORT_SYMBOL(vga_switcheroo_register_client); * vga_switcheroo_register_audio_client - register audio client * @pdev: client pci device * @ops: client callbacks - * @id: client identifier + * @vga_dev: pci device which is bound to current audio client * * Register audio client (audio device on a GPU). The client is assumed * to use runtime PM. Beforehand, vga_switcheroo_client_probe_defer() * shall be called to ensure that all prerequisites are met. * - * Return: 0 on success, -ENOMEM on memory allocation error. + * Return: 0 on success, -ENOMEM on memory allocation error, -EINVAL on getting + * client id error. */ int vga_switcheroo_register_audio_client(struct pci_dev *pdev, const struct vga_switcheroo_client_ops *ops, - enum vga_switcheroo_client_id id) + struct pci_dev *vga_dev) { - return register_client(pdev, ops, id | ID_BIT_AUDIO, false, true); + enum vga_switcheroo_client_id id = VGA_SWITCHEROO_UNKNOWN_ID; + + /* + * if vga_switcheroo has enabled, that mean two GPU clients and also + * handler are registered. Get audio client id from bound GPU client + * id directly, otherwise, set it as VGA_SWITCHEROO_UNKNOWN_ID, + * it will set to correct id in later when vga_switcheroo_enable() + * is called. + */ + mutex_lock(&vgasr_mutex); + if (vgasr_priv.active) { + id = vgasr_priv.handler->get_client_id(vga_dev); + if (id < 0) { + mutex_unlock(&vgasr_mutex); + return -EINVAL; + } + } + mutex_unlock(&vgasr_mutex); + + return register_client(pdev, ops, id | ID_BIT_AUDIO, vga_dev, + false, true); } EXPORT_SYMBOL(vga_switcheroo_register_audio_client); diff --git a/drivers/media/platform/pxa_camera.c b/drivers/media/platform/pxa_camera.c index d85ffbfb7c1f..b6e9e93bde7a 100644 --- a/drivers/media/platform/pxa_camera.c +++ b/drivers/media/platform/pxa_camera.c @@ -2375,8 +2375,6 @@ static int pxa_camera_probe(struct platform_device *pdev) .src_maxburst = 8, .direction = DMA_DEV_TO_MEM, }; - dma_cap_mask_t mask; - struct pxad_param params; char clk_name[V4L2_CLK_NAME_SIZE]; int irq; int err = 0, i; @@ -2450,34 +2448,20 @@ static int pxa_camera_probe(struct platform_device *pdev) pcdev->base = base; /* request dma */ - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - dma_cap_set(DMA_PRIVATE, mask); - - params.prio = 0; - params.drcmr = 68; - pcdev->dma_chans[0] = - dma_request_slave_channel_compat(mask, pxad_filter_fn, - ¶ms, &pdev->dev, "CI_Y"); + pcdev->dma_chans[0] = dma_request_slave_channel(&pdev->dev, "CI_Y"); if (!pcdev->dma_chans[0]) { dev_err(&pdev->dev, "Can't request DMA for Y\n"); return -ENODEV; } - params.drcmr = 69; - pcdev->dma_chans[1] = - dma_request_slave_channel_compat(mask, pxad_filter_fn, - ¶ms, &pdev->dev, "CI_U"); + pcdev->dma_chans[1] = dma_request_slave_channel(&pdev->dev, "CI_U"); if (!pcdev->dma_chans[1]) { dev_err(&pdev->dev, "Can't request DMA for Y\n"); err = -ENODEV; goto exit_free_dma_y; } - params.drcmr = 70; - pcdev->dma_chans[2] = - dma_request_slave_channel_compat(mask, pxad_filter_fn, - ¶ms, &pdev->dev, "CI_V"); + pcdev->dma_chans[2] = dma_request_slave_channel(&pdev->dev, "CI_V"); if (!pcdev->dma_chans[2]) { dev_err(&pdev->dev, "Can't request DMA for V\n"); err = -ENODEV; diff --git a/drivers/mmc/host/pxamci.c b/drivers/mmc/host/pxamci.c index c763b404510f..6c94474e36f4 100644 --- a/drivers/mmc/host/pxamci.c +++ b/drivers/mmc/host/pxamci.c @@ -24,7 +24,6 @@ #include <linux/interrupt.h> #include <linux/dmaengine.h> #include <linux/dma-mapping.h> -#include <linux/dma/pxa-dma.h> #include <linux/clk.h> #include <linux/err.h> #include <linux/mmc/host.h> @@ -637,10 +636,8 @@ static int pxamci_probe(struct platform_device *pdev) { struct mmc_host *mmc; struct pxamci_host *host = NULL; - struct resource *r, *dmarx, *dmatx; - struct pxad_param param_rx, param_tx; + struct resource *r; int ret, irq, gpio_cd = -1, gpio_ro = -1, gpio_power = -1; - dma_cap_mask_t mask; ret = pxamci_of_init(pdev); if (ret) @@ -739,34 +736,14 @@ static int pxamci_probe(struct platform_device *pdev) platform_set_drvdata(pdev, mmc); - if (!pdev->dev.of_node) { - dmarx = platform_get_resource(pdev, IORESOURCE_DMA, 0); - dmatx = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!dmarx || !dmatx) { - ret = -ENXIO; - goto out; - } - param_rx.prio = PXAD_PRIO_LOWEST; - param_rx.drcmr = dmarx->start; - param_tx.prio = PXAD_PRIO_LOWEST; - param_tx.drcmr = dmatx->start; - } - - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - - host->dma_chan_rx = - dma_request_slave_channel_compat(mask, pxad_filter_fn, - ¶m_rx, &pdev->dev, "rx"); + host->dma_chan_rx = dma_request_slave_channel(&pdev->dev, "rx"); if (host->dma_chan_rx == NULL) { dev_err(&pdev->dev, "unable to request rx dma channel\n"); ret = -ENODEV; goto out; } - host->dma_chan_tx = - dma_request_slave_channel_compat(mask, pxad_filter_fn, - ¶m_tx, &pdev->dev, "tx"); + host->dma_chan_tx = dma_request_slave_channel(&pdev->dev, "tx"); if (host->dma_chan_tx == NULL) { dev_err(&pdev->dev, "unable to request tx dma channel\n"); ret = -ENODEV; diff --git a/drivers/mtd/nand/raw/marvell_nand.c b/drivers/mtd/nand/raw/marvell_nand.c index 218e09431d3d..7af4d6213ee5 100644 --- a/drivers/mtd/nand/raw/marvell_nand.c +++ b/drivers/mtd/nand/raw/marvell_nand.c @@ -2618,8 +2618,6 @@ static int marvell_nfc_init_dma(struct marvell_nfc *nfc) dev); struct dma_slave_config config = {}; struct resource *r; - dma_cap_mask_t mask; - struct pxad_param param; int ret; if (!IS_ENABLED(CONFIG_PXA_DMA)) { @@ -2632,20 +2630,7 @@ static int marvell_nfc_init_dma(struct marvell_nfc *nfc) if (ret) return ret; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) { - dev_err(nfc->dev, "No resource defined for data DMA\n"); - return -ENXIO; - } - - param.drcmr = r->start; - param.prio = PXAD_PRIO_LOWEST; - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - nfc->dma_chan = - dma_request_slave_channel_compat(mask, pxad_filter_fn, - ¶m, nfc->dev, - "data"); + nfc->dma_chan = dma_request_slave_channel(nfc->dev, "data"); if (!nfc->dma_chan) { dev_err(nfc->dev, "Unable to request data DMA channel\n"); diff --git a/drivers/staging/most/sound/sound.c b/drivers/staging/most/sound/sound.c index 04c18323c2ea..89b02fc305b8 100644 --- a/drivers/staging/most/sound/sound.c +++ b/drivers/staging/most/sound/sound.c @@ -457,7 +457,6 @@ static const struct snd_pcm_ops pcm_ops = { .trigger = pcm_trigger, .pointer = pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static int split_arg_list(char *buf, char **card_name, u16 *ch_num, diff --git a/include/drm/drm_audio_component.h b/include/drm/drm_audio_component.h new file mode 100644 index 000000000000..4923b00328c1 --- /dev/null +++ b/include/drm/drm_audio_component.h @@ -0,0 +1,118 @@ +// SPDX-License-Identifier: MIT +// Copyright © 2014 Intel Corporation + +#ifndef _DRM_AUDIO_COMPONENT_H_ +#define _DRM_AUDIO_COMPONENT_H_ + +struct drm_audio_component; + +/** + * struct drm_audio_component_ops - Ops implemented by DRM driver, called by hda driver + */ +struct drm_audio_component_ops { + /** + * @owner: drm module to pin down + */ + struct module *owner; + /** + * @get_power: get the POWER_DOMAIN_AUDIO power well + * + * Request the power well to be turned on. + */ + void (*get_power)(struct device *); + /** + * @put_power: put the POWER_DOMAIN_AUDIO power well + * + * Allow the power well to be turned off. + */ + void (*put_power)(struct device *); + /** + * @codec_wake_override: Enable/disable codec wake signal + */ + void (*codec_wake_override)(struct device *, bool enable); + /** + * @get_cdclk_freq: Get the Core Display Clock in kHz + */ + int (*get_cdclk_freq)(struct device *); + /** + * @sync_audio_rate: set n/cts based on the sample rate + * + * Called from audio driver. After audio driver sets the + * sample rate, it will call this function to set n/cts + */ + int (*sync_audio_rate)(struct device *, int port, int pipe, int rate); + /** + * @get_eld: fill the audio state and ELD bytes for the given port + * + * Called from audio driver to get the HDMI/DP audio state of the given + * digital port, and also fetch ELD bytes to the given pointer. + * + * It returns the byte size of the original ELD (not the actually + * copied size), zero for an invalid ELD, or a negative error code. + * + * Note that the returned size may be over @max_bytes. Then it + * implies that only a part of ELD has been copied to the buffer. + */ + int (*get_eld)(struct device *, int port, int pipe, bool *enabled, + unsigned char *buf, int max_bytes); +}; + +/** + * struct drm_audio_component_audio_ops - Ops implemented by hda driver, called by DRM driver + */ +struct drm_audio_component_audio_ops { + /** + * @audio_ptr: Pointer to be used in call to pin_eld_notify + */ + void *audio_ptr; + /** + * @pin_eld_notify: Notify the HDA driver that pin sense and/or ELD information has changed + * + * Called when the DRM driver has set up audio pipeline or has just + * begun to tear it down. This allows the HDA driver to update its + * status accordingly (even when the HDA controller is in power save + * mode). + */ + void (*pin_eld_notify)(void *audio_ptr, int port, int pipe); + /** + * @pin2port: Check and convert from pin node to port number + * + * Called by HDA driver to check and convert from the pin widget node + * number to a port number in the graphics side. + */ + int (*pin2port)(void *audio_ptr, int pin); + /** + * @master_bind: (Optional) component master bind callback + * + * Called at binding master component, for HDA codec-specific + * handling of dynamic binding. + */ + int (*master_bind)(struct device *dev, struct drm_audio_component *); + /** + * @master_unbind: (Optional) component master unbind callback + * + * Called at unbinding master component, for HDA codec-specific + * handling of dynamic unbinding. + */ + void (*master_unbind)(struct device *dev, struct drm_audio_component *); +}; + +/** + * struct drm_audio_component - Used for direct communication between DRM and hda drivers + */ +struct drm_audio_component { + /** + * @dev: DRM device, used as parameter for ops + */ + struct device *dev; + /** + * @ops: Ops implemented by DRM driver, called by hda driver + */ + const struct drm_audio_component_ops *ops; + /** + * @audio_ops: Ops implemented by hda driver, called by DRM driver + */ + const struct drm_audio_component_audio_ops *audio_ops; +}; + +#endif /* _DRM_AUDIO_COMPONENT_H_ */ diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index 346b1f5cb180..fca22d463e1b 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -24,101 +24,26 @@ #ifndef _I915_COMPONENT_H_ #define _I915_COMPONENT_H_ +#include "drm_audio_component.h" + /* MAX_PORT is the number of port * It must be sync with I915_MAX_PORTS defined i915_drv.h */ #define MAX_PORTS 6 /** - * struct i915_audio_component_ops - Ops implemented by i915 driver, called by hda driver - */ -struct i915_audio_component_ops { - /** - * @owner: i915 module - */ - struct module *owner; - /** - * @get_power: get the POWER_DOMAIN_AUDIO power well - * - * Request the power well to be turned on. - */ - void (*get_power)(struct device *); - /** - * @put_power: put the POWER_DOMAIN_AUDIO power well - * - * Allow the power well to be turned off. - */ - void (*put_power)(struct device *); - /** - * @codec_wake_override: Enable/disable codec wake signal - */ - void (*codec_wake_override)(struct device *, bool enable); - /** - * @get_cdclk_freq: Get the Core Display Clock in kHz - */ - int (*get_cdclk_freq)(struct device *); - /** - * @sync_audio_rate: set n/cts based on the sample rate - * - * Called from audio driver. After audio driver sets the - * sample rate, it will call this function to set n/cts - */ - int (*sync_audio_rate)(struct device *, int port, int pipe, int rate); - /** - * @get_eld: fill the audio state and ELD bytes for the given port - * - * Called from audio driver to get the HDMI/DP audio state of the given - * digital port, and also fetch ELD bytes to the given pointer. - * - * It returns the byte size of the original ELD (not the actually - * copied size), zero for an invalid ELD, or a negative error code. - * - * Note that the returned size may be over @max_bytes. Then it - * implies that only a part of ELD has been copied to the buffer. - */ - int (*get_eld)(struct device *, int port, int pipe, bool *enabled, - unsigned char *buf, int max_bytes); -}; - -/** - * struct i915_audio_component_audio_ops - Ops implemented by hda driver, called by i915 driver - */ -struct i915_audio_component_audio_ops { - /** - * @audio_ptr: Pointer to be used in call to pin_eld_notify - */ - void *audio_ptr; - /** - * @pin_eld_notify: Notify the HDA driver that pin sense and/or ELD information has changed - * - * Called when the i915 driver has set up audio pipeline or has just - * begun to tear it down. This allows the HDA driver to update its - * status accordingly (even when the HDA controller is in power save - * mode). - */ - void (*pin_eld_notify)(void *audio_ptr, int port, int pipe); -}; - -/** * struct i915_audio_component - Used for direct communication between i915 and hda drivers */ struct i915_audio_component { /** - * @dev: i915 device, used as parameter for ops + * @base: the drm_audio_component base class */ - struct device *dev; + struct drm_audio_component base; + /** * @aud_sample_rate: the array of audio sample rate per port */ int aud_sample_rate[MAX_PORTS]; - /** - * @ops: Ops implemented by i915 driver, called by hda driver - */ - const struct i915_audio_component_ops *ops; - /** - * @audio_ops: Ops implemented by hda driver, called by i915 driver - */ - const struct i915_audio_component_audio_ops *audio_ops; }; #endif /* _I915_COMPONENT_H_ */ diff --git a/include/linux/clk-provider.h b/include/linux/clk-provider.h index b7cfa037e593..08b1aa70a38d 100644 --- a/include/linux/clk-provider.h +++ b/include/linux/clk-provider.h @@ -38,6 +38,8 @@ #define CLK_IS_CRITICAL BIT(11) /* do not gate, ever */ /* parents need enable during gate/ungate, set rate and re-parent */ #define CLK_OPS_PARENT_ENABLE BIT(12) +/* duty cycle call may be forwarded to the parent clock */ +#define CLK_DUTY_CYCLE_PARENT BIT(13) struct clk; struct clk_hw; @@ -67,6 +69,17 @@ struct clk_rate_request { }; /** + * struct clk_duty - Struture encoding the duty cycle ratio of a clock + * + * @num: Numerator of the duty cycle ratio + * @den: Denominator of the duty cycle ratio + */ +struct clk_duty { + unsigned int num; + unsigned int den; +}; + +/** * struct clk_ops - Callback operations for hardware clocks; these are to * be provided by the clock implementation, and will be called by drivers * through the clk_* api. @@ -169,6 +182,15 @@ struct clk_rate_request { * by the second argument. Valid values for degrees are * 0-359. Return 0 on success, otherwise -EERROR. * + * @get_duty_cycle: Queries the hardware to get the current duty cycle ratio + * of a clock. Returned values denominator cannot be 0 and must be + * superior or equal to the numerator. + * + * @set_duty_cycle: Apply the duty cycle ratio to this clock signal specified by + * the numerator (2nd argurment) and denominator (3rd argument). + * Argument must be a valid ratio (denominator > 0 + * and >= numerator) Return 0 on success, otherwise -EERROR. + * * @init: Perform platform-specific initialization magic. * This is not not used by any of the basic clock types. * Please consider other ways of solving initialization problems @@ -218,6 +240,10 @@ struct clk_ops { unsigned long parent_accuracy); int (*get_phase)(struct clk_hw *hw); int (*set_phase)(struct clk_hw *hw, int degrees); + int (*get_duty_cycle)(struct clk_hw *hw, + struct clk_duty *duty); + int (*set_duty_cycle)(struct clk_hw *hw, + struct clk_duty *duty); void (*init)(struct clk_hw *hw); void (*debug_init)(struct clk_hw *hw, struct dentry *dentry); }; diff --git a/include/linux/clk.h b/include/linux/clk.h index 0dbd0885b2c2..4f750c481b82 100644 --- a/include/linux/clk.h +++ b/include/linux/clk.h @@ -142,6 +142,27 @@ int clk_set_phase(struct clk *clk, int degrees); int clk_get_phase(struct clk *clk); /** + * clk_set_duty_cycle - adjust the duty cycle ratio of a clock signal + * @clk: clock signal source + * @num: numerator of the duty cycle ratio to be applied + * @den: denominator of the duty cycle ratio to be applied + * + * Adjust the duty cycle of a clock signal by the specified ratio. Returns 0 on + * success, -EERROR otherwise. + */ +int clk_set_duty_cycle(struct clk *clk, unsigned int num, unsigned int den); + +/** + * clk_get_duty_cycle - return the duty cycle ratio of a clock signal + * @clk: clock signal source + * @scale: scaling factor to be applied to represent the ratio as an integer + * + * Returns the duty cycle ratio multiplied by the scale provided, otherwise + * returns -EERROR. + */ +int clk_get_scaled_duty_cycle(struct clk *clk, unsigned int scale); + +/** * clk_is_match - check if two clk's point to the same hardware clock * @p: clk compared against q * @q: clk compared against p @@ -183,6 +204,18 @@ static inline long clk_get_phase(struct clk *clk) return -ENOTSUPP; } +static inline int clk_set_duty_cycle(struct clk *clk, unsigned int num, + unsigned int den) +{ + return -ENOTSUPP; +} + +static inline unsigned int clk_get_scaled_duty_cycle(struct clk *clk, + unsigned int scale) +{ + return 0; +} + static inline bool clk_is_match(const struct clk *p, const struct clk *q) { return p == q; diff --git a/include/linux/dma/pxa-dma.h b/include/linux/dma/pxa-dma.h index e56ec7af4fd7..9fc594f69eff 100644 --- a/include/linux/dma/pxa-dma.h +++ b/include/linux/dma/pxa-dma.h @@ -9,6 +9,15 @@ enum pxad_chan_prio { PXAD_PRIO_LOWEST, }; +/** + * struct pxad_param - dma channel request parameters + * @drcmr: requestor line number + * @prio: minimal mandatory priority of the channel + * + * If a requested channel is granted, its priority will be at least @prio, + * ie. if PXAD_PRIO_LOW is required, the requested channel will be either + * PXAD_PRIO_LOW, PXAD_PRIO_NORMAL or PXAD_PRIO_HIGHEST. + */ struct pxad_param { unsigned int drcmr; enum pxad_chan_prio prio; diff --git a/include/linux/platform_data/mmp_dma.h b/include/linux/platform_data/mmp_dma.h index d1397c8ed94e..6397b9c8149a 100644 --- a/include/linux/platform_data/mmp_dma.h +++ b/include/linux/platform_data/mmp_dma.h @@ -12,9 +12,13 @@ #ifndef MMP_DMA_H #define MMP_DMA_H +struct dma_slave_map; + struct mmp_dma_platdata { int dma_channels; int nb_requestors; + int slave_map_cnt; + const struct dma_slave_map *slave_map; }; #endif /* MMP_DMA_H */ diff --git a/include/linux/pxa2xx_ssp.h b/include/linux/pxa2xx_ssp.h index 8461b18e4608..13b4244d44c1 100644 --- a/include/linux/pxa2xx_ssp.h +++ b/include/linux/pxa2xx_ssp.h @@ -171,6 +171,14 @@ #define SSACD_SCDB (1 << 3) /* SSPSYSCLK Divider Bypass */ #define SSACD_ACPS(x) ((x) << 4) /* Audio clock PLL select */ #define SSACD_ACDS(x) ((x) << 0) /* Audio clock divider select */ +#define SSACD_ACDS_1 (0) +#define SSACD_ACDS_2 (1) +#define SSACD_ACDS_4 (2) +#define SSACD_ACDS_8 (3) +#define SSACD_ACDS_16 (4) +#define SSACD_ACDS_32 (5) +#define SSACD_SCDB_4X (0) +#define SSACD_SCDB_1X (1) #define SSACD_SCDX8 (1 << 7) /* SYSCLK division ratio select */ /* LPSS SSP */ @@ -212,8 +220,6 @@ struct ssp_device { int type; int use_count; int irq; - int drcmr_rx; - int drcmr_tx; struct device_node *of_node; }; diff --git a/include/linux/usb/audio-v3.h b/include/linux/usb/audio-v3.h index a710e28b5215..6b708434b7f9 100644 --- a/include/linux/usb/audio-v3.h +++ b/include/linux/usb/audio-v3.h @@ -387,6 +387,12 @@ struct uac3_interrupt_data_msg { #define UAC3_CONNECTORS 0x0f #define UAC3_POWER_DOMAIN 0x10 +/* A.20 PROCESSING UNIT PROCESS TYPES */ +#define UAC3_PROCESS_UNDEFINED 0x00 +#define UAC3_PROCESS_UP_DOWNMIX 0x01 +#define UAC3_PROCESS_STEREO_EXTENDER 0x02 +#define UAC3_PROCESS_MULTI_FUNCTION 0x03 + /* A.22 AUDIO CLASS-SPECIFIC REQUEST CODES */ /* see audio-v2.h for the rest, which is identical to v2 */ #define UAC3_CS_REQ_INTEN 0x04 @@ -406,6 +412,15 @@ struct uac3_interrupt_data_msg { #define UAC3_TE_OVERFLOW 0x04 #define UAC3_TE_LATENCY 0x05 +/* A.23.10 PROCESSING UNITS CONTROL SELECTROS */ + +/* Up/Down Mixer */ +#define UAC3_UD_MODE_SELECT 0x01 + +/* Stereo Extender */ +#define UAC3_EXT_WIDTH_CONTROL 0x01 + + /* BADD predefined Unit/Terminal values */ #define UAC3_BADD_IT_ID1 1 /* Input Terminal ID1: bTerminalID = 1 */ #define UAC3_BADD_FU_ID2 2 /* Feature Unit ID2: bUnitID = 2 */ @@ -432,4 +447,8 @@ struct uac3_interrupt_data_msg { /* BADD sample rate is always fixed to 48kHz */ #define UAC3_BADD_SAMPLING_RATE 48000 +/* BADD power domains recovery times in 50us increments */ +#define UAC3_BADD_PD_RECOVER_D1D0 0x0258 /* 30ms */ +#define UAC3_BADD_PD_RECOVER_D2D0 0x1770 /* 300ms */ + #endif /* __LINUX_USB_AUDIO_V3_H */ diff --git a/include/linux/vga_switcheroo.h b/include/linux/vga_switcheroo.h index 77f0f0af3a71..a34539b7f750 100644 --- a/include/linux/vga_switcheroo.h +++ b/include/linux/vga_switcheroo.h @@ -84,8 +84,8 @@ enum vga_switcheroo_state { * Client identifier. Audio clients use the same identifier & 0x100. */ enum vga_switcheroo_client_id { - VGA_SWITCHEROO_UNKNOWN_ID = -1, - VGA_SWITCHEROO_IGD, + VGA_SWITCHEROO_UNKNOWN_ID = 0x1000, + VGA_SWITCHEROO_IGD = 0, VGA_SWITCHEROO_DIS, VGA_SWITCHEROO_MAX_CLIENTS, }; @@ -151,7 +151,7 @@ int vga_switcheroo_register_client(struct pci_dev *dev, bool driver_power_control); int vga_switcheroo_register_audio_client(struct pci_dev *pdev, const struct vga_switcheroo_client_ops *ops, - enum vga_switcheroo_client_id id); + struct pci_dev *vga_dev); void vga_switcheroo_client_fb_set(struct pci_dev *dev, struct fb_info *info); @@ -180,7 +180,7 @@ static inline int vga_switcheroo_register_handler(const struct vga_switcheroo_ha enum vga_switcheroo_handler_flags_t handler_flags) { return 0; } static inline int vga_switcheroo_register_audio_client(struct pci_dev *pdev, const struct vga_switcheroo_client_ops *ops, - enum vga_switcheroo_client_id id) { return 0; } + struct pci_dev *vga_dev) { return 0; } static inline void vga_switcheroo_unregister_handler(void) {} static inline enum vga_switcheroo_handler_flags_t vga_switcheroo_handler_flags(void) { return 0; } static inline int vga_switcheroo_lock_ddc(struct pci_dev *pdev) { return -ENODEV; } diff --git a/include/sound/ac97/codec.h b/include/sound/ac97/codec.h index ec04be9ab119..9792d25fa369 100644 --- a/include/sound/ac97/codec.h +++ b/include/sound/ac97/codec.h @@ -1,10 +1,8 @@ -/* - * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr> +/* SPDX-License-Identifier: GPL-2.0 * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. + * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr> */ + #ifndef __SOUND_AC97_CODEC2_H #define __SOUND_AC97_CODEC2_H diff --git a/include/sound/ac97/compat.h b/include/sound/ac97/compat.h index 1351cba40048..57e19afa31ab 100644 --- a/include/sound/ac97/compat.h +++ b/include/sound/ac97/compat.h @@ -1,14 +1,11 @@ -/* - * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr> +/* SPDX-License-Identifier: GPL-2.0 * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. + * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr> * * This file is for backward compatibility with snd_ac97 structure and its * multiple usages, such as the snd_ac97_bus and snd_ac97_build_ops. - * */ + #ifndef AC97_COMPAT_H #define AC97_COMPAT_H diff --git a/include/sound/ac97/controller.h b/include/sound/ac97/controller.h index b36ecdd64f14..06b5afb7fa6b 100644 --- a/include/sound/ac97/controller.h +++ b/include/sound/ac97/controller.h @@ -1,10 +1,8 @@ -/* - * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr> +/* SPDX-License-Identifier: GPL-2.0 * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. + * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr> */ + #ifndef AC97_CONTROLLER_H #define AC97_CONTROLLER_H diff --git a/include/sound/ac97/regs.h b/include/sound/ac97/regs.h index 9a4fa0c3264a..843f73f3705a 100644 --- a/include/sound/ac97/regs.h +++ b/include/sound/ac97/regs.h @@ -1,27 +1,11 @@ -/* +/* SPDX-License-Identifier: GPL-2.0+ + * * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.1 * by Intel Corporation (http://developer.intel.com). - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * */ - /* * AC'97 codec registers */ diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 89d311a503d3..cc383991c0fe 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -1,30 +1,15 @@ -#ifndef __SOUND_AC97_CODEC_H -#define __SOUND_AC97_CODEC_H - -/* +/* SPDX-License-Identifier: GPL-2.0+ + * * Copyright (c) by Jaroslav Kysela <perex@perex.cz> * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.1 * by Intel Corporation (http://developer.intel.com). - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * */ +#ifndef __SOUND_AC97_CODEC_H +#define __SOUND_AC97_CODEC_H + #include <linux/bitops.h> #include <linux/device.h> #include <linux/workqueue.h> diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 9924bc9cbc7c..ea8c93bbb0e0 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -1,27 +1,12 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * compress_driver.h - compress offload driver definations * * Copyright (C) 2011 Intel Corporation * Authors: Vinod Koul <vinod.koul@linux.intel.com> * Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * */ + #ifndef __COMPRESS_DRIVER_H #define __COMPRESS_DRIVER_H diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index e3481eebdd98..2c4cfaa135a6 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -1,17 +1,9 @@ -/* +/* SPDX-License-Identifier: GPL-2.0+ + * * Copyright (C) 2012, Analog Devices Inc. * Author: Lars-Peter Clausen <lars@metafoo.de> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 675 Mass Ave, Cambridge, MA 02139, USA. - * */ + #ifndef __SOUND_DMAENGINE_PCM_H__ #define __SOUND_DMAENGINE_PCM_H__ diff --git a/include/sound/hda_component.h b/include/sound/hda_component.h new file mode 100644 index 000000000000..78626cde7081 --- /dev/null +++ b/include/sound/hda_component.h @@ -0,0 +1,61 @@ +// SPDX-License-Identifier: GPL-2.0 +// HD-Audio helpers to sync with DRM driver + +#ifndef __SOUND_HDA_COMPONENT_H +#define __SOUND_HDA_COMPONENT_H + +#include <drm/drm_audio_component.h> + +#ifdef CONFIG_SND_HDA_COMPONENT +int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); +int snd_hdac_display_power(struct hdac_bus *bus, bool enable); +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, + int dev_id, int rate); +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, + bool *audio_enabled, char *buffer, int max_bytes); +int snd_hdac_acomp_init(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *aops, + int (*match_master)(struct device *, void *), + size_t extra_size); +int snd_hdac_acomp_exit(struct hdac_bus *bus); +int snd_hdac_acomp_register_notifier(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *ops); +#else +static inline int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) +{ + return 0; +} +static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable) +{ + return 0; +} +static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec, + hda_nid_t nid, int dev_id, int rate) +{ + return 0; +} +static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, + int dev_id, bool *audio_enabled, + char *buffer, int max_bytes) +{ + return -ENODEV; +} +static inline int snd_hdac_acomp_init(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *aops, + int (*match_master)(struct device *, void *), + size_t extra_size) +{ + return -ENODEV; +} +static inline int snd_hdac_acomp_exit(struct hdac_bus *bus) +{ + return 0; +} +static inline int snd_hdac_acomp_register_notifier(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *ops) +{ + return -ENODEV; +} +#endif + +#endif /* __SOUND_HDA_COMPONENT_H */ diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h index a94f5b6f92ac..6b79614a893b 100644 --- a/include/sound/hda_i915.h +++ b/include/sound/hda_i915.h @@ -5,54 +5,23 @@ #ifndef __SOUND_HDA_I915_H #define __SOUND_HDA_I915_H -#include <drm/i915_component.h> +#include "hda_component.h" #ifdef CONFIG_SND_HDA_I915 -int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); -int snd_hdac_display_power(struct hdac_bus *bus, bool enable); void snd_hdac_i915_set_bclk(struct hdac_bus *bus); -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, - int dev_id, int rate); -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, - bool *audio_enabled, char *buffer, int max_bytes); int snd_hdac_i915_init(struct hdac_bus *bus); -int snd_hdac_i915_exit(struct hdac_bus *bus); -int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *); #else -static inline int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) -{ - return 0; -} -static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable) -{ - return 0; -} static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { } -static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec, - hda_nid_t nid, int dev_id, int rate) -{ - return 0; -} -static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, - int dev_id, bool *audio_enabled, - char *buffer, int max_bytes) -{ - return -ENODEV; -} static inline int snd_hdac_i915_init(struct hdac_bus *bus) { return -ENODEV; } +#endif static inline int snd_hdac_i915_exit(struct hdac_bus *bus) { - return 0; + return snd_hdac_acomp_exit(bus); } -static inline int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *ops) -{ - return -ENODEV; -} -#endif #endif /* __SOUND_HDA_I915_H */ diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index c052afc27547..6f1e1f3b3063 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -8,8 +8,10 @@ #include <linux/device.h> #include <linux/interrupt.h> +#include <linux/pm_runtime.h> #include <linux/timecounter.h> #include <sound/core.h> +#include <sound/pcm.h> #include <sound/memalloc.h> #include <sound/hda_verbs.h> #include <drm/i915_component.h> @@ -132,7 +134,7 @@ int snd_hdac_get_sub_nodes(struct hdac_device *codec, hda_nid_t nid, hda_nid_t *start_id); unsigned int snd_hdac_calc_stream_format(unsigned int rate, unsigned int channels, - unsigned int format, + snd_pcm_format_t format, unsigned int maxbps, unsigned short spdif_ctls); int snd_hdac_query_supported_pcm(struct hdac_device *codec, hda_nid_t nid, @@ -171,12 +173,38 @@ int snd_hdac_power_down(struct hdac_device *codec); int snd_hdac_power_up_pm(struct hdac_device *codec); int snd_hdac_power_down_pm(struct hdac_device *codec); int snd_hdac_keep_power_up(struct hdac_device *codec); + +/* call this at entering into suspend/resume callbacks in codec driver */ +static inline void snd_hdac_enter_pm(struct hdac_device *codec) +{ + atomic_inc(&codec->in_pm); +} + +/* call this at leaving from suspend/resume callbacks in codec driver */ +static inline void snd_hdac_leave_pm(struct hdac_device *codec) +{ + atomic_dec(&codec->in_pm); +} + +static inline bool snd_hdac_is_in_pm(struct hdac_device *codec) +{ + return atomic_read(&codec->in_pm); +} + +static inline bool snd_hdac_is_power_on(struct hdac_device *codec) +{ + return !pm_runtime_suspended(&codec->dev); +} #else static inline int snd_hdac_power_up(struct hdac_device *codec) { return 0; } static inline int snd_hdac_power_down(struct hdac_device *codec) { return 0; } static inline int snd_hdac_power_up_pm(struct hdac_device *codec) { return 0; } static inline int snd_hdac_power_down_pm(struct hdac_device *codec) { return 0; } static inline int snd_hdac_keep_power_up(struct hdac_device *codec) { return 0; } +static inline void snd_hdac_enter_pm(struct hdac_device *codec) {} +static inline void snd_hdac_leave_pm(struct hdac_device *codec) {} +static inline bool snd_hdac_is_in_pm(struct hdac_device *codec) { return 0; } +static inline bool snd_hdac_is_power_on(struct hdac_device *codec) { return 1; } #endif /* @@ -188,6 +216,11 @@ struct hdac_driver { const struct hda_device_id *id_table; int (*match)(struct hdac_device *dev, struct hdac_driver *drv); void (*unsol_event)(struct hdac_device *dev, unsigned int event); + + /* fields used by ext bus APIs */ + int (*probe)(struct hdac_device *dev); + int (*remove)(struct hdac_device *dev); + void (*shutdown)(struct hdac_device *dev); }; #define drv_to_hdac_driver(_drv) container_of(_drv, struct hdac_driver, driver) @@ -209,6 +242,14 @@ struct hdac_bus_ops { }; /* + * ops used for ASoC HDA codec drivers + */ +struct hdac_ext_bus_ops { + int (*hdev_attach)(struct hdac_device *hdev); + int (*hdev_detach)(struct hdac_device *hdev); +}; + +/* * Lowlevel I/O operators */ struct hdac_io_ops { @@ -250,11 +291,17 @@ struct hdac_rb { * @mlcap: MultiLink capabilities pointer * @gtscap: gts capabilities pointer * @drsmcap: dma resume capabilities pointer + * @num_streams: streams supported + * @idx: HDA link index + * @hlink_list: link list of HDA links + * @lock: lock for link mgmt + * @cmd_dma_state: state of cmd DMAs: CORB and RIRB */ struct hdac_bus { struct device *dev; const struct hdac_bus_ops *ops; const struct hdac_io_ops *io_ops; + const struct hdac_ext_bus_ops *ext_ops; /* h/w resources */ unsigned long addr; @@ -314,9 +361,19 @@ struct hdac_bus { spinlock_t reg_lock; struct mutex cmd_mutex; - /* i915 component interface */ - struct i915_audio_component *audio_component; - int i915_power_refcount; + /* DRM component interface */ + struct drm_audio_component *audio_component; + int drm_power_refcount; + + /* parameters required for enhanced capabilities */ + int num_streams; + int idx; + + struct list_head hlink_list; + + struct mutex lock; + bool cmd_dma_state; + }; int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 9c14e21dda85..f34aced69ca8 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -4,38 +4,16 @@ #include <sound/hdaudio.h> -/** - * hdac_ext_bus: HDAC extended bus for extended HDA caps - * - * @bus: hdac bus - * @num_streams: streams supported - * @hlink_list: link list of HDA links - * @lock: lock for link mgmt - * @cmd_dma_state: state of cmd DMAs: CORB and RIRB - */ -struct hdac_ext_bus { - struct hdac_bus bus; - int num_streams; - int idx; - - struct list_head hlink_list; - - struct mutex lock; - bool cmd_dma_state; -}; - -int snd_hdac_ext_bus_init(struct hdac_ext_bus *sbus, struct device *dev, +int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, const struct hdac_bus_ops *ops, - const struct hdac_io_ops *io_ops); + const struct hdac_io_ops *io_ops, + const struct hdac_ext_bus_ops *ext_ops); -void snd_hdac_ext_bus_exit(struct hdac_ext_bus *sbus); -int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *sbus, int addr); +void snd_hdac_ext_bus_exit(struct hdac_bus *bus); +int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, + struct hdac_device *hdev); void snd_hdac_ext_bus_device_exit(struct hdac_device *hdev); -void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus); - -#define ebus_to_hbus(ebus) (&(ebus)->bus) -#define hbus_to_ebus(_bus) \ - container_of(_bus, struct hdac_ext_bus, bus) +void snd_hdac_ext_bus_device_remove(struct hdac_bus *bus); #define HDA_CODEC_REV_EXT_ENTRY(_vid, _rev, _name, drv_data) \ { .vendor_id = (_vid), .rev_id = (_rev), .name = (_name), \ @@ -44,14 +22,14 @@ void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus); #define HDA_CODEC_EXT_ENTRY(_vid, _revid, _name, _drv_data) \ HDA_CODEC_REV_EXT_ENTRY(_vid, _revid, _name, _drv_data) -void snd_hdac_ext_bus_ppcap_enable(struct hdac_ext_bus *chip, bool enable); -void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_ext_bus *chip, bool enable); +void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *chip, bool enable); +void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *chip, bool enable); -void snd_hdac_ext_stream_spbcap_enable(struct hdac_ext_bus *chip, +void snd_hdac_ext_stream_spbcap_enable(struct hdac_bus *chip, bool enable, int index); -int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *bus); -struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_ext_bus *bus, +int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_bus *bus); +struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_bus *bus, const char *codec_name); enum hdac_ext_stream_type { @@ -100,28 +78,28 @@ struct hdac_ext_stream { #define stream_to_hdac_ext_stream(s) \ container_of(s, struct hdac_ext_stream, hstream) -void snd_hdac_ext_stream_init(struct hdac_ext_bus *bus, +void snd_hdac_ext_stream_init(struct hdac_bus *bus, struct hdac_ext_stream *stream, int idx, int direction, int tag); -int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx, +int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx, int num_stream, int dir); -void snd_hdac_stream_free_all(struct hdac_ext_bus *ebus); -void snd_hdac_link_free_all(struct hdac_ext_bus *ebus); -struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_ext_bus *bus, +void snd_hdac_stream_free_all(struct hdac_bus *bus); +void snd_hdac_link_free_all(struct hdac_bus *bus); +struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream, int type); void snd_hdac_ext_stream_release(struct hdac_ext_stream *azx_dev, int type); -void snd_hdac_ext_stream_decouple(struct hdac_ext_bus *bus, +void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, struct hdac_ext_stream *azx_dev, bool decouple); -void snd_hdac_ext_stop_streams(struct hdac_ext_bus *sbus); +void snd_hdac_ext_stop_streams(struct hdac_bus *bus); -int snd_hdac_ext_stream_set_spib(struct hdac_ext_bus *ebus, +int snd_hdac_ext_stream_set_spib(struct hdac_bus *bus, struct hdac_ext_stream *stream, u32 value); -int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_ext_bus *ebus, +int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_bus *bus, struct hdac_ext_stream *stream); -void snd_hdac_ext_stream_drsm_enable(struct hdac_ext_bus *ebus, +void snd_hdac_ext_stream_drsm_enable(struct hdac_bus *bus, bool enable, int index); -int snd_hdac_ext_stream_set_dpibr(struct hdac_ext_bus *ebus, +int snd_hdac_ext_stream_set_dpibr(struct hdac_bus *bus, struct hdac_ext_stream *stream, u32 value); int snd_hdac_ext_stream_set_lpib(struct hdac_ext_stream *stream, u32 value); @@ -144,17 +122,15 @@ struct hdac_ext_link { int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link); int snd_hdac_ext_bus_link_power_down(struct hdac_ext_link *link); -int snd_hdac_ext_bus_link_power_up_all(struct hdac_ext_bus *ebus); -int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus); +int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus); +int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus); void snd_hdac_ext_link_set_stream_id(struct hdac_ext_link *link, int stream); void snd_hdac_ext_link_clear_stream_id(struct hdac_ext_link *link, int stream); -int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus, - struct hdac_ext_link *link); -int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, - struct hdac_ext_link *link); +int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, struct hdac_ext_link *link); +int snd_hdac_ext_bus_link_put(struct hdac_bus *bus, struct hdac_ext_link *link); /* update register macro */ #define snd_hdac_updatel(addr, reg, mask, val) \ @@ -181,53 +157,12 @@ struct hda_dai_map { u32 maxbps; }; -#define HDA_MAX_NIDS 16 - -/** - * struct hdac_ext_device - HDAC Ext device - * - * @hdac: hdac core device - * @nid_list - the dai map which matches the dai-name with the nid - * @map_cur_idx - the idx in use in dai_map - * @ops - the hda codec ops common to all codec drivers - * @pvt_data - private data, for asoc contains asoc codec object - */ -struct hdac_ext_device { - struct hdac_device hdev; - struct hdac_ext_bus *ebus; - - /* soc-dai to nid map */ - struct hda_dai_map nid_list[HDA_MAX_NIDS]; - unsigned int map_cur_idx; - - /* codec ops */ - struct hdac_ext_codec_ops ops; - - struct snd_card *card; - void *scodec; - void *private_data; -}; - struct hdac_ext_dma_params { u32 format; u8 stream_tag; }; -#define to_ehdac_device(dev) (container_of((dev), \ - struct hdac_ext_device, hdev)) -/* - * HD-audio codec base driver - */ -struct hdac_ext_driver { - struct hdac_driver hdac; - - int (*probe)(struct hdac_ext_device *dev); - int (*remove)(struct hdac_ext_device *dev); - void (*shutdown)(struct hdac_ext_device *dev); -}; - -int snd_hda_ext_driver_register(struct hdac_ext_driver *drv); -void snd_hda_ext_driver_unregister(struct hdac_ext_driver *drv); -#define to_ehdac_driver(_drv) container_of(_drv, struct hdac_ext_driver, hdac) +int snd_hda_ext_driver_register(struct hdac_driver *drv); +void snd_hda_ext_driver_unregister(struct hdac_driver *drv); #endif /* __SOUND_HDAUDIO_EXT_H */ diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 9c3db3dce32b..67561b997915 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -24,6 +24,8 @@ #ifndef __SOUND_MEMALLOC_H #define __SOUND_MEMALLOC_H +#include <asm/page.h> + struct device; /* @@ -67,6 +69,14 @@ struct snd_dma_buffer { void *private_data; /* private for allocator; don't touch */ }; +/* + * return the pages matching with the given byte size + */ +static inline unsigned int snd_sgbuf_aligned_pages(size_t size) +{ + return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; +} + #ifdef CONFIG_SND_DMA_SGBUF /* * Scatter-Gather generic device pages @@ -91,14 +101,6 @@ struct snd_sg_buf { }; /* - * return the pages matching with the given byte size - */ -static inline unsigned int snd_sgbuf_aligned_pages(size_t size) -{ - return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; -} - -/* * return the physical address at the corresponding offset */ static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab, diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e054c583d3b3..d6bd3caf6878 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -462,6 +462,7 @@ struct snd_pcm_substream { /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ + long wait_time; /* time in ms for R/W to wait for avail */ /* -- next substream -- */ struct snd_pcm_substream *next; /* -- linked substreams -- */ @@ -1089,14 +1090,14 @@ static inline snd_pcm_sframes_t snd_pcm_lib_write(struct snd_pcm_substream *substream, const void __user *buf, snd_pcm_uframes_t frames) { - return __snd_pcm_lib_xfer(substream, (void *)buf, true, frames, false); + return __snd_pcm_lib_xfer(substream, (void __force *)buf, true, frames, false); } static inline snd_pcm_sframes_t snd_pcm_lib_read(struct snd_pcm_substream *substream, void __user *buf, snd_pcm_uframes_t frames) { - return __snd_pcm_lib_xfer(substream, (void *)buf, true, frames, false); + return __snd_pcm_lib_xfer(substream, (void __force *)buf, true, frames, false); } static inline snd_pcm_sframes_t @@ -1341,8 +1342,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s #define snd_pcm_lib_mmap_iomem NULL #endif -#define snd_pcm_lib_mmap_vmalloc NULL - /** * snd_pcm_limit_isa_dma_size - Get the max size fitting with ISA DMA transfer * @dma: DMA number diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index c704357775fc..2dd37cada7c0 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -87,6 +87,13 @@ static inline void snd_mask_set(struct snd_mask *mask, unsigned int val) mask->bits[MASK_OFS(val)] |= MASK_BIT(val); } +/* Most of drivers need only this one */ +static inline void snd_mask_set_format(struct snd_mask *mask, + snd_pcm_format_t format) +{ + snd_mask_set(mask, (__force unsigned int)format); +} + static inline void snd_mask_reset(struct snd_mask *mask, unsigned int val) { mask->bits[MASK_OFS(val)] &= ~MASK_BIT(val); @@ -369,8 +376,7 @@ static inline int params_physical_width(const struct snd_pcm_hw_params *p) static inline void params_set_format(struct snd_pcm_hw_params *p, snd_pcm_format_t fmt) { - snd_mask_set(hw_param_mask(p, SNDRV_PCM_HW_PARAM_FORMAT), - (__force int)fmt); + snd_mask_set_format(hw_param_mask(p, SNDRV_PCM_HW_PARAM_FORMAT), fmt); } #endif /* __SOUND_PCM_PARAMS_H */ diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h index 63f75450d3db..6758fc12fa84 100644 --- a/include/sound/pxa2xx-lib.h +++ b/include/sound/pxa2xx-lib.h @@ -8,20 +8,23 @@ /* PCM */ struct snd_pcm_substream; struct snd_pcm_hw_params; +struct snd_soc_pcm_runtime; struct snd_pcm; -extern int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, +extern int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); -extern int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream); +extern int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream); extern int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd); extern snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream); -extern int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream); -extern int __pxa2xx_pcm_open(struct snd_pcm_substream *substream); -extern int __pxa2xx_pcm_close(struct snd_pcm_substream *substream); +extern int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream); +extern int pxa2xx_pcm_open(struct snd_pcm_substream *substream); +extern int pxa2xx_pcm_close(struct snd_pcm_substream *substream); extern int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); extern int pxa2xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream); extern void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm); +extern int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd); +extern const struct snd_pcm_ops pxa2xx_pcm_ops; /* AC97 */ diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h new file mode 100644 index 000000000000..0251797ab438 --- /dev/null +++ b/include/sound/rt5682.h @@ -0,0 +1,40 @@ +/* + * linux/sound/rt5682.h -- Platform data for RT5682 + * + * Copyright 2018 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT5682_H +#define __LINUX_SND_RT5682_H + +enum rt5682_dmic1_data_pin { + RT5682_DMIC1_NULL, + RT5682_DMIC1_DATA_GPIO2, + RT5682_DMIC1_DATA_GPIO5, +}; + +enum rt5682_dmic1_clk_pin { + RT5682_DMIC1_CLK_GPIO1, + RT5682_DMIC1_CLK_GPIO3, +}; + +enum rt5682_jd_src { + RT5682_JD_NULL, + RT5682_JD1, +}; + +struct rt5682_platform_data { + + int ldo1_en; /* GPIO for LDO1_EN */ + + enum rt5682_dmic1_data_pin dmic1_data_pin; + enum rt5682_dmic1_clk_pin dmic1_clk_pin; + enum rt5682_jd_src jd_src; +}; + +#endif + diff --git a/include/sound/sb16_csp.h b/include/sound/sb16_csp.h index c7c7788005e4..7817e88bd08d 100644 --- a/include/sound/sb16_csp.h +++ b/include/sound/sb16_csp.h @@ -46,7 +46,7 @@ enum { struct snd_sb_csp_ops { int (*csp_use) (struct snd_sb_csp * p); int (*csp_unuse) (struct snd_sb_csp * p); - int (*csp_autoload) (struct snd_sb_csp * p, int pcm_sfmt, int play_rec_mode); + int (*csp_autoload) (struct snd_sb_csp * p, snd_pcm_format_t pcm_sfmt, int play_rec_mode); int (*csp_start) (struct snd_sb_csp * p, int sample_width, int channels); int (*csp_stop) (struct snd_sb_csp * p); int (*csp_qsound_transfer) (struct snd_sb_csp * p); diff --git a/include/sound/seq_midi_event.h b/include/sound/seq_midi_event.h index e40f43e6fc7b..2f135bccf457 100644 --- a/include/sound/seq_midi_event.h +++ b/include/sound/seq_midi_event.h @@ -43,10 +43,8 @@ void snd_midi_event_free(struct snd_midi_event *dev); void snd_midi_event_reset_encode(struct snd_midi_event *dev); void snd_midi_event_reset_decode(struct snd_midi_event *dev); void snd_midi_event_no_status(struct snd_midi_event *dev, int on); -/* encode from byte stream - return number of written bytes if success */ -long snd_midi_event_encode(struct snd_midi_event *dev, unsigned char *buf, long count, - struct snd_seq_event *ev); -int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, struct snd_seq_event *ev); +bool snd_midi_event_encode_byte(struct snd_midi_event *dev, unsigned char c, + struct snd_seq_event *ev); /* decode from event to bytes - return number of written bytes if success */ long snd_midi_event_decode(struct snd_midi_event *dev, unsigned char *buf, long count, struct snd_seq_event *ev); diff --git a/include/sound/seq_virmidi.h b/include/sound/seq_virmidi.h index 695257ae64ac..796ce7772213 100644 --- a/include/sound/seq_virmidi.h +++ b/include/sound/seq_virmidi.h @@ -36,11 +36,12 @@ struct snd_virmidi { int seq_mode; int client; int port; - unsigned int trigger: 1; + bool trigger; struct snd_midi_event *parser; struct snd_seq_event event; struct snd_virmidi_dev *rdev; struct snd_rawmidi_substream *substream; + struct work_struct output_work; }; #define SNDRV_VIRMIDI_SUBSCRIBE (1<<0) diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index 7a9710b4b799..89eafe23ef88 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -1,16 +1,13 @@ -#ifndef __SOUND_FSI_H -#define __SOUND_FSI_H - -/* +/* SPDX-License-Identifier: GPL-2.0 + * * Fifo-attached Serial Interface (FSI) support for SH7724 * * Copyright (C) 2009 Renesas Solutions Corp. * Kuninori Morimoto <morimoto.kuninori@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ +#ifndef __SOUND_FSI_H +#define __SOUND_FSI_H + #include <linux/clk.h> #include <sound/soc.h> diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index a6a2e1547092..d264e5463f22 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -1,12 +1,9 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * ASoC simple sound card support * * Copyright (C) 2012 Renesas Solutions Corp. * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __SIMPLE_CARD_H diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 7e25afce6566..8bc5e2d8b13c 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -1,17 +1,20 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * simple_card_utils.h * * Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ + #ifndef __SIMPLE_CARD_UTILS_H #define __SIMPLE_CARD_UTILS_H #include <sound/soc.h> +#define asoc_simple_card_init_hp(card, sjack, prefix) \ + asoc_simple_card_init_jack(card, sjack, 1, prefix) +#define asoc_simple_card_init_mic(card, sjack, prefix) \ + asoc_simple_card_init_jack(card, sjack, 0, prefix) + struct asoc_simple_dai { const char *name; unsigned int sysclk; @@ -28,6 +31,12 @@ struct asoc_simple_card_data { u32 convert_channels; }; +struct asoc_simple_jack { + struct snd_soc_jack jack; + struct snd_soc_jack_pin pin; + struct snd_soc_jack_gpio gpio; +}; + int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *node, struct device_node *codec, @@ -107,4 +116,8 @@ int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, char *prefix); +int asoc_simple_card_init_jack(struct snd_soc_card *card, + struct asoc_simple_jack *sjack, + int is_hp, char *prefix); + #endif /* __SIMPLE_CARD_UTILS_H */ diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h index 9da6388c20a1..bb1d24b703fb 100644 --- a/include/sound/soc-acpi-intel-match.h +++ b/include/sound/soc-acpi-intel-match.h @@ -1,16 +1,6 @@ - -/* - * Copyright (C) 2017, Intel Corporation. All rights reserved. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License version - * 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. +/* SPDX-License-Identifier: GPL-2.0 * + * Copyright (C) 2017, Intel Corporation. All rights reserved. */ #ifndef __LINUX_SND_SOC_ACPI_INTEL_MATCH_H @@ -29,5 +19,10 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_legacy_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_skl_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[]; #endif diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index 082224275f52..e45b2330d16a 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -1,15 +1,6 @@ -/* - * Copyright (C) 2013-15, Intel Corporation. All rights reserved. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License version - * 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. +/* SPDX-License-Identifier: GPL-2.0 * + * Copyright (C) 2013-15, Intel Corporation. All rights reserved. */ #ifndef __LINUX_SND_SOC_ACPI_H diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index e6f8c40ed43c..f5d70041108f 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -1,12 +1,9 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * linux/sound/soc-dai.h -- ALSA SoC Layer * * Copyright: 2005-2008 Wolfson Microelectronics. PLC. * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * * Digital Audio Interface (DAI) API. */ @@ -141,6 +138,11 @@ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, int direction); + +int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai, + unsigned int *tx_num, unsigned int *tx_slot, + unsigned int *rx_num, unsigned int *rx_slot); + int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); struct snd_soc_dai_ops { @@ -168,6 +170,9 @@ struct snd_soc_dai_ops { int (*set_channel_map)(struct snd_soc_dai *dai, unsigned int tx_num, unsigned int *tx_slot, unsigned int rx_num, unsigned int *rx_slot); + int (*get_channel_map)(struct snd_soc_dai *dai, + unsigned int *tx_num, unsigned int *tx_slot, + unsigned int *rx_num, unsigned int *rx_slot); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); int (*set_sdw_stream)(struct snd_soc_dai *dai, diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a6ce2de4e20a..af9ef16cc34d 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -1,13 +1,10 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * linux/sound/soc-dapm.h -- ALSA SoC Dynamic Audio Power Management * - * Author: Liam Girdwood - * Created: Aug 11th 2005 + * Author: Liam Girdwood + * Created: Aug 11th 2005 * Copyright: Wolfson Microelectronics. PLC. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __LINUX_SND_SOC_DAPM_H diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 806059052bfc..9bb92f187af8 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -1,11 +1,8 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * linux/sound/soc-dpcm.h -- ALSA SoC Dynamic PCM Support * * Author: Liam Girdwood <lrg@ti.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __LINUX_SND_SOC_DPCM_H diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index f552c3f56368..fa4b8413d2e2 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -1,13 +1,10 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * linux/sound/soc-topology.h -- ALSA SoC Firmware Controls and DAPM * * Copyright (C) 2012 Texas Instruments Inc. * Copyright (C) 2015 Intel Corporation. * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * * Simple file API to load FW that includes mixers, coefficients, DAPM graphs, * algorithms, equalisers, DAIs, widgets, FE caps, BE caps, codec link caps etc. */ @@ -30,6 +27,9 @@ struct snd_soc_dapm_context; struct snd_soc_card; struct snd_kcontrol_new; struct snd_soc_dai_link; +struct snd_soc_dai_driver; +struct snd_soc_dai; +struct snd_soc_dapm_route; /* object scan be loaded and unloaded in groups with identfying indexes */ #define SND_SOC_TPLG_INDEX_ALL 0 /* ID that matches all FW objects */ @@ -109,35 +109,44 @@ struct snd_soc_tplg_widget_events { struct snd_soc_tplg_ops { /* external kcontrol init - used for any driver specific init */ - int (*control_load)(struct snd_soc_component *, + int (*control_load)(struct snd_soc_component *, int index, struct snd_kcontrol_new *, struct snd_soc_tplg_ctl_hdr *); int (*control_unload)(struct snd_soc_component *, struct snd_soc_dobj *); + /* DAPM graph route element loading and unloading */ + int (*dapm_route_load)(struct snd_soc_component *, int index, + struct snd_soc_dapm_route *route); + int (*dapm_route_unload)(struct snd_soc_component *, + struct snd_soc_dobj *); + /* external widget init - used for any driver specific init */ - int (*widget_load)(struct snd_soc_component *, + int (*widget_load)(struct snd_soc_component *, int index, struct snd_soc_dapm_widget *, struct snd_soc_tplg_dapm_widget *); - int (*widget_ready)(struct snd_soc_component *, + int (*widget_ready)(struct snd_soc_component *, int index, struct snd_soc_dapm_widget *, struct snd_soc_tplg_dapm_widget *); int (*widget_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* FE DAI - used for any driver specific init */ - int (*dai_load)(struct snd_soc_component *, - struct snd_soc_dai_driver *dai_drv); + int (*dai_load)(struct snd_soc_component *, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai); + int (*dai_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* DAI link - used for any driver specific init */ - int (*link_load)(struct snd_soc_component *, - struct snd_soc_dai_link *link); + int (*link_load)(struct snd_soc_component *, int index, + struct snd_soc_dai_link *link, + struct snd_soc_tplg_link_config *cfg); int (*link_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* callback to handle vendor bespoke data */ - int (*vendor_load)(struct snd_soc_component *, + int (*vendor_load)(struct snd_soc_component *, int index, struct snd_soc_tplg_hdr *); int (*vendor_unload)(struct snd_soc_component *, struct snd_soc_tplg_hdr *); @@ -146,7 +155,7 @@ struct snd_soc_tplg_ops { void (*complete)(struct snd_soc_component *); /* manifest - optional to inform component of manifest */ - int (*manifest)(struct snd_soc_component *, + int (*manifest)(struct snd_soc_component *, int index, struct snd_soc_tplg_manifest *); /* vendor specific kcontrol handlers available for binding */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 1378dcd2128a..41cec42fb456 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1,13 +1,10 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * linux/sound/soc.h -- ALSA SoC Layer * - * Author: Liam Girdwood - * Created: Aug 11th 2005 + * Author: Liam Girdwood + * Created: Aug 11th 2005 * Copyright: Wolfson Microelectronics. PLC. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __LINUX_SND_SOC_H @@ -806,6 +803,14 @@ struct snd_soc_component_driver { unsigned int use_pmdown_time:1; /* care pmdown_time at stop */ unsigned int endianness:1; unsigned int non_legacy_dai_naming:1; + + /* this component uses topology and ignore machine driver FEs */ + const char *ignore_machine; + const char *topology_name_prefix; + int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); + bool use_dai_pcm_id; /* use the DAI link PCM ID as PCM device number */ + int be_pcm_base; /* base device ID for all BE PCMs */ }; struct snd_soc_component { @@ -957,10 +962,17 @@ struct snd_soc_dai_link { /* DPCM used FE & BE merged format */ unsigned int dpcm_merged_format:1; + /* DPCM used FE & BE merged channel */ + unsigned int dpcm_merged_chan:1; + /* DPCM used FE & BE merged rate */ + unsigned int dpcm_merged_rate:1; /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; + /* Do not create a PCM for this DAI link (Backend link) */ + unsigned int ignore:1; + struct list_head list; /* DAI link list of the soc card */ struct snd_soc_dobj dobj; /* For topology */ }; @@ -1000,6 +1012,7 @@ struct snd_soc_card { const char *long_name; const char *driver_name; char dmi_longname[80]; + char topology_shortname[32]; struct device *dev; struct snd_card *snd_card; @@ -1009,6 +1022,7 @@ struct snd_soc_card { struct mutex dapm_mutex; bool instantiated; + bool topology_shortname_created; int (*probe)(struct snd_soc_card *card); int (*late_probe)(struct snd_soc_card *card); @@ -1412,6 +1426,9 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, const char *propname); +int snd_soc_of_get_slot_mask(struct device_node *np, + const char *prop_name, + unsigned int *mask); int snd_soc_of_parse_tdm_slot(struct device_node *np, unsigned int *tx_mask, unsigned int *rx_mask, diff --git a/include/trace/events/clk.h b/include/trace/events/clk.h index 2cd449328aee..9004ffff7f32 100644 --- a/include/trace/events/clk.h +++ b/include/trace/events/clk.h @@ -192,6 +192,42 @@ DEFINE_EVENT(clk_phase, clk_set_phase_complete, TP_ARGS(core, phase) ); +DECLARE_EVENT_CLASS(clk_duty_cycle, + + TP_PROTO(struct clk_core *core, struct clk_duty *duty), + + TP_ARGS(core, duty), + + TP_STRUCT__entry( + __string( name, core->name ) + __field( unsigned int, num ) + __field( unsigned int, den ) + ), + + TP_fast_assign( + __assign_str(name, core->name); + __entry->num = duty->num; + __entry->den = duty->den; + ), + + TP_printk("%s %u/%u", __get_str(name), (unsigned int)__entry->num, + (unsigned int)__entry->den) +); + +DEFINE_EVENT(clk_duty_cycle, clk_set_duty_cycle, + + TP_PROTO(struct clk_core *core, struct clk_duty *duty), + + TP_ARGS(core, duty) +); + +DEFINE_EVENT(clk_duty_cycle, clk_set_duty_cycle_complete, + + TP_PROTO(struct clk_core *core, struct clk_duty *duty), + + TP_ARGS(core, duty) +); + #endif /* _TRACE_CLK_H */ /* This part must be outside protection */ diff --git a/include/uapi/linux/usb/audio.h b/include/uapi/linux/usb/audio.h index 74e520fb944f..ddc5396800aa 100644 --- a/include/uapi/linux/usb/audio.h +++ b/include/uapi/linux/usb/audio.h @@ -390,33 +390,64 @@ static inline __u8 uac_processing_unit_iChannelNames(struct uac_processing_unit_ static inline __u8 uac_processing_unit_bControlSize(struct uac_processing_unit_descriptor *desc, int protocol) { - return (protocol == UAC_VERSION_1) ? - desc->baSourceID[desc->bNrInPins + 4] : - 2; /* in UAC2, this value is constant */ + switch (protocol) { + case UAC_VERSION_1: + return desc->baSourceID[desc->bNrInPins + 4]; + case UAC_VERSION_2: + return 2; /* in UAC2, this value is constant */ + case UAC_VERSION_3: + return 4; /* in UAC3, this value is constant */ + default: + return 1; + } } static inline __u8 *uac_processing_unit_bmControls(struct uac_processing_unit_descriptor *desc, int protocol) { - return (protocol == UAC_VERSION_1) ? - &desc->baSourceID[desc->bNrInPins + 5] : - &desc->baSourceID[desc->bNrInPins + 6]; + switch (protocol) { + case UAC_VERSION_1: + return &desc->baSourceID[desc->bNrInPins + 5]; + case UAC_VERSION_2: + return &desc->baSourceID[desc->bNrInPins + 6]; + case UAC_VERSION_3: + return &desc->baSourceID[desc->bNrInPins + 2]; + default: + return NULL; + } } static inline __u8 uac_processing_unit_iProcessing(struct uac_processing_unit_descriptor *desc, int protocol) { __u8 control_size = uac_processing_unit_bControlSize(desc, protocol); - return *(uac_processing_unit_bmControls(desc, protocol) - + control_size); + + switch (protocol) { + case UAC_VERSION_1: + case UAC_VERSION_2: + default: + return *(uac_processing_unit_bmControls(desc, protocol) + + control_size); + case UAC_VERSION_3: + return 0; /* UAC3 does not have this field */ + } } static inline __u8 *uac_processing_unit_specific(struct uac_processing_unit_descriptor *desc, int protocol) { __u8 control_size = uac_processing_unit_bControlSize(desc, protocol); - return uac_processing_unit_bmControls(desc, protocol) + + switch (protocol) { + case UAC_VERSION_1: + case UAC_VERSION_2: + default: + return uac_processing_unit_bmControls(desc, protocol) + control_size + 1; + case UAC_VERSION_3: + return uac_processing_unit_bmControls(desc, protocol) + + control_size; + } } /* 4.5.2 Class-Specific AS Interface Descriptor */ diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 31f858eceffc..7a0dfca03a57 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -13,6 +13,7 @@ #include <linux/idr.h> #include <linux/list.h> #include <linux/mutex.h> +#include <linux/of.h> #include <linux/pm.h> #include <linux/pm_runtime.h> #include <linux/slab.h> @@ -68,6 +69,27 @@ ac97_codec_find(struct ac97_controller *ac97_ctrl, unsigned int codec_num) return ac97_ctrl->codecs[codec_num]; } +static struct device_node * +ac97_of_get_child_device(struct ac97_controller *ac97_ctrl, int idx, + unsigned int vendor_id) +{ + struct device_node *node; + u32 reg; + char compat[] = "ac97,0000,0000"; + + snprintf(compat, sizeof(compat), "ac97,%04x,%04x", + vendor_id >> 16, vendor_id & 0xffff); + + for_each_child_of_node(ac97_ctrl->parent->of_node, node) { + if ((idx != of_property_read_u32(node, "reg", ®)) || + !of_device_is_compatible(node, compat)) + continue; + return of_node_get(node); + } + + return NULL; +} + static void ac97_codec_release(struct device *dev) { struct ac97_codec_device *adev; @@ -76,6 +98,7 @@ static void ac97_codec_release(struct device *dev) adev = to_ac97_device(dev); ac97_ctrl = adev->ac97_ctrl; ac97_ctrl->codecs[adev->num] = NULL; + of_node_put(dev->of_node); kfree(adev); } @@ -98,6 +121,8 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx, device_initialize(&codec->dev); dev_set_name(&codec->dev, "%s:%u", dev_name(ac97_ctrl->parent), idx); + codec->dev.of_node = ac97_of_get_child_device(ac97_ctrl, idx, + vendor_id); ret = device_add(&codec->dev); if (ret) @@ -105,6 +130,7 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx, return 0; err_free_codec: + of_node_put(codec->dev.of_node); put_device(&codec->dev); kfree(codec); ac97_ctrl->codecs[idx] = NULL; diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c index 71960089e207..65557421fe0b 100644 --- a/sound/aoa/core/gpio-feature.c +++ b/sound/aoa/core/gpio-feature.c @@ -88,8 +88,10 @@ static struct device_node *get_gpio(char *name, } reg = of_get_property(np, "reg", NULL); - if (!reg) + if (!reg) { + of_node_put(np); return NULL; + } *gpioptr = *reg; diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 65171f6657a2..5fbd47a9177e 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -17,14 +17,9 @@ config SND_ARMAACI select SND_PCM select SND_AC97_CODEC -config SND_PXA2XX_PCM - tristate - select SND_PCM - config SND_PXA2XX_AC97 tristate "AC97 driver for the Intel PXA2xx chip" depends on ARCH_PXA - select SND_PXA2XX_PCM select SND_AC97_CODEC select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 diff --git a/sound/arm/Makefile b/sound/arm/Makefile index e10d5b169565..34c769489877 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -6,9 +6,6 @@ obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o snd-aaci-objs := aaci.o -obj-$(CONFIG_SND_PXA2XX_PCM) += snd-pxa2xx-pcm.o -snd-pxa2xx-pcm-objs := pxa2xx-pcm.o - obj-$(CONFIG_SND_PXA2XX_LIB) += snd-pxa2xx-lib.o snd-pxa2xx-lib-y := pxa2xx-pcm-lib.o snd-pxa2xx-lib-$(CONFIG_SND_PXA2XX_LIB_AC97) += pxa2xx-ac97-lib.o diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 5950a9e218d9..8eafd3d3dff6 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/io.h> #include <linux/gpio.h> +#include <linux/of_gpio.h> #include <sound/pxa2xx-lib.h> @@ -337,6 +338,17 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev) dev_err(&dev->dev, "Invalid reset GPIO %d\n", pdata->reset_gpio); } + } else if (!pdata && dev->dev.of_node) { + pdata = devm_kzalloc(&dev->dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return -ENOMEM; + pdata->reset_gpio = of_get_named_gpio(dev->dev.of_node, + "reset-gpios", 0); + if (pdata->reset_gpio == -ENOENT) + pdata->reset_gpio = -1; + else if (pdata->reset_gpio < 0) + return pdata->reset_gpio; + reset_gpio = pdata->reset_gpio; } else { if (cpu_is_pxa27x()) reset_gpio = 113; diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 4bc244c40f80..1f72672262d0 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -15,7 +15,7 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/dmaengine.h> -#include <linux/dma/pxa-dma.h> +#include <linux/dma-mapping.h> #include <sound/core.h> #include <sound/pcm.h> @@ -27,8 +27,6 @@ #include <mach/regs-ac97.h> #include <mach/audio.h> -#include "pxa2xx-pcm.h" - static void pxa2xx_ac97_legacy_reset(struct snd_ac97 *ac97) { if (!pxa2xx_ac97_try_cold_reset()) @@ -63,61 +61,46 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_legacy_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, -}; - -static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { - .addr = __PREG(PCDR), - .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, - .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_out_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, -}; - -static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { - .addr = __PREG(PCDR), - .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, - .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_in_req, -}; - static struct snd_pcm *pxa2xx_ac97_pcm; static struct snd_ac97 *pxa2xx_ac97_ac97; -static int pxa2xx_ac97_pcm_startup(struct snd_pcm_substream *substream) +static int pxa2xx_ac97_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; pxa2xx_audio_ops_t *platform_ops; - int r; + int ret, i; + + ret = pxa2xx_pcm_open(substream); + if (ret) + return ret; runtime->hw.channels_min = 2; runtime->hw.channels_max = 2; - r = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - AC97_RATES_FRONT_DAC : AC97_RATES_ADC; - runtime->hw.rates = pxa2xx_ac97_ac97->rates[r]; + i = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + AC97_RATES_FRONT_DAC : AC97_RATES_ADC; + runtime->hw.rates = pxa2xx_ac97_ac97->rates[i]; snd_pcm_limit_hw_rates(runtime); - platform_ops = substream->pcm->card->dev->platform_data; - if (platform_ops && platform_ops->startup) - return platform_ops->startup(substream, platform_ops->priv); - else - return 0; + platform_ops = substream->pcm->card->dev->platform_data; + if (platform_ops && platform_ops->startup) { + ret = platform_ops->startup(substream, platform_ops->priv); + if (ret < 0) + pxa2xx_pcm_close(substream); + } + + return ret; } -static void pxa2xx_ac97_pcm_shutdown(struct snd_pcm_substream *substream) +static int pxa2xx_ac97_pcm_close(struct snd_pcm_substream *substream) { pxa2xx_audio_ops_t *platform_ops; - platform_ops = substream->pcm->card->dev->platform_data; + platform_ops = substream->pcm->card->dev->platform_data; if (platform_ops && platform_ops->shutdown) platform_ops->shutdown(substream, platform_ops->priv); + + return 0; } static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream) @@ -125,17 +108,15 @@ static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; + int ret; + + ret = pxa2xx_pcm_prepare(substream); + if (ret < 0) + return ret; + return snd_ac97_set_rate(pxa2xx_ac97_ac97, reg, runtime->rate); } -static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = { - .playback_params = &pxa2xx_ac97_pcm_out, - .capture_params = &pxa2xx_ac97_pcm_in, - .startup = pxa2xx_ac97_pcm_startup, - .shutdown = pxa2xx_ac97_pcm_shutdown, - .prepare = pxa2xx_ac97_pcm_prepare, -}; - #ifdef CONFIG_PM_SLEEP static int pxa2xx_ac97_do_suspend(struct snd_card *card) @@ -193,6 +174,53 @@ static int pxa2xx_ac97_resume(struct device *dev) static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume); #endif +static const struct snd_pcm_ops pxa2xx_ac97_pcm_ops = { + .open = pxa2xx_ac97_pcm_open, + .close = pxa2xx_ac97_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = pxa2xx_pcm_hw_params, + .hw_free = pxa2xx_pcm_hw_free, + .prepare = pxa2xx_ac97_pcm_prepare, + .trigger = pxa2xx_pcm_trigger, + .pointer = pxa2xx_pcm_pointer, + .mmap = pxa2xx_pcm_mmap, +}; + + +static int pxa2xx_ac97_pcm_new(struct snd_card *card) +{ + struct snd_pcm *pcm; + int stream, ret; + + ret = snd_pcm_new(card, "PXA2xx-PCM", 0, 1, 1, &pcm); + if (ret) + goto out; + + pcm->private_free = pxa2xx_pcm_free_dma_buffers; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) + goto out; + + stream = SNDRV_PCM_STREAM_PLAYBACK; + snd_pcm_set_ops(pcm, stream, &pxa2xx_ac97_pcm_ops); + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); + if (ret) + goto out; + + stream = SNDRV_PCM_STREAM_CAPTURE; + snd_pcm_set_ops(pcm, stream, &pxa2xx_ac97_pcm_ops); + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); + if (ret) + goto out; + + pxa2xx_ac97_pcm = pcm; + ret = 0; + + out: + return ret; +} + static int pxa2xx_ac97_probe(struct platform_device *dev) { struct snd_card *card; @@ -214,7 +242,7 @@ static int pxa2xx_ac97_probe(struct platform_device *dev) strlcpy(card->driver, dev->dev.driver->name, sizeof(card->driver)); - ret = pxa2xx_pcm_new(card, &pxa2xx_ac97_pcm_client, &pxa2xx_ac97_pcm); + ret = pxa2xx_ac97_pcm_new(card); if (ret) goto err; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index e8da3b8ee721..7931789d4a9f 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -16,8 +16,6 @@ #include <sound/pxa2xx-lib.h> #include <sound/dmaengine_pcm.h> -#include "pxa2xx-pcm.h" - static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -25,8 +23,8 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, .period_bytes_max = 8192 - 32, .periods_min = 1, @@ -35,8 +33,8 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { .fifo_size = 32, }; -int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -64,14 +62,14 @@ int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -EXPORT_SYMBOL(__pxa2xx_pcm_hw_params); +EXPORT_SYMBOL(pxa2xx_pcm_hw_params); -int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) +int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { snd_pcm_set_runtime_buffer(substream, NULL); return 0; } -EXPORT_SYMBOL(__pxa2xx_pcm_hw_free); +EXPORT_SYMBOL(pxa2xx_pcm_hw_free); int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -86,13 +84,13 @@ pxa2xx_pcm_pointer(struct snd_pcm_substream *substream) } EXPORT_SYMBOL(pxa2xx_pcm_pointer); -int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) +int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) { return 0; } -EXPORT_SYMBOL(__pxa2xx_pcm_prepare); +EXPORT_SYMBOL(pxa2xx_pcm_prepare); -int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) +int pxa2xx_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; @@ -125,17 +123,17 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) return ret; - return snd_dmaengine_pcm_open_request_chan(substream, - pxad_filter_fn, - dma_params->filter_data); + return snd_dmaengine_pcm_open( + substream, dma_request_slave_channel(rtd->cpu_dai->dev, + dma_params->chan_name)); } -EXPORT_SYMBOL(__pxa2xx_pcm_open); +EXPORT_SYMBOL(pxa2xx_pcm_open); -int __pxa2xx_pcm_close(struct snd_pcm_substream *substream) +int pxa2xx_pcm_close(struct snd_pcm_substream *substream) { return snd_dmaengine_pcm_close_release_chan(substream); } -EXPORT_SYMBOL(__pxa2xx_pcm_close); +EXPORT_SYMBOL(pxa2xx_pcm_close); int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) @@ -181,6 +179,47 @@ void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm) } EXPORT_SYMBOL(pxa2xx_pcm_free_dma_buffers); +int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} +EXPORT_SYMBOL(pxa2xx_soc_pcm_new); + +const struct snd_pcm_ops pxa2xx_pcm_ops = { + .open = pxa2xx_pcm_open, + .close = pxa2xx_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = pxa2xx_pcm_hw_params, + .hw_free = pxa2xx_pcm_hw_free, + .prepare = pxa2xx_pcm_prepare, + .trigger = pxa2xx_pcm_trigger, + .pointer = pxa2xx_pcm_pointer, + .mmap = pxa2xx_pcm_mmap, +}; +EXPORT_SYMBOL(pxa2xx_pcm_ops); + MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("Intel PXA2xx sound library"); MODULE_LICENSE("GPL"); diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c deleted file mode 100644 index 1c6f4b436de3..000000000000 --- a/sound/arm/pxa2xx-pcm.c +++ /dev/null @@ -1,129 +0,0 @@ -/* - * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/dma-mapping.h> -#include <linux/dmaengine.h> - -#include <mach/dma.h> - -#include <sound/core.h> -#include <sound/pxa2xx-lib.h> -#include <sound/dmaengine_pcm.h> - -#include "pxa2xx-pcm.h" - -static int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct pxa2xx_pcm_client *client = substream->private_data; - - __pxa2xx_pcm_prepare(substream); - - return client->prepare(substream); -} - -static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) -{ - struct pxa2xx_pcm_client *client = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd; - int ret; - - ret = __pxa2xx_pcm_open(substream); - if (ret) - goto out; - - rtd = runtime->private_data; - - rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - client->playback_params : client->capture_params; - - ret = client->startup(substream); - if (!ret) - goto err2; - - return 0; - - err2: - __pxa2xx_pcm_close(substream); - out: - return ret; -} - -static int pxa2xx_pcm_close(struct snd_pcm_substream *substream) -{ - struct pxa2xx_pcm_client *client = substream->private_data; - - client->shutdown(substream); - - return __pxa2xx_pcm_close(substream); -} - -static const struct snd_pcm_ops pxa2xx_pcm_ops = { - .open = pxa2xx_pcm_open, - .close = pxa2xx_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = __pxa2xx_pcm_hw_params, - .hw_free = __pxa2xx_pcm_hw_free, - .prepare = pxa2xx_pcm_prepare, - .trigger = pxa2xx_pcm_trigger, - .pointer = pxa2xx_pcm_pointer, - .mmap = pxa2xx_pcm_mmap, -}; - -int pxa2xx_pcm_new(struct snd_card *card, struct pxa2xx_pcm_client *client, - struct snd_pcm **rpcm) -{ - struct snd_pcm *pcm; - int play = client->playback_params ? 1 : 0; - int capt = client->capture_params ? 1 : 0; - int ret; - - ret = snd_pcm_new(card, "PXA2xx-PCM", 0, play, capt, &pcm); - if (ret) - goto out; - - pcm->private_data = client; - pcm->private_free = pxa2xx_pcm_free_dma_buffers; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - goto out; - - if (play) { - int stream = SNDRV_PCM_STREAM_PLAYBACK; - snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); - if (ret) - goto out; - } - if (capt) { - int stream = SNDRV_PCM_STREAM_CAPTURE; - snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); - if (ret) - goto out; - } - - if (rpcm) - *rpcm = pcm; - ret = 0; - - out: - return ret; -} - -EXPORT_SYMBOL(pxa2xx_pcm_new); - -MODULE_AUTHOR("Nicolas Pitre"); -MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h deleted file mode 100644 index 8fa2b7c9e6b8..000000000000 --- a/sound/arm/pxa2xx-pcm.h +++ /dev/null @@ -1,27 +0,0 @@ -/* - * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -struct pxa2xx_runtime_data { - int dma_ch; - struct snd_dmaengine_dai_dma_data *params; -}; - -struct pxa2xx_pcm_client { - struct snd_dmaengine_dai_dma_data *playback_params; - struct snd_dmaengine_dai_dma_data *capture_params; - int (*startup)(struct snd_pcm_substream *); - void (*shutdown)(struct snd_pcm_substream *); - int (*prepare)(struct snd_pcm_substream *); -}; - -extern int pxa2xx_pcm_new(struct snd_card *, struct pxa2xx_pcm_client *, struct snd_pcm **); - diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 4b01a37c836e..26b5e245b074 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -1160,18 +1160,6 @@ int snd_compress_deregister(struct snd_compr *device) } EXPORT_SYMBOL_GPL(snd_compress_deregister); -static int __init snd_compress_init(void) -{ - return 0; -} - -static void __exit snd_compress_exit(void) -{ -} - -module_init(snd_compress_init); -module_exit(snd_compress_exit); - MODULE_DESCRIPTION("ALSA Compressed offload framework"); MODULE_AUTHOR("Vinod Koul <vinod.koul@linux.intel.com>"); MODULE_LICENSE("GPL v2"); diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 7f89d3c79a4b..753d5fc4b284 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -242,16 +242,12 @@ int snd_dma_alloc_pages_fallback(int type, struct device *device, size_t size, int err; while ((err = snd_dma_alloc_pages(type, device, size, dmab)) < 0) { - size_t aligned_size; if (err != -ENOMEM) return err; if (size <= PAGE_SIZE) return -ENOMEM; - aligned_size = PAGE_SIZE << get_order(size); - if (size != aligned_size) - size = aligned_size; - else - size >>= 1; + size >>= 1; + size = PAGE_SIZE << get_order(size); } if (! dmab->area) return -ENOMEM; diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 905a53c1cde5..f8d4a419f3af 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1851,7 +1851,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) format_mask = hw_param_mask_c(params, SNDRV_PCM_HW_PARAM_FORMAT); for (fmt = 0; fmt < 32; ++fmt) { if (snd_mask_test(format_mask, fmt)) { - int f = snd_pcm_oss_format_to(fmt); + int f = snd_pcm_oss_format_to((__force snd_pcm_format_t)fmt); if (f >= 0) formats |= f; } diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 85a56af104bd..0391cb1a4f19 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -281,10 +281,10 @@ static int snd_pcm_plug_formats(const struct snd_mask *mask, SNDRV_PCM_FMTBIT_U32_BE | SNDRV_PCM_FMTBIT_S32_BE); snd_mask_set(&formats, (__force int)SNDRV_PCM_FORMAT_MU_LAW); - if (formats.bits[0] & (u32)linfmts) - formats.bits[0] |= (u32)linfmts; - if (formats.bits[1] & (u32)(linfmts >> 32)) - formats.bits[1] |= (u32)(linfmts >> 32); + if (formats.bits[0] & lower_32_bits(linfmts)) + formats.bits[0] |= lower_32_bits(linfmts); + if (formats.bits[1] & upper_32_bits(linfmts)) + formats.bits[1] |= upper_32_bits(linfmts); return snd_mask_test(&formats, (__force int)format); } @@ -353,6 +353,7 @@ snd_pcm_format_t snd_pcm_plug_slave_format(snd_pcm_format_t format, if (snd_mask_test(format_mask, (__force int)format1)) return format1; } + /* fall through */ default: return (__force snd_pcm_format_t)-EINVAL; } diff --git a/sound/core/pcm.c b/sound/core/pcm.c index c352bfb973cc..fdb9b92fc8d6 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -492,13 +492,8 @@ static void snd_pcm_xrun_injection_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_pcm_substream *substream = entry->private_data; - struct snd_pcm_runtime *runtime; - snd_pcm_stream_lock_irq(substream); - runtime = substream->runtime; - if (runtime && runtime->status->state == SNDRV_PCM_STATE_RUNNING) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irq(substream); + snd_pcm_stop_xrun(substream); } static void snd_pcm_xrun_debug_read(struct snd_info_entry *entry, diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..4e6110d778bd 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -153,7 +153,8 @@ EXPORT_SYMBOL(snd_pcm_debug_name); dump_stack(); \ } while (0) -static void xrun(struct snd_pcm_substream *substream) +/* call with stream lock held */ +void __snd_pcm_xrun(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -201,7 +202,7 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, } } else { if (avail >= runtime->stop_threshold) { - xrun(substream); + __snd_pcm_xrun(substream); return -EPIPE; } } @@ -297,7 +298,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, } if (pos == SNDRV_PCM_POS_XRUN) { - xrun(substream); + __snd_pcm_xrun(substream); return -EPIPE; } if (pos >= runtime->buffer_size) { @@ -626,27 +627,33 @@ EXPORT_SYMBOL(snd_interval_refine); static int snd_interval_refine_first(struct snd_interval *i) { + const unsigned int last_max = i->max; + if (snd_BUG_ON(snd_interval_empty(i))) return -EINVAL; if (snd_interval_single(i)) return 0; i->max = i->min; - i->openmax = i->openmin; - if (i->openmax) + if (i->openmin) i->max++; + /* only exclude max value if also excluded before refine */ + i->openmax = (i->openmax && i->max >= last_max); return 1; } static int snd_interval_refine_last(struct snd_interval *i) { + const unsigned int last_min = i->min; + if (snd_BUG_ON(snd_interval_empty(i))) return -EINVAL; if (snd_interval_single(i)) return 0; i->min = i->max; - i->openmin = i->openmax; - if (i->openmin) + if (i->openmax) i->min--; + /* only exclude min value if also excluded before refine */ + i->openmin = (i->openmin && i->min <= last_min); return 1; } @@ -1832,12 +1839,19 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { - wait_time = 10; - if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; - wait_time = max(t, wait_time); + /* use wait time from substream if available */ + if (substream->wait_time) { + wait_time = substream->wait_time; + } else { + wait_time = 10; + + if (runtime->rate) { + long t = runtime->period_size * 2 / + runtime->rate; + wait_time = max(t, wait_time); + } + wait_time = msecs_to_jiffies(wait_time * 1000); } - wait_time = msecs_to_jiffies(wait_time * 1000); } for (;;) { diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index 7a499d02df6c..c515612969a4 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -65,4 +65,6 @@ static inline void snd_pcm_timer_init(struct snd_pcm_substream *substream) {} static inline void snd_pcm_timer_done(struct snd_pcm_substream *substream) {} #endif +void __snd_pcm_xrun(struct snd_pcm_substream *substream); + #endif /* __SOUND_CORE_PCM_LOCAL_H */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index cecc79772c94..66c90f486af9 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1337,13 +1337,12 @@ int snd_pcm_drain_done(struct snd_pcm_substream *substream) int snd_pcm_stop_xrun(struct snd_pcm_substream *substream) { unsigned long flags; - int ret = 0; snd_pcm_stream_lock_irqsave(substream, flags); - if (snd_pcm_running(substream)) - ret = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + if (substream->runtime && snd_pcm_running(substream)) + __snd_pcm_xrun(substream); snd_pcm_stream_unlock_irqrestore(substream, flags); - return ret; + return 0; } EXPORT_SYMBOL_GPL(snd_pcm_stop_xrun); @@ -1591,7 +1590,8 @@ static int snd_pcm_xrun(struct snd_pcm_substream *substream) result = 0; /* already there */ break; case SNDRV_PCM_STATE_RUNNING: - result = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + __snd_pcm_xrun(substream); + result = 0; break; default: result = -EBADFD; diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index b53026a72e73..69517e18ef07 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -29,6 +29,7 @@ #include <linux/mutex.h> #include <linux/module.h> #include <linux/delay.h> +#include <linux/mm.h> #include <sound/rawmidi.h> #include <sound/info.h> #include <sound/control.h> @@ -88,6 +89,7 @@ static inline unsigned short snd_rawmidi_file_flags(struct file *file) static inline int snd_rawmidi_ready(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime = substream->runtime; + return runtime->avail >= runtime->avail_min; } @@ -95,6 +97,7 @@ static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substre size_t count) { struct snd_rawmidi_runtime *runtime = substream->runtime; + return runtime->avail >= runtime->avail_min && (!substream->append || runtime->avail >= count); } @@ -103,6 +106,7 @@ static void snd_rawmidi_input_event_work(struct work_struct *work) { struct snd_rawmidi_runtime *runtime = container_of(work, struct snd_rawmidi_runtime, event_work); + if (runtime->event) runtime->event(runtime->substream); } @@ -111,7 +115,8 @@ static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime; - if ((runtime = kzalloc(sizeof(*runtime), GFP_KERNEL)) == NULL) + runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); + if (!runtime) return -ENOMEM; runtime->substream = substream; spin_lock_init(&runtime->lock); @@ -124,7 +129,8 @@ static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream) runtime->avail = 0; else runtime->avail = runtime->buffer_size; - if ((runtime->buffer = kmalloc(runtime->buffer_size, GFP_KERNEL)) == NULL) { + runtime->buffer = kvmalloc(runtime->buffer_size, GFP_KERNEL); + if (!runtime->buffer) { kfree(runtime); return -ENOMEM; } @@ -137,13 +143,13 @@ static int snd_rawmidi_runtime_free(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime = substream->runtime; - kfree(runtime->buffer); + kvfree(runtime->buffer); kfree(runtime); substream->runtime = NULL; return 0; } -static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *substream,int up) +static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *substream, int up) { if (!substream->opened) return; @@ -159,17 +165,28 @@ static void snd_rawmidi_input_trigger(struct snd_rawmidi_substream *substream, i cancel_work_sync(&substream->runtime->event_work); } -int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream) +static void __reset_runtime_ptrs(struct snd_rawmidi_runtime *runtime, + bool is_input) +{ + runtime->drain = 0; + runtime->appl_ptr = runtime->hw_ptr = 0; + runtime->avail = is_input ? 0 : runtime->buffer_size; +} + +static void reset_runtime_ptrs(struct snd_rawmidi_runtime *runtime, + bool is_input) { unsigned long flags; - struct snd_rawmidi_runtime *runtime = substream->runtime; - snd_rawmidi_output_trigger(substream, 0); - runtime->drain = 0; spin_lock_irqsave(&runtime->lock, flags); - runtime->appl_ptr = runtime->hw_ptr = 0; - runtime->avail = runtime->buffer_size; + __reset_runtime_ptrs(runtime, is_input); spin_unlock_irqrestore(&runtime->lock, flags); +} + +int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream) +{ + snd_rawmidi_output_trigger(substream, 0); + reset_runtime_ptrs(substream->runtime, false); return 0; } EXPORT_SYMBOL(snd_rawmidi_drop_output); @@ -208,15 +225,8 @@ EXPORT_SYMBOL(snd_rawmidi_drain_output); int snd_rawmidi_drain_input(struct snd_rawmidi_substream *substream) { - unsigned long flags; - struct snd_rawmidi_runtime *runtime = substream->runtime; - snd_rawmidi_input_trigger(substream, 0); - runtime->drain = 0; - spin_lock_irqsave(&runtime->lock, flags); - runtime->appl_ptr = runtime->hw_ptr = 0; - runtime->avail = 0; - spin_unlock_irqrestore(&runtime->lock, flags); + reset_runtime_ptrs(substream->runtime, true); return 0; } EXPORT_SYMBOL(snd_rawmidi_drain_input); @@ -330,25 +340,23 @@ static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode, /* called from sound/core/seq/seq_midi.c */ int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, - int mode, struct snd_rawmidi_file * rfile) + int mode, struct snd_rawmidi_file *rfile) { struct snd_rawmidi *rmidi; - int err; + int err = 0; if (snd_BUG_ON(!rfile)) return -EINVAL; mutex_lock(®ister_mutex); rmidi = snd_rawmidi_search(card, device); - if (rmidi == NULL) { - mutex_unlock(®ister_mutex); - return -ENODEV; - } - if (!try_module_get(rmidi->card->module)) { - mutex_unlock(®ister_mutex); - return -ENXIO; - } + if (!rmidi) + err = -ENODEV; + else if (!try_module_get(rmidi->card->module)) + err = -ENXIO; mutex_unlock(®ister_mutex); + if (err < 0) + return err; mutex_lock(&rmidi->open_mutex); err = rawmidi_open_priv(rmidi, subdevice, mode, rfile); @@ -370,7 +378,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) struct snd_rawmidi_file *rawmidi_file = NULL; wait_queue_entry_t wait; - if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK)) + if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK)) return -EINVAL; /* invalid combination */ err = nonseekable_open(inode, file); @@ -520,7 +528,7 @@ int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile) if (snd_BUG_ON(!rfile)) return -ENXIO; - + rmidi = rfile->rmidi; rawmidi_release_priv(rfile); module_put(rmidi->card->module); @@ -548,7 +556,7 @@ static int snd_rawmidi_info(struct snd_rawmidi_substream *substream, struct snd_rawmidi_info *info) { struct snd_rawmidi *rmidi; - + if (substream == NULL) return -ENODEV; rmidi = substream->rmidi; @@ -568,11 +576,13 @@ static int snd_rawmidi_info(struct snd_rawmidi_substream *substream, } static int snd_rawmidi_info_user(struct snd_rawmidi_substream *substream, - struct snd_rawmidi_info __user * _info) + struct snd_rawmidi_info __user *_info) { struct snd_rawmidi_info info; int err; - if ((err = snd_rawmidi_info(substream, &info)) < 0) + + err = snd_rawmidi_info(substream, &info); + if (err < 0) return err; if (copy_to_user(_info, &info, sizeof(struct snd_rawmidi_info))) return -EFAULT; @@ -619,85 +629,68 @@ static int snd_rawmidi_info_select_user(struct snd_card *card, { int err; struct snd_rawmidi_info info; + if (get_user(info.device, &_info->device)) return -EFAULT; if (get_user(info.stream, &_info->stream)) return -EFAULT; if (get_user(info.subdevice, &_info->subdevice)) return -EFAULT; - if ((err = snd_rawmidi_info_select(card, &info)) < 0) + err = snd_rawmidi_info_select(card, &info); + if (err < 0) return err; if (copy_to_user(_info, &info, sizeof(struct snd_rawmidi_info))) return -EFAULT; return 0; } -int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, - struct snd_rawmidi_params * params) +static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, + struct snd_rawmidi_params *params, + bool is_input) { char *newbuf, *oldbuf; - struct snd_rawmidi_runtime *runtime = substream->runtime; - - if (substream->append && substream->use_count > 1) - return -EBUSY; - snd_rawmidi_drain_output(substream); - if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) { + + if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) return -EINVAL; - } - if (params->avail_min < 1 || params->avail_min > params->buffer_size) { + if (params->avail_min < 1 || params->avail_min > params->buffer_size) return -EINVAL; - } if (params->buffer_size != runtime->buffer_size) { - newbuf = kmalloc(params->buffer_size, GFP_KERNEL); + newbuf = kvmalloc(params->buffer_size, GFP_KERNEL); if (!newbuf) return -ENOMEM; spin_lock_irq(&runtime->lock); oldbuf = runtime->buffer; runtime->buffer = newbuf; runtime->buffer_size = params->buffer_size; - runtime->avail = runtime->buffer_size; - runtime->appl_ptr = runtime->hw_ptr = 0; + __reset_runtime_ptrs(runtime, is_input); spin_unlock_irq(&runtime->lock); - kfree(oldbuf); + kvfree(oldbuf); } runtime->avail_min = params->avail_min; - substream->active_sensing = !params->no_active_sensing; return 0; } + +int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, + struct snd_rawmidi_params *params) +{ + if (substream->append && substream->use_count > 1) + return -EBUSY; + snd_rawmidi_drain_output(substream); + substream->active_sensing = !params->no_active_sensing; + return resize_runtime_buffer(substream->runtime, params, false); +} EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, - struct snd_rawmidi_params * params) + struct snd_rawmidi_params *params) { - char *newbuf, *oldbuf; - struct snd_rawmidi_runtime *runtime = substream->runtime; - snd_rawmidi_drain_input(substream); - if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) { - return -EINVAL; - } - if (params->avail_min < 1 || params->avail_min > params->buffer_size) { - return -EINVAL; - } - if (params->buffer_size != runtime->buffer_size) { - newbuf = kmalloc(params->buffer_size, GFP_KERNEL); - if (!newbuf) - return -ENOMEM; - spin_lock_irq(&runtime->lock); - oldbuf = runtime->buffer; - runtime->buffer = newbuf; - runtime->buffer_size = params->buffer_size; - runtime->appl_ptr = runtime->hw_ptr = 0; - spin_unlock_irq(&runtime->lock); - kfree(oldbuf); - } - runtime->avail_min = params->avail_min; - return 0; + return resize_runtime_buffer(substream->runtime, params, true); } EXPORT_SYMBOL(snd_rawmidi_input_params); static int snd_rawmidi_output_status(struct snd_rawmidi_substream *substream, - struct snd_rawmidi_status * status) + struct snd_rawmidi_status *status) { struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -710,7 +703,7 @@ static int snd_rawmidi_output_status(struct snd_rawmidi_substream *substream, } static int snd_rawmidi_input_status(struct snd_rawmidi_substream *substream, - struct snd_rawmidi_status * status) + struct snd_rawmidi_status *status) { struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -739,6 +732,7 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long { int stream; struct snd_rawmidi_info __user *info = argp; + if (get_user(stream, &info->stream)) return -EFAULT; switch (stream) { @@ -753,6 +747,7 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long case SNDRV_RAWMIDI_IOCTL_PARAMS: { struct snd_rawmidi_params params; + if (copy_from_user(¶ms, argp, sizeof(struct snd_rawmidi_params))) return -EFAULT; switch (params.stream) { @@ -772,6 +767,7 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long { int err = 0; struct snd_rawmidi_status status; + if (copy_from_user(&status, argp, sizeof(struct snd_rawmidi_status))) return -EFAULT; switch (status.stream) { @@ -797,6 +793,7 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long case SNDRV_RAWMIDI_IOCTL_DROP: { int val; + if (get_user(val, (int __user *) argp)) return -EFAULT; switch (val) { @@ -811,6 +808,7 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long case SNDRV_RAWMIDI_IOCTL_DRAIN: { int val; + if (get_user(val, (int __user *) argp)) return -EFAULT; switch (val) { @@ -844,7 +842,7 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, case SNDRV_CTL_IOCTL_RAWMIDI_NEXT_DEVICE: { int device; - + if (get_user(device, (int __user *)argp)) return -EFAULT; if (device >= SNDRV_RAWMIDI_DEVICES) /* next device is -1 */ @@ -866,7 +864,7 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, case SNDRV_CTL_IOCTL_RAWMIDI_PREFER_SUBDEVICE: { int val; - + if (get_user(val, (int __user *)argp)) return -EFAULT; control->preferred_subdevice[SND_CTL_SUBDEV_RAWMIDI] = val; @@ -1020,6 +1018,7 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun spin_lock_irq(&runtime->lock); while (!snd_rawmidi_ready(substream)) { wait_queue_entry_t wait; + if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) { spin_unlock_irq(&runtime->lock); return result > 0 ? result : -EAGAIN; @@ -1072,7 +1071,7 @@ int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream) spin_lock_irqsave(&runtime->lock, flags); result = runtime->avail >= runtime->buffer_size; spin_unlock_irqrestore(&runtime->lock, flags); - return result; + return result; } EXPORT_SYMBOL(snd_rawmidi_transmit_empty); @@ -1210,7 +1209,7 @@ EXPORT_SYMBOL(snd_rawmidi_transmit_ack); * @substream: the rawmidi substream * @buffer: the buffer pointer * @count: the data size to transfer - * + * * Copies data from the buffer to the device and advances the pointer. * * Return: The copied size if successful, or a negative error code on failure. @@ -1324,6 +1323,7 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, spin_lock_irq(&runtime->lock); while (!snd_rawmidi_ready_append(substream, count)) { wait_queue_entry_t wait; + if (file->f_flags & O_NONBLOCK) { spin_unlock_irq(&runtime->lock); return result > 0 ? result : -EAGAIN; @@ -1357,6 +1357,7 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, while (runtime->avail != runtime->buffer_size) { wait_queue_entry_t wait; unsigned int last_avail = runtime->avail; + init_waitqueue_entry(&wait, current); add_wait_queue(&runtime->sleep, &wait); set_current_state(TASK_INTERRUPTIBLE); @@ -1374,7 +1375,7 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, return result; } -static __poll_t snd_rawmidi_poll(struct file *file, poll_table * wait) +static __poll_t snd_rawmidi_poll(struct file *file, poll_table *wait) { struct snd_rawmidi_file *rfile; struct snd_rawmidi_runtime *runtime; @@ -1411,7 +1412,6 @@ static __poll_t snd_rawmidi_poll(struct file *file, poll_table * wait) #endif /* - */ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, @@ -1479,8 +1479,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, * Register functions */ -static const struct file_operations snd_rawmidi_f_ops = -{ +static const struct file_operations snd_rawmidi_f_ops = { .owner = THIS_MODULE, .read = snd_rawmidi_read, .write = snd_rawmidi_write, @@ -1535,7 +1534,7 @@ static void release_rawmidi_device(struct device *dev) */ int snd_rawmidi_new(struct snd_card *card, char *id, int device, int output_count, int input_count, - struct snd_rawmidi ** rrawmidi) + struct snd_rawmidi **rrawmidi) { struct snd_rawmidi *rmidi; int err; @@ -1566,27 +1565,29 @@ int snd_rawmidi_new(struct snd_card *card, char *id, int device, rmidi->dev.release = release_rawmidi_device; dev_set_name(&rmidi->dev, "midiC%iD%i", card->number, device); - if ((err = snd_rawmidi_alloc_substreams(rmidi, - &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT], - SNDRV_RAWMIDI_STREAM_INPUT, - input_count)) < 0) { - snd_rawmidi_free(rmidi); - return err; - } - if ((err = snd_rawmidi_alloc_substreams(rmidi, - &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT], - SNDRV_RAWMIDI_STREAM_OUTPUT, - output_count)) < 0) { - snd_rawmidi_free(rmidi); - return err; - } - if ((err = snd_device_new(card, SNDRV_DEV_RAWMIDI, rmidi, &ops)) < 0) { - snd_rawmidi_free(rmidi); - return err; - } + err = snd_rawmidi_alloc_substreams(rmidi, + &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT], + SNDRV_RAWMIDI_STREAM_INPUT, + input_count); + if (err < 0) + goto error; + err = snd_rawmidi_alloc_substreams(rmidi, + &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT], + SNDRV_RAWMIDI_STREAM_OUTPUT, + output_count); + if (err < 0) + goto error; + err = snd_device_new(card, SNDRV_DEV_RAWMIDI, rmidi, &ops); + if (err < 0) + goto error; + if (rrawmidi) *rrawmidi = rmidi; return 0; + + error: + snd_rawmidi_free(rmidi); + return err; } EXPORT_SYMBOL(snd_rawmidi_new); @@ -1624,6 +1625,7 @@ static int snd_rawmidi_free(struct snd_rawmidi *rmidi) static int snd_rawmidi_dev_free(struct snd_device *device) { struct snd_rawmidi *rmidi = device->device_data; + return snd_rawmidi_free(rmidi); } @@ -1631,6 +1633,7 @@ static int snd_rawmidi_dev_free(struct snd_device *device) static void snd_rawmidi_dev_seq_free(struct snd_seq_device *device) { struct snd_rawmidi *rmidi = device->private_data; + rmidi->seq_dev = NULL; } #endif @@ -1644,30 +1647,27 @@ static int snd_rawmidi_dev_register(struct snd_device *device) if (rmidi->device >= SNDRV_RAWMIDI_DEVICES) return -ENOMEM; + err = 0; mutex_lock(®ister_mutex); - if (snd_rawmidi_search(rmidi->card, rmidi->device)) { - mutex_unlock(®ister_mutex); - return -EBUSY; - } - list_add_tail(&rmidi->list, &snd_rawmidi_devices); + if (snd_rawmidi_search(rmidi->card, rmidi->device)) + err = -EBUSY; + else + list_add_tail(&rmidi->list, &snd_rawmidi_devices); mutex_unlock(®ister_mutex); + if (err < 0) + return err; + err = snd_register_device(SNDRV_DEVICE_TYPE_RAWMIDI, rmidi->card, rmidi->device, &snd_rawmidi_f_ops, rmidi, &rmidi->dev); if (err < 0) { rmidi_err(rmidi, "unable to register\n"); - mutex_lock(®ister_mutex); - list_del(&rmidi->list); - mutex_unlock(®ister_mutex); - return err; + goto error; } - if (rmidi->ops && rmidi->ops->dev_register && - (err = rmidi->ops->dev_register(rmidi)) < 0) { - snd_unregister_device(&rmidi->dev); - mutex_lock(®ister_mutex); - list_del(&rmidi->list); - mutex_unlock(®ister_mutex); - return err; + if (rmidi->ops && rmidi->ops->dev_register) { + err = rmidi->ops->dev_register(rmidi); + if (err < 0) + goto error_unregister; } #ifdef CONFIG_SND_OSSEMUL rmidi->ossreg = 0; @@ -1719,6 +1719,14 @@ static int snd_rawmidi_dev_register(struct snd_device *device) } #endif return 0; + + error_unregister: + snd_unregister_device(&rmidi->dev); + error: + mutex_lock(®ister_mutex); + list_del(&rmidi->list); + mutex_unlock(®ister_mutex); + return err; } static int snd_rawmidi_dev_disconnect(struct snd_device *device) @@ -1732,6 +1740,7 @@ static int snd_rawmidi_dev_disconnect(struct snd_device *device) list_del_init(&rmidi->list); for (dir = 0; dir < 2; dir++) { struct snd_rawmidi_substream *s; + list_for_each_entry(s, &rmidi->streams[dir].substreams, list) { if (s->runtime) wake_up(&s->runtime->sleep); @@ -1769,7 +1778,7 @@ void snd_rawmidi_set_ops(struct snd_rawmidi *rmidi, int stream, const struct snd_rawmidi_ops *ops) { struct snd_rawmidi_substream *substream; - + list_for_each_entry(substream, &rmidi->streams[stream].substreams, list) substream->ops = ops; } diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index 5f64d0d88320..e1f44fc86885 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -203,7 +203,7 @@ odev_poll(struct file *file, poll_table * wait) struct seq_oss_devinfo *dp; dp = file->private_data; if (snd_BUG_ON(!dp)) - return -ENXIO; + return EPOLLERR; return snd_seq_oss_poll(dp, file, wait); } diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 9debd1b8fd28..0d5f8b16d057 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -637,7 +637,7 @@ snd_seq_oss_midi_putc(struct seq_oss_devinfo *dp, int dev, unsigned char c, stru if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; - if (snd_midi_event_encode_byte(mdev->coder, c, ev) > 0) { + if (snd_midi_event_encode_byte(mdev->coder, c, ev)) { snd_seq_oss_fill_addr(dp, ev, mdev->client, mdev->port); snd_use_lock_free(&mdev->use_lock); return 0; diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c index 4f24ea9fad93..ba127c22539a 100644 --- a/sound/core/seq/oss/seq_oss_timer.c +++ b/sound/core/seq/oss/seq_oss_timer.c @@ -92,7 +92,7 @@ snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *ev) case TMR_WAIT_REL: parm += rec->cur_tick; rec->realtime = 0; - /* continue to next */ + /* fall through and continue to next */ case TMR_WAIT_ABS: if (parm == 0) { rec->realtime = 1; diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index 639544b4fb04..7de98d71f2aa 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -84,30 +84,32 @@ static int __init alsa_seq_init(void) { int err; - if ((err = client_init_data()) < 0) - goto error; - - /* init memory, room for selected events */ - if ((err = snd_sequencer_memory_init()) < 0) - goto error; - - /* init event queues */ - if ((err = snd_seq_queues_init()) < 0) + err = client_init_data(); + if (err < 0) goto error; /* register sequencer device */ - if ((err = snd_sequencer_device_init()) < 0) + err = snd_sequencer_device_init(); + if (err < 0) goto error; /* register proc interface */ - if ((err = snd_seq_info_init()) < 0) - goto error; + err = snd_seq_info_init(); + if (err < 0) + goto error_device; /* register our internal client */ - if ((err = snd_seq_system_client_init()) < 0) - goto error; + err = snd_seq_system_client_init(); + if (err < 0) + goto error_info; snd_seq_autoload_init(); + return 0; + + error_info: + snd_seq_info_done(); + error_device: + snd_sequencer_device_done(); error: return err; } @@ -126,9 +128,6 @@ static void __exit alsa_seq_exit(void) /* unregister sequencer device */ snd_sequencer_device_done(); - /* release event memory */ - snd_sequencer_memory_done(); - snd_seq_autoload_exit(); } diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 56ca78423040..92e6524a3a9d 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -311,10 +311,9 @@ static int snd_seq_open(struct inode *inode, struct file *file) if (err < 0) return err; - if (mutex_lock_interruptible(®ister_mutex)) - return -ERESTARTSYS; + mutex_lock(®ister_mutex); client = seq_create_client1(-1, SNDRV_SEQ_DEFAULT_EVENTS); - if (client == NULL) { + if (!client) { mutex_unlock(®ister_mutex); return -ENOMEM; /* failure code */ } @@ -1101,7 +1100,7 @@ static __poll_t snd_seq_poll(struct file *file, poll_table * wait) /* check client structures are in place */ if (snd_BUG_ON(!client)) - return -ENXIO; + return EPOLLERR; if ((snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_INPUT) && client->data.user.fifo) { @@ -1704,10 +1703,7 @@ static int snd_seq_ioctl_get_queue_timer(struct snd_seq_client *client, if (queue == NULL) return -EINVAL; - if (mutex_lock_interruptible(&queue->timer_mutex)) { - queuefree(queue); - return -ERESTARTSYS; - } + mutex_lock(&queue->timer_mutex); tmr = queue->timer; memset(timer, 0, sizeof(*timer)); timer->queue = queue->queue; @@ -1741,10 +1737,7 @@ static int snd_seq_ioctl_set_queue_timer(struct snd_seq_client *client, q = queueptr(timer->queue); if (q == NULL) return -ENXIO; - if (mutex_lock_interruptible(&q->timer_mutex)) { - queuefree(q); - return -ERESTARTSYS; - } + mutex_lock(&q->timer_mutex); tmr = q->timer; snd_seq_queue_timer_close(timer->queue); tmr->type = timer->type; @@ -2180,8 +2173,7 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index, if (card == NULL && client_index >= SNDRV_SEQ_GLOBAL_CLIENTS) return -EINVAL; - if (mutex_lock_interruptible(®ister_mutex)) - return -ERESTARTSYS; + mutex_lock(®ister_mutex); if (card) { client_index += SNDRV_SEQ_GLOBAL_CLIENTS @@ -2522,19 +2514,15 @@ int __init snd_sequencer_device_init(void) snd_device_initialize(&seq_dev, NULL); dev_set_name(&seq_dev, "seq"); - if (mutex_lock_interruptible(®ister_mutex)) - return -ERESTARTSYS; - + mutex_lock(®ister_mutex); err = snd_register_device(SNDRV_DEVICE_TYPE_SEQUENCER, NULL, 0, &snd_seq_f_ops, NULL, &seq_dev); + mutex_unlock(®ister_mutex); if (err < 0) { - mutex_unlock(®ister_mutex); put_device(&seq_dev); return err; } - mutex_unlock(®ister_mutex); - return 0; } @@ -2543,7 +2531,7 @@ int __init snd_sequencer_device_init(void) /* * unregister sequencer device */ -void __exit snd_sequencer_device_done(void) +void snd_sequencer_device_done(void) { snd_unregister_device(&seq_dev); put_device(&seq_dev); diff --git a/sound/core/seq/seq_info.c b/sound/core/seq/seq_info.c index 97015447b9b3..b27fedd435b6 100644 --- a/sound/core/seq/seq_info.c +++ b/sound/core/seq/seq_info.c @@ -50,7 +50,7 @@ create_info_entry(char *name, void (*read)(struct snd_info_entry *, return entry; } -static void free_info_entries(void) +void snd_seq_info_done(void) { snd_info_free_entry(queues_entry); snd_info_free_entry(clients_entry); @@ -70,12 +70,6 @@ int __init snd_seq_info_init(void) return 0; error: - free_info_entries(); + snd_seq_info_done(); return -ENOMEM; } - -int __exit snd_seq_info_done(void) -{ - free_info_entries(); - return 0; -} diff --git a/sound/core/seq/seq_info.h b/sound/core/seq/seq_info.h index f8549f81a645..2cdf8f6e63f5 100644 --- a/sound/core/seq/seq_info.h +++ b/sound/core/seq/seq_info.h @@ -30,11 +30,11 @@ void snd_seq_info_queues_read(struct snd_info_entry *entry, struct snd_info_buff #ifdef CONFIG_SND_PROC_FS -int snd_seq_info_init( void ); -int snd_seq_info_done( void ); +int snd_seq_info_init(void); +void snd_seq_info_done(void); #else static inline int snd_seq_info_init(void) { return 0; } -static inline int snd_seq_info_done(void) { return 0; } +static inline void snd_seq_info_done(void) {} #endif #endif diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index a4c8543176b2..5b0388202bac 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -504,18 +504,6 @@ int snd_seq_pool_delete(struct snd_seq_pool **ppool) return 0; } -/* initialize sequencer memory */ -int __init snd_sequencer_memory_init(void) -{ - return 0; -} - -/* release sequencer memory */ -void __exit snd_sequencer_memory_done(void) -{ -} - - /* exported to seq_clientmgr.c */ void snd_seq_info_pool(struct snd_info_buffer *buffer, struct snd_seq_pool *pool, char *space) diff --git a/sound/core/seq/seq_memory.h b/sound/core/seq/seq_memory.h index 3abe306c394a..1292fe91f02e 100644 --- a/sound/core/seq/seq_memory.h +++ b/sound/core/seq/seq_memory.h @@ -94,12 +94,6 @@ struct snd_seq_pool *snd_seq_pool_new(int poolsize); /* remove pool */ int snd_seq_pool_delete(struct snd_seq_pool **pool); -/* init memory */ -int snd_sequencer_memory_init(void); - -/* release event memory */ -void snd_sequencer_memory_done(void); - /* polling */ int snd_seq_pool_poll_wait(struct snd_seq_pool *pool, struct file *file, poll_table *wait); diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 5dd0ee258359..9e0dabd3ce5f 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -78,7 +78,7 @@ static void snd_midi_input_event(struct snd_rawmidi_substream *substream) struct seq_midisynth *msynth; struct snd_seq_event ev; char buf[16], *pbuf; - long res, count; + long res; if (substream == NULL) return; @@ -94,19 +94,15 @@ static void snd_midi_input_event(struct snd_rawmidi_substream *substream) if (msynth->parser == NULL) continue; pbuf = buf; - while (res > 0) { - count = snd_midi_event_encode(msynth->parser, pbuf, res, &ev); - if (count < 0) - break; - pbuf += count; - res -= count; - if (ev.type != SNDRV_SEQ_EVENT_NONE) { - ev.source.port = msynth->seq_port; - ev.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS; - snd_seq_kernel_client_dispatch(msynth->seq_client, &ev, 1, 0); - /* clear event and reset header */ - memset(&ev, 0, sizeof(ev)); - } + while (res-- > 0) { + if (!snd_midi_event_encode_byte(msynth->parser, + *pbuf++, &ev)) + continue; + ev.source.port = msynth->seq_port; + ev.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS; + snd_seq_kernel_client_dispatch(msynth->seq_client, &ev, 1, 0); + /* clear event and reset header */ + memset(&ev, 0, sizeof(ev)); } } } diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index 288f839a554b..c1975dd31871 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -318,7 +318,7 @@ do_control(struct snd_midi_op *ops, void *drv, struct snd_midi_channel_set *chse break; case MIDI_CTL_MSB_DATA_ENTRY: chan->control[MIDI_CTL_LSB_DATA_ENTRY] = 0; - /* go through here */ + /* fall through */ case MIDI_CTL_LSB_DATA_ENTRY: if (chan->param_type == SNDRV_MIDI_PARAM_TYPE_REGISTERED) rpn(ops, drv, chan, chset); @@ -728,15 +728,3 @@ void snd_midi_channel_free_set(struct snd_midi_channel_set *chset) kfree(chset); } EXPORT_SYMBOL(snd_midi_channel_free_set); - -static int __init alsa_seq_midi_emul_init(void) -{ - return 0; -} - -static void __exit alsa_seq_midi_emul_exit(void) -{ -} - -module_init(alsa_seq_midi_emul_init) -module_exit(alsa_seq_midi_emul_exit) diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c index 90bbbdbeba03..b11419537062 100644 --- a/sound/core/seq/seq_midi_event.c +++ b/sound/core/seq/seq_midi_event.c @@ -175,14 +175,6 @@ void snd_midi_event_reset_decode(struct snd_midi_event *dev) } EXPORT_SYMBOL(snd_midi_event_reset_decode); -#if 0 -void snd_midi_event_init(struct snd_midi_event *dev) -{ - snd_midi_event_reset_encode(dev); - snd_midi_event_reset_decode(dev); -} -#endif /* 0 */ - void snd_midi_event_no_status(struct snd_midi_event *dev, int on) { dev->nostat = on ? 1 : 0; @@ -190,69 +182,16 @@ void snd_midi_event_no_status(struct snd_midi_event *dev, int on) EXPORT_SYMBOL(snd_midi_event_no_status); /* - * resize buffer - */ -#if 0 -int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) -{ - unsigned char *new_buf, *old_buf; - unsigned long flags; - - if (bufsize == dev->bufsize) - return 0; - new_buf = kmalloc(bufsize, GFP_KERNEL); - if (new_buf == NULL) - return -ENOMEM; - spin_lock_irqsave(&dev->lock, flags); - old_buf = dev->buf; - dev->buf = new_buf; - dev->bufsize = bufsize; - reset_encode(dev); - spin_unlock_irqrestore(&dev->lock, flags); - kfree(old_buf); - return 0; -} -#endif /* 0 */ - -/* - * read bytes and encode to sequencer event if finished - * return the size of encoded bytes - */ -long snd_midi_event_encode(struct snd_midi_event *dev, unsigned char *buf, long count, - struct snd_seq_event *ev) -{ - long result = 0; - int rc; - - ev->type = SNDRV_SEQ_EVENT_NONE; - - while (count-- > 0) { - rc = snd_midi_event_encode_byte(dev, *buf++, ev); - result++; - if (rc < 0) - return rc; - else if (rc > 0) - return result; - } - - return result; -} -EXPORT_SYMBOL(snd_midi_event_encode); - -/* * read one byte and encode to sequencer event: - * return 1 if MIDI bytes are encoded to an event - * 0 data is not finished - * negative for error + * return true if MIDI bytes are encoded to an event + * false data is not finished */ -int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, - struct snd_seq_event *ev) +bool snd_midi_event_encode_byte(struct snd_midi_event *dev, unsigned char c, + struct snd_seq_event *ev) { - int rc = 0; + bool rc = false; unsigned long flags; - c &= 0xff; - if (c >= MIDI_CMD_COMMON_CLOCK) { /* real-time event */ ev->type = status_event[ST_SPECIAL + c - 0xf0].event; @@ -293,7 +232,7 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, status_event[dev->type].encode(dev, ev); if (dev->type >= ST_SPECIAL) dev->type = ST_INVALID; - rc = 1; + rc = true; } else if (dev->type == ST_SYSEX) { if (c == MIDI_CMD_COMMON_SYSEX_END || dev->read >= dev->bufsize) { @@ -306,7 +245,7 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, dev->read = 0; /* continue to parse */ else reset_encode(dev); /* all parsed */ - rc = 1; + rc = true; } } @@ -531,15 +470,3 @@ static int extra_decode_xrpn(struct snd_midi_event *dev, unsigned char *buf, } return idx; } - -static int __init alsa_seq_midi_event_init(void) -{ - return 0; -} - -static void __exit alsa_seq_midi_event_exit(void) -{ -} - -module_init(alsa_seq_midi_event_init) -module_exit(alsa_seq_midi_event_exit) diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index b377f5048352..3b3ac96f1f5f 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -159,18 +159,8 @@ static void queue_delete(struct snd_seq_queue *q) /*----------------------------------------------------------------*/ -/* setup queues */ -int __init snd_seq_queues_init(void) -{ - /* - memset(queue_list, 0, sizeof(queue_list)); - num_queues = 0; - */ - return 0; -} - /* delete all existing queues */ -void __exit snd_seq_queues_delete(void) +void snd_seq_queues_delete(void) { int i; diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h index 719093489a2c..e006fc8e3a36 100644 --- a/sound/core/seq/seq_queue.h +++ b/sound/core/seq/seq_queue.h @@ -63,9 +63,6 @@ struct snd_seq_queue { /* get the number of current queues */ int snd_seq_queue_get_cur_queues(void); -/* init queues structure */ -int snd_seq_queues_init(void); - /* delete queues */ void snd_seq_queues_delete(void); @@ -112,28 +109,4 @@ int snd_seq_queue_is_used(int queueid, int client); int snd_seq_control_queue(struct snd_seq_event *ev, int atomic, int hop); -/* - * 64bit division - for sync stuff.. - */ -#if defined(i386) || defined(i486) - -#define udiv_qrnnd(q, r, n1, n0, d) \ - __asm__ ("divl %4" \ - : "=a" ((u32)(q)), \ - "=d" ((u32)(r)) \ - : "0" ((u32)(n0)), \ - "1" ((u32)(n1)), \ - "rm" ((u32)(d))) - -#define u64_div(x,y,q) do {u32 __tmp; udiv_qrnnd(q, __tmp, (x)>>32, x, y);} while (0) -#define u64_mod(x,y,r) do {u32 __tmp; udiv_qrnnd(__tmp, q, (x)>>32, x, y);} while (0) -#define u64_divmod(x,y,q,r) udiv_qrnnd(q, r, (x)>>32, x, y) - -#else -#define u64_div(x,y,q) ((q) = (u32)((u64)(x) / (u64)(y))) -#define u64_mod(x,y,r) ((r) = (u32)((u64)(x) % (u64)(y))) -#define u64_divmod(x,y,q,r) (u64_div(x,y,q), u64_mod(x,y,r)) -#endif - - #endif diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index 289ae6bb81d9..a2f1c6b58693 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -89,7 +89,7 @@ static int snd_virmidi_dev_receive_event(struct snd_virmidi_dev *rdev, else down_read(&rdev->filelist_sem); list_for_each_entry(vmidi, &rdev->filelist, list) { - if (!vmidi->trigger) + if (!READ_ONCE(vmidi->trigger)) continue; if (ev->type == SNDRV_SEQ_EVENT_SYSEX) { if ((ev->flags & SNDRV_SEQ_EVENT_LENGTH_MASK) != SNDRV_SEQ_EVENT_LENGTH_VARIABLE) @@ -110,23 +110,6 @@ static int snd_virmidi_dev_receive_event(struct snd_virmidi_dev *rdev, } /* - * receive an event from the remote virmidi port - * - * for rawmidi inputs, you can call this function from the event - * handler of a remote port which is attached to the virmidi via - * SNDRV_VIRMIDI_SEQ_ATTACH. - */ -#if 0 -int snd_virmidi_receive(struct snd_rawmidi *rmidi, struct snd_seq_event *ev) -{ - struct snd_virmidi_dev *rdev; - - rdev = rmidi->private_data; - return snd_virmidi_dev_receive_event(rdev, ev, true); -} -#endif /* 0 */ - -/* * event handler of virmidi port */ static int snd_virmidi_event_input(struct snd_seq_event *ev, int direct, @@ -147,68 +130,62 @@ static void snd_virmidi_input_trigger(struct snd_rawmidi_substream *substream, i { struct snd_virmidi *vmidi = substream->runtime->private_data; - if (up) { - vmidi->trigger = 1; - } else { - vmidi->trigger = 0; - } + WRITE_ONCE(vmidi->trigger, !!up); } -/* - * trigger rawmidi stream for output +/* process rawmidi bytes and send events; + * we need no lock here for vmidi->event since it's handled only in this work */ -static void snd_virmidi_output_trigger(struct snd_rawmidi_substream *substream, int up) +static void snd_vmidi_output_work(struct work_struct *work) { - struct snd_virmidi *vmidi = substream->runtime->private_data; - int count, res; - unsigned char buf[32], *pbuf; - unsigned long flags; - - if (up) { - vmidi->trigger = 1; - if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH && - !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) { - while (snd_rawmidi_transmit(substream, buf, - sizeof(buf)) > 0) { - /* ignored */ - } - return; - } - spin_lock_irqsave(&substream->runtime->lock, flags); + struct snd_virmidi *vmidi; + struct snd_rawmidi_substream *substream; + unsigned char input; + int ret; + + vmidi = container_of(work, struct snd_virmidi, output_work); + substream = vmidi->substream; + + /* discard the outputs in dispatch mode unless subscribed */ + if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH && + !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) { + while (!snd_rawmidi_transmit_empty(substream)) + snd_rawmidi_transmit_ack(substream, 1); + return; + } + + while (READ_ONCE(vmidi->trigger)) { + if (snd_rawmidi_transmit(substream, &input, 1) != 1) + break; + if (!snd_midi_event_encode_byte(vmidi->parser, input, + &vmidi->event)) + continue; if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) { - if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0) - goto out; + ret = snd_seq_kernel_client_dispatch(vmidi->client, + &vmidi->event, + false, 0); vmidi->event.type = SNDRV_SEQ_EVENT_NONE; - } - while (1) { - count = __snd_rawmidi_transmit_peek(substream, buf, sizeof(buf)); - if (count <= 0) + if (ret < 0) break; - pbuf = buf; - while (count > 0) { - res = snd_midi_event_encode(vmidi->parser, pbuf, count, &vmidi->event); - if (res < 0) { - snd_midi_event_reset_encode(vmidi->parser); - continue; - } - __snd_rawmidi_transmit_ack(substream, res); - pbuf += res; - count -= res; - if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) { - if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0) - goto out; - vmidi->event.type = SNDRV_SEQ_EVENT_NONE; - } - } } - out: - spin_unlock_irqrestore(&substream->runtime->lock, flags); - } else { - vmidi->trigger = 0; + /* rawmidi input might be huge, allow to have a break */ + cond_resched(); } } /* + * trigger rawmidi stream for output + */ +static void snd_virmidi_output_trigger(struct snd_rawmidi_substream *substream, int up) +{ + struct snd_virmidi *vmidi = substream->runtime->private_data; + + WRITE_ONCE(vmidi->trigger, !!up); + if (up) + queue_work(system_highpri_wq, &vmidi->output_work); +} + +/* * open rawmidi handle for input */ static int snd_virmidi_input_open(struct snd_rawmidi_substream *substream) @@ -260,6 +237,7 @@ static int snd_virmidi_output_open(struct snd_rawmidi_substream *substream) vmidi->port = rdev->port; snd_virmidi_init_event(vmidi, &vmidi->event); vmidi->rdev = rdev; + INIT_WORK(&vmidi->output_work, snd_vmidi_output_work); runtime->private_data = vmidi; return 0; } @@ -289,6 +267,9 @@ static int snd_virmidi_input_close(struct snd_rawmidi_substream *substream) static int snd_virmidi_output_close(struct snd_rawmidi_substream *substream) { struct snd_virmidi *vmidi = substream->runtime->private_data; + + WRITE_ONCE(vmidi->trigger, false); /* to be sure */ + cancel_work_sync(&vmidi->output_work); snd_midi_event_free(vmidi->parser); substream->runtime->private_data = NULL; kfree(vmidi); @@ -546,19 +527,3 @@ int snd_virmidi_new(struct snd_card *card, int device, struct snd_rawmidi **rrmi return 0; } EXPORT_SYMBOL(snd_virmidi_new); - -/* - * ENTRY functions - */ - -static int __init alsa_virmidi_init(void) -{ - return 0; -} - -static void __exit alsa_virmidi_exit(void) -{ -} - -module_init(alsa_virmidi_init) -module_exit(alsa_virmidi_exit) diff --git a/sound/core/timer.c b/sound/core/timer.c index b6f076bbc72d..61a0cec6e1f6 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -883,6 +883,11 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid, if (snd_BUG_ON(!tid)) return -EINVAL; + if (tid->dev_class == SNDRV_TIMER_CLASS_CARD || + tid->dev_class == SNDRV_TIMER_CLASS_PCM) { + if (WARN_ON(!card)) + return -EINVAL; + } if (rtimer) *rtimer = NULL; timer = kzalloc(sizeof(*timer), GFP_KERNEL); diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 78a2fdc38531..1e34e6381baa 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -778,7 +778,6 @@ static const struct snd_pcm_ops loopback_pcm_ops = { .trigger = loopback_trigger, .pointer = loopback_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static int loopback_pcm_new(struct loopback *loopback, diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index 3e745f47dd2f..dae26e856b26 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -617,19 +617,3 @@ free_device: } EXPORT_SYMBOL(snd_mpu401_uart_new); - -/* - * INIT part - */ - -static int __init alsa_mpu401_uart_init(void) -{ - return 0; -} - -static void __exit alsa_mpu401_uart_exit(void) -{ -} - -module_init(alsa_mpu401_uart_init) -module_exit(alsa_mpu401_uart_exit) diff --git a/sound/drivers/opl3/opl3_drums.c b/sound/drivers/opl3/opl3_drums.c index 73694380734a..14929822956c 100644 --- a/sound/drivers/opl3/opl3_drums.c +++ b/sound/drivers/opl3/opl3_drums.c @@ -21,8 +21,6 @@ #include "opl3_voice.h" -extern char snd_opl3_regmap[MAX_OPL2_VOICES][4]; - static char snd_opl3_drum_table[47] = { OPL3_BASSDRUM_ON, OPL3_BASSDRUM_ON, OPL3_HIHAT_ON, /* 35 - 37 */ diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index 588963d6be28..cf86c36c7c3b 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -31,13 +31,12 @@ #include <linux/slab.h> #include <linux/ioport.h> #include <sound/minors.h> +#include "opl3_voice.h" MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Hannu Savolainen 1993-1996, Rob Hooft"); MODULE_DESCRIPTION("Routines for control of AdLib FM cards (OPL2/OPL3/OPL4 chips)"); MODULE_LICENSE("GPL"); -extern char snd_opl3_regmap[MAX_OPL2_VOICES][4]; - static void snd_opl2_command(struct snd_opl3 * opl3, unsigned short cmd, unsigned char val) { unsigned long flags; @@ -539,19 +538,3 @@ int snd_opl3_hwdep_new(struct snd_opl3 * opl3, } EXPORT_SYMBOL(snd_opl3_hwdep_new); - -/* - * INIT part - */ - -static int __init alsa_opl3_init(void) -{ - return 0; -} - -static void __exit alsa_opl3_exit(void) -{ -} - -module_init(alsa_opl3_init) -module_exit(alsa_opl3_exit) diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index bb3f3a5a6951..a33cb744e96c 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -25,10 +25,6 @@ #include "opl3_voice.h" #include <sound/asoundef.h> -extern char snd_opl3_regmap[MAX_OPL2_VOICES][4]; - -extern bool use_internal_drums; - static void snd_opl3_note_off_unsafe(void *p, int note, int vel, struct snd_midi_channel *chan); /* @@ -372,6 +368,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) instr_4op = 1; break; } + /* fall through */ default: spin_unlock_irqrestore(&opl3->voice_lock, flags); return; @@ -721,9 +718,6 @@ void snd_opl3_note_off(void *p, int note, int vel, */ void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *chan) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "Key pressure, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); @@ -735,9 +729,6 @@ void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *cha */ void snd_opl3_terminate_note(void *p, int note, struct snd_midi_channel *chan) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "Terminate note, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); @@ -861,9 +852,6 @@ void snd_opl3_control(void *p, int type, struct snd_midi_channel *chan) void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, struct snd_midi_channel_set *chset) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "NRPN, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); @@ -876,9 +864,6 @@ void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, void snd_opl3_sysex(void *p, unsigned char *buf, int len, int parsed, struct snd_midi_channel_set *chset) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "SYSEX\n"); #endif diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index 22c3e4bca220..869220ced4ed 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -29,8 +29,6 @@ static int snd_opl3_reset_seq_oss(struct snd_seq_oss_arg *arg); /* operators */ -extern struct snd_midi_op opl3_ops; - static struct snd_seq_oss_callback oss_callback = { .owner = THIS_MODULE, .open = snd_opl3_open_seq_oss, @@ -233,11 +231,8 @@ static int snd_opl3_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd, unsigned long ioarg) { - struct snd_opl3 *opl3; - if (snd_BUG_ON(!arg)) return -ENXIO; - opl3 = arg->private_data; switch (cmd) { case SNDCTL_FM_LOAD_INSTR: snd_printk(KERN_ERR "OPL3: " @@ -261,11 +256,8 @@ static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd, /* reset device */ static int snd_opl3_reset_seq_oss(struct snd_seq_oss_arg *arg) { - struct snd_opl3 *opl3; - if (snd_BUG_ON(!arg)) return -ENXIO; - opl3 = arg->private_data; return 0; } diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 42920a243328..d522925fc5c0 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -24,6 +24,7 @@ #include <linux/nospec.h> #include <sound/opl3.h> #include <sound/asound_fm.h> +#include "opl3_voice.h" #if IS_ENABLED(CONFIG_SND_SEQUENCER) #define OPL3_SUPPORT_SYNTH diff --git a/sound/drivers/opl3/opl3_voice.h b/sound/drivers/opl3/opl3_voice.h index a2445163008e..5b02bd49fde4 100644 --- a/sound/drivers/opl3/opl3_voice.h +++ b/sound/drivers/opl3/opl3_voice.h @@ -52,4 +52,8 @@ void snd_opl3_free_seq_oss(struct snd_opl3 *opl3); #define snd_opl3_free_seq_oss(opl3) /* NOP */ #endif +extern char snd_opl3_regmap[MAX_OPL2_VOICES][4]; +extern bool use_internal_drums; +extern struct snd_midi_op opl3_ops; + #endif diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c index db76a5bf2bd2..819d2dce2a19 100644 --- a/sound/drivers/opl4/opl4_lib.c +++ b/sound/drivers/opl4/opl4_lib.c @@ -263,15 +263,3 @@ int snd_opl4_create(struct snd_card *card, } EXPORT_SYMBOL(snd_opl4_create); - -static int __init alsa_opl4_init(void) -{ - return 0; -} - -static void __exit alsa_opl4_exit(void) -{ -} - -module_init(alsa_opl4_init) -module_exit(alsa_opl4_exit) diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index 121357397a6d..04368dd59a4c 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -815,18 +815,3 @@ struct vx_core *snd_vx_create(struct snd_card *card, struct snd_vx_hardware *hw, } EXPORT_SYMBOL(snd_vx_create); - -/* - * module entries - */ -static int __init alsa_vx_core_init(void) -{ - return 0; -} - -static void __exit alsa_vx_core_exit(void) -{ -} - -module_init(alsa_vx_core_init) -module_exit(alsa_vx_core_exit) diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 380a028469c4..ba80f459bdc5 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -883,7 +883,6 @@ static const struct snd_pcm_ops vx_pcm_playback_ops = { .trigger = vx_pcm_trigger, .pointer = vx_pcm_playback_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; @@ -1105,7 +1104,6 @@ static const struct snd_pcm_ops vx_pcm_capture_ops = { .trigger = vx_pcm_trigger, .pointer = vx_pcm_capture_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index e6adab3ef42e..ea9b86450580 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -373,7 +373,6 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c index b2efb1c71a98..218292bdace6 100644 --- a/sound/firewire/dice/dice-alesis.c +++ b/sound/firewire/dice/dice-alesis.c @@ -37,7 +37,7 @@ int snd_dice_detect_alesis_formats(struct snd_dice *dice) MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); } else { - memcpy(dice->rx_pcm_chs, alesis_io26_tx_pcm_chs, + memcpy(dice->tx_pcm_chs, alesis_io26_tx_pcm_chs, MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); } diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 80351b29fe0d..bb3ef5ff3488 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -412,7 +412,6 @@ int snd_dice_create_pcm(struct snd_dice *dice) .pointer = capture_pointer, .ack = capture_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -425,7 +424,6 @@ int snd_dice_create_pcm(struct snd_dice *dice) .pointer = playback_pointer, .ack = playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; unsigned int capture, playback; diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index 796f4b4645f5..fdcff0460c53 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -352,7 +352,6 @@ int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index e3c16308363d..bf47f9ec8703 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -383,7 +383,6 @@ int snd_ff_create_pcm_devices(struct snd_ff *ff) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index 40faed5e6968..aed566d82726 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -397,7 +397,6 @@ int snd_efw_create_pcm_devices(struct snd_efw *efw) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 3919e186a30b..30957477e005 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -454,7 +454,6 @@ static int isight_create_pcm(struct isight *isight) .trigger = isight_trigger, .pointer = isight_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index 4330220890e8..ab69d7e6ac05 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -363,7 +363,6 @@ int snd_motu_create_pcm_devices(struct snd_motu *motu) .pointer = capture_pointer, .ack = capture_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -376,7 +375,6 @@ int snd_motu_create_pcm_devices(struct snd_motu *motu) .pointer = playback_pointer, .ack = playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c index 525b746330be..453fc29fade7 100644 --- a/sound/firewire/motu/motu-protocol-v2.c +++ b/sound/firewire/motu/motu-protocol-v2.c @@ -13,6 +13,8 @@ #define V2_CLOCK_RATE_SHIFT 3 #define V2_CLOCK_SRC_MASK 0x00000007 #define V2_CLOCK_SRC_SHIFT 0 +#define V2_CLOCK_TRAVELER_FETCH_DISABLE 0x04000000 +#define V2_CLOCK_TRAVELER_FETCH_ENABLE 0x03000000 #define V2_IN_OUT_CONF_OFFSET 0x0c04 #define V2_OPT_OUT_IFACE_MASK 0x00000c00 @@ -66,6 +68,11 @@ static int v2_set_clock_rate(struct snd_motu *motu, unsigned int rate) data &= ~V2_CLOCK_RATE_MASK; data |= i << V2_CLOCK_RATE_SHIFT; + if (motu->spec == &snd_motu_spec_traveler) { + data &= ~V2_CLOCK_TRAVELER_FETCH_ENABLE; + data |= V2_CLOCK_TRAVELER_FETCH_DISABLE; + } + reg = cpu_to_be32(data); return snd_motu_transaction_write(motu, V2_CLOCK_STATUS_OFFSET, ®, sizeof(reg)); @@ -121,8 +128,31 @@ static int v2_get_clock_source(struct snd_motu *motu, static int v2_switch_fetching_mode(struct snd_motu *motu, bool enable) { - /* V2 protocol doesn't have this feature. */ - return 0; + __be32 reg; + u32 data; + int err = 0; + + if (motu->spec == &snd_motu_spec_traveler) { + err = snd_motu_transaction_read(motu, V2_CLOCK_STATUS_OFFSET, + ®, sizeof(reg)); + if (err < 0) + return err; + data = be32_to_cpu(reg); + + data &= ~(V2_CLOCK_TRAVELER_FETCH_DISABLE | + V2_CLOCK_TRAVELER_FETCH_ENABLE); + + if (enable) + data |= V2_CLOCK_TRAVELER_FETCH_ENABLE; + else + data |= V2_CLOCK_TRAVELER_FETCH_DISABLE; + + reg = cpu_to_be32(data); + err = snd_motu_transaction_write(motu, V2_CLOCK_STATUS_OFFSET, + ®, sizeof(reg)); + } + + return err; } static void calculate_fixed_part(struct snd_motu_packet_format *formats, @@ -149,11 +179,20 @@ static void calculate_fixed_part(struct snd_motu_packet_format *formats, pcm_chunks[1] += 2; } } else { - /* - * Packets to v2 units transfer main-out-1/2 and phone-out-1/2. - */ - pcm_chunks[0] += 4; - pcm_chunks[1] += 4; + if (flags & SND_MOTU_SPEC_RX_SEPARETED_MAIN) { + pcm_chunks[0] += 2; + pcm_chunks[1] += 2; + } + + // Packets to v2 units include 2 chunks for phone 1/2, except + // for 176.4/192.0 kHz. + pcm_chunks[0] += 2; + pcm_chunks[1] += 2; + } + + if (flags & SND_MOTU_SPEC_HAS_AESEBU_IFACE) { + pcm_chunks[0] += 2; + pcm_chunks[1] += 2; } /* @@ -164,19 +203,16 @@ static void calculate_fixed_part(struct snd_motu_packet_format *formats, pcm_chunks[0] += 2; pcm_chunks[1] += 2; - /* This part should be multiples of 4. */ - formats->fixed_part_pcm_chunks[0] = round_up(2 + pcm_chunks[0], 4) - 2; - formats->fixed_part_pcm_chunks[1] = round_up(2 + pcm_chunks[1], 4) - 2; - if (flags & SND_MOTU_SPEC_SUPPORT_CLOCK_X4) - formats->fixed_part_pcm_chunks[2] = - round_up(2 + pcm_chunks[2], 4) - 2; + formats->fixed_part_pcm_chunks[0] = pcm_chunks[0]; + formats->fixed_part_pcm_chunks[1] = pcm_chunks[1]; + formats->fixed_part_pcm_chunks[2] = pcm_chunks[2]; } static void calculate_differed_part(struct snd_motu_packet_format *formats, enum snd_motu_spec_flags flags, u32 data, u32 mask, u32 shift) { - unsigned char pcm_chunks[3] = {0, 0}; + unsigned char pcm_chunks[2] = {0, 0}; /* * When optical interfaces are configured for S/PDIF (TOSLINK), diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index c7cd9864dc4d..7cc80a05e91f 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -188,11 +188,20 @@ static void calculate_fixed_part(struct snd_motu_packet_format *formats, pcm_chunks[1] += 2; } } else { - /* - * Packets to v2 units transfer main-out-1/2 and phone-out-1/2. - */ - pcm_chunks[0] += 4; - pcm_chunks[1] += 4; + if (flags & SND_MOTU_SPEC_RX_SEPARETED_MAIN) { + pcm_chunks[0] += 2; + pcm_chunks[1] += 2; + } + + // Packets to v3 units include 2 chunks for phone 1/2, except + // for 176.4/192.0 kHz. + pcm_chunks[0] += 2; + pcm_chunks[1] += 2; + } + + if (flags & SND_MOTU_SPEC_HAS_AESEBU_IFACE) { + pcm_chunks[0] += 2; + pcm_chunks[1] += 2; } /* diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 0d6b526105ab..300d31b6f191 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -200,6 +200,22 @@ static const struct snd_motu_spec motu_828mk2 = { .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | SND_MOTU_SPEC_TX_MICINST_CHUNK | SND_MOTU_SPEC_TX_RETURN_CHUNK | + SND_MOTU_SPEC_RX_SEPARETED_MAIN | + SND_MOTU_SPEC_HAS_OPT_IFACE_A | + SND_MOTU_SPEC_RX_MIDI_2ND_Q | + SND_MOTU_SPEC_TX_MIDI_2ND_Q, + + .analog_in_ports = 8, + .analog_out_ports = 8, +}; + +const struct snd_motu_spec snd_motu_spec_traveler = { + .name = "Traveler", + .protocol = &snd_motu_protocol_v2, + .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | + SND_MOTU_SPEC_SUPPORT_CLOCK_X4 | + SND_MOTU_SPEC_TX_RETURN_CHUNK | + SND_MOTU_SPEC_HAS_AESEBU_IFACE | SND_MOTU_SPEC_HAS_OPT_IFACE_A | SND_MOTU_SPEC_RX_MIDI_2ND_Q | SND_MOTU_SPEC_TX_MIDI_2ND_Q, @@ -216,6 +232,7 @@ static const struct snd_motu_spec motu_828mk3 = { SND_MOTU_SPEC_TX_MICINST_CHUNK | SND_MOTU_SPEC_TX_RETURN_CHUNK | SND_MOTU_SPEC_TX_REVERB_CHUNK | + SND_MOTU_SPEC_RX_SEPARETED_MAIN | SND_MOTU_SPEC_HAS_OPT_IFACE_A | SND_MOTU_SPEC_HAS_OPT_IFACE_B | SND_MOTU_SPEC_RX_MIDI_3RD_Q | @@ -231,6 +248,7 @@ static const struct snd_motu_spec motu_audio_express = { .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | SND_MOTU_SPEC_TX_MICINST_CHUNK | SND_MOTU_SPEC_TX_RETURN_CHUNK | + SND_MOTU_SPEC_RX_SEPARETED_MAIN | SND_MOTU_SPEC_RX_MIDI_2ND_Q | SND_MOTU_SPEC_TX_MIDI_3RD_Q, .analog_in_ports = 2, @@ -250,6 +268,7 @@ static const struct snd_motu_spec motu_audio_express = { static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x101800, &motu_828mk2), + SND_MOTU_DEV_ENTRY(0x107800, &snd_motu_spec_traveler), SND_MOTU_DEV_ENTRY(0x106800, &motu_828mk3), /* FireWire only. */ SND_MOTU_DEV_ENTRY(0x100800, &motu_828mk3), /* Hybrid. */ SND_MOTU_DEV_ENTRY(0x104800, &motu_audio_express), diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 4b23cf337c4b..fd5327d30ab1 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -79,13 +79,14 @@ enum snd_motu_spec_flags { SND_MOTU_SPEC_TX_MICINST_CHUNK = 0x0004, SND_MOTU_SPEC_TX_RETURN_CHUNK = 0x0008, SND_MOTU_SPEC_TX_REVERB_CHUNK = 0x0010, - SND_MOTU_SPEC_TX_AESEBU_CHUNK = 0x0020, + SND_MOTU_SPEC_HAS_AESEBU_IFACE = 0x0020, SND_MOTU_SPEC_HAS_OPT_IFACE_A = 0x0040, SND_MOTU_SPEC_HAS_OPT_IFACE_B = 0x0080, SND_MOTU_SPEC_RX_MIDI_2ND_Q = 0x0100, SND_MOTU_SPEC_RX_MIDI_3RD_Q = 0x0200, SND_MOTU_SPEC_TX_MIDI_2ND_Q = 0x0400, SND_MOTU_SPEC_TX_MIDI_3RD_Q = 0x0800, + SND_MOTU_SPEC_RX_SEPARETED_MAIN = 0x1000, }; #define SND_MOTU_CLOCK_RATE_COUNT 6 @@ -128,6 +129,8 @@ struct snd_motu_spec { extern const struct snd_motu_protocol snd_motu_protocol_v2; extern const struct snd_motu_protocol snd_motu_protocol_v3; +extern const struct snd_motu_spec snd_motu_spec_traveler; + int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir, const struct snd_motu_protocol *const protocol); diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 3dd46285c0e2..b3f6503dd34d 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -389,7 +389,6 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw) .pointer = pcm_capture_pointer, .ack = pcm_capture_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -402,7 +401,6 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; unsigned int cap = 0; diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index 6ec8ec634d4d..e4cc8990e195 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -279,7 +279,6 @@ int snd_tscm_create_pcm_devices(struct snd_tscm *tscm) .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; int err; diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3129546398d0..2d90e11b3eaa 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -5,11 +5,12 @@ config SND_HDA_CORE config SND_HDA_DSP_LOADER bool +config SND_HDA_COMPONENT + bool + config SND_HDA_I915 bool - default y - depends on DRM_I915 - depends on SND_HDA_CORE + select SND_HDA_COMPONENT config SND_HDA_EXT_CORE tristate diff --git a/sound/hda/Makefile b/sound/hda/Makefile index e4e726f2ce98..2160202e2dc1 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -6,6 +6,7 @@ snd-hda-core-objs += trace.o CFLAGS_trace.o := -I$(src) # for sync with i915 gfx driver +snd-hda-core-$(CONFIG_SND_HDA_COMPONENT) += hdac_component.o snd-hda-core-$(CONFIG_SND_HDA_I915) += hdac_i915.o obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 0daf31383084..9c37d9af3023 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -87,9 +87,10 @@ static const struct hdac_io_ops hdac_ext_default_io = { * * Returns 0 if successful, or a negative error code. */ -int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev, +int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, const struct hdac_bus_ops *ops, - const struct hdac_io_ops *io_ops) + const struct hdac_io_ops *io_ops, + const struct hdac_ext_bus_ops *ext_ops) { int ret; static int idx; @@ -98,15 +99,16 @@ int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev, if (io_ops == NULL) io_ops = &hdac_ext_default_io; - ret = snd_hdac_bus_init(&ebus->bus, dev, ops, io_ops); + ret = snd_hdac_bus_init(bus, dev, ops, io_ops); if (ret < 0) return ret; - INIT_LIST_HEAD(&ebus->hlink_list); - ebus->idx = idx++; + bus->ext_ops = ext_ops; + INIT_LIST_HEAD(&bus->hlink_list); + bus->idx = idx++; - mutex_init(&ebus->lock); - ebus->cmd_dma_state = true; + mutex_init(&bus->lock); + bus->cmd_dma_state = true; return 0; } @@ -116,10 +118,10 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init); * snd_hdac_ext_bus_exit - clean up a HD-audio extended bus * @ebus: the pointer to extended bus object */ -void snd_hdac_ext_bus_exit(struct hdac_ext_bus *ebus) +void snd_hdac_ext_bus_exit(struct hdac_bus *bus) { - snd_hdac_bus_exit(&ebus->bus); - WARN_ON(!list_empty(&ebus->hlink_list)); + snd_hdac_bus_exit(bus); + WARN_ON(!list_empty(&bus->hlink_list)); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_exit); @@ -135,21 +137,15 @@ static void default_release(struct device *dev) * * Returns zero for success or a negative error code. */ -int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) +int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, + struct hdac_device *hdev) { - struct hdac_ext_device *edev; - struct hdac_device *hdev = NULL; - struct hdac_bus *bus = ebus_to_hbus(ebus); char name[15]; int ret; - edev = kzalloc(sizeof(*edev), GFP_KERNEL); - if (!edev) - return -ENOMEM; - hdev = &edev->hdev; - edev->ebus = ebus; + hdev->bus = bus; - snprintf(name, sizeof(name), "ehdaudio%dD%d", ebus->idx, addr); + snprintf(name, sizeof(name), "ehdaudio%dD%d", bus->idx, addr); ret = snd_hdac_device_init(hdev, bus, name, addr); if (ret < 0) { @@ -176,10 +172,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_init); */ void snd_hdac_ext_bus_device_exit(struct hdac_device *hdev) { - struct hdac_ext_device *edev = to_ehdac_device(hdev); - snd_hdac_device_exit(hdev); - kfree(edev); + kfree(hdev); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_exit); @@ -188,14 +182,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_exit); * * @ebus: HD-audio extended bus */ -void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus) +void snd_hdac_ext_bus_device_remove(struct hdac_bus *bus) { struct hdac_device *codec, *__codec; /* * we need to remove all the codec devices objects created in the * snd_hdac_ext_bus_device_init */ - list_for_each_entry_safe(codec, __codec, &ebus->bus.codec_list, list) { + list_for_each_entry_safe(codec, __codec, &bus->codec_list, list) { snd_hdac_device_unregister(codec); put_device(&codec->dev); } @@ -204,35 +198,31 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_remove); #define dev_to_hdac(dev) (container_of((dev), \ struct hdac_device, dev)) -static inline struct hdac_ext_driver *get_edrv(struct device *dev) +static inline struct hdac_driver *get_hdrv(struct device *dev) { struct hdac_driver *hdrv = drv_to_hdac_driver(dev->driver); - struct hdac_ext_driver *edrv = to_ehdac_driver(hdrv); - - return edrv; + return hdrv; } -static inline struct hdac_ext_device *get_edev(struct device *dev) +static inline struct hdac_device *get_hdev(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); - struct hdac_ext_device *edev = to_ehdac_device(hdev); - - return edev; + return hdev; } static int hda_ext_drv_probe(struct device *dev) { - return (get_edrv(dev))->probe(get_edev(dev)); + return (get_hdrv(dev))->probe(get_hdev(dev)); } static int hdac_ext_drv_remove(struct device *dev) { - return (get_edrv(dev))->remove(get_edev(dev)); + return (get_hdrv(dev))->remove(get_hdev(dev)); } static void hdac_ext_drv_shutdown(struct device *dev) { - return (get_edrv(dev))->shutdown(get_edev(dev)); + return (get_hdrv(dev))->shutdown(get_hdev(dev)); } /** @@ -240,20 +230,20 @@ static void hdac_ext_drv_shutdown(struct device *dev) * * @drv: ext hda driver structure */ -int snd_hda_ext_driver_register(struct hdac_ext_driver *drv) +int snd_hda_ext_driver_register(struct hdac_driver *drv) { - drv->hdac.type = HDA_DEV_ASOC; - drv->hdac.driver.bus = &snd_hda_bus_type; + drv->type = HDA_DEV_ASOC; + drv->driver.bus = &snd_hda_bus_type; /* we use default match */ if (drv->probe) - drv->hdac.driver.probe = hda_ext_drv_probe; + drv->driver.probe = hda_ext_drv_probe; if (drv->remove) - drv->hdac.driver.remove = hdac_ext_drv_remove; + drv->driver.remove = hdac_ext_drv_remove; if (drv->shutdown) - drv->hdac.driver.shutdown = hdac_ext_drv_shutdown; + drv->driver.shutdown = hdac_ext_drv_shutdown; - return driver_register(&drv->hdac.driver); + return driver_register(&drv->driver); } EXPORT_SYMBOL_GPL(snd_hda_ext_driver_register); @@ -262,8 +252,8 @@ EXPORT_SYMBOL_GPL(snd_hda_ext_driver_register); * * @drv: ext hda driver structure */ -void snd_hda_ext_driver_unregister(struct hdac_ext_driver *drv) +void snd_hda_ext_driver_unregister(struct hdac_driver *drv) { - driver_unregister(&drv->hdac.driver); + driver_unregister(&drv->driver); } EXPORT_SYMBOL_GPL(snd_hda_ext_driver_unregister); diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 84f3b8168716..5bc4a1d587d4 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -39,9 +39,8 @@ * @ebus: HD-audio extended core bus * @enable: flag to turn on/off the capability */ -void snd_hdac_ext_bus_ppcap_enable(struct hdac_ext_bus *ebus, bool enable) +void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *bus, bool enable) { - struct hdac_bus *bus = &ebus->bus; if (!bus->ppcap) { dev_err(bus->dev, "Address of PP capability is NULL"); @@ -60,9 +59,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_enable); * @ebus: HD-audio extended core bus * @enable: flag to enable/disable interrupt */ -void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_ext_bus *ebus, bool enable) +void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *bus, bool enable) { - struct hdac_bus *bus = &ebus->bus; if (!bus->ppcap) { dev_err(bus->dev, "Address of PP capability is NULL\n"); @@ -89,12 +87,11 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_int_enable); * in hlink_list of extended hdac bus * Note: this will be freed on bus exit by driver */ -int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus) +int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_bus *bus) { int idx; u32 link_count; struct hdac_ext_link *hlink; - struct hdac_bus *bus = &ebus->bus; link_count = readl(bus->mlcap + AZX_REG_ML_MLCD) + 1; @@ -114,7 +111,7 @@ int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus) /* since link in On, update the ref */ hlink->ref_count = 1; - list_add_tail(&hlink->list, &ebus->hlink_list); + list_add_tail(&hlink->list, &bus->hlink_list); } return 0; @@ -127,12 +124,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_get_ml_capabilities); * @ebus: HD-audio ext core bus */ -void snd_hdac_link_free_all(struct hdac_ext_bus *ebus) +void snd_hdac_link_free_all(struct hdac_bus *bus) { struct hdac_ext_link *l; - while (!list_empty(&ebus->hlink_list)) { - l = list_first_entry(&ebus->hlink_list, struct hdac_ext_link, list); + while (!list_empty(&bus->hlink_list)) { + l = list_first_entry(&bus->hlink_list, struct hdac_ext_link, list); list_del(&l->list); kfree(l); } @@ -144,7 +141,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_link_free_all); * @ebus: HD-audio extended core bus * @codec_name: codec name */ -struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_ext_bus *ebus, +struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_bus *bus, const char *codec_name) { int i; @@ -153,10 +150,10 @@ struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_ext_bus *ebus, if (sscanf(codec_name, "ehdaudio%dD%d", &bus_idx, &addr) != 2) return NULL; - if (ebus->idx != bus_idx) + if (bus->idx != bus_idx) return NULL; - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { for (i = 0; i < HDA_MAX_CODECS; i++) { if (hlink->lsdiid & (0x1 << addr)) return hlink; @@ -219,12 +216,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down); * snd_hdac_ext_bus_link_power_up_all -power up all hda link * @ebus: HD-audio extended bus */ -int snd_hdac_ext_bus_link_power_up_all(struct hdac_ext_bus *ebus) +int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus) { struct hdac_ext_link *hlink = NULL; int ret; - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA); ret = check_hdac_link_power_active(hlink, true); @@ -240,12 +237,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_up_all); * snd_hdac_ext_bus_link_power_down_all -power down all hda link * @ebus: HD-audio extended bus */ -int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus) +int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus) { struct hdac_ext_link *hlink = NULL; int ret; - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, AZX_MLCTL_SPA, 0); ret = check_hdac_link_power_active(hlink, false); if (ret < 0) @@ -256,39 +253,48 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus) } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all); -int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus, +int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, struct hdac_ext_link *link) { int ret = 0; - mutex_lock(&ebus->lock); + mutex_lock(&bus->lock); /* * if we move from 0 to 1, count will be 1 so power up this link * as well, also check the dma status and trigger that */ if (++link->ref_count == 1) { - if (!ebus->cmd_dma_state) { - snd_hdac_bus_init_cmd_io(&ebus->bus); - ebus->cmd_dma_state = true; + if (!bus->cmd_dma_state) { + snd_hdac_bus_init_cmd_io(bus); + bus->cmd_dma_state = true; } ret = snd_hdac_ext_bus_link_power_up(link); + + /* + * wait for 521usec for codec to report status + * HDA spec section 4.3 - Codec Discovery + */ + udelay(521); + bus->codec_mask = snd_hdac_chip_readw(bus, STATESTS); + dev_dbg(bus->dev, "codec_mask = 0x%lx\n", bus->codec_mask); + snd_hdac_chip_writew(bus, STATESTS, bus->codec_mask); } - mutex_unlock(&ebus->lock); + mutex_unlock(&bus->lock); return ret; } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_get); -int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, +int snd_hdac_ext_bus_link_put(struct hdac_bus *bus, struct hdac_ext_link *link) { int ret = 0; struct hdac_ext_link *hlink; bool link_up = false; - mutex_lock(&ebus->lock); + mutex_lock(&bus->lock); /* * if we move from 1 to 0, count will be 0 @@ -301,7 +307,7 @@ int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, * now check if all links are off, if so turn off * cmd dma as well */ - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { if (hlink->ref_count) { link_up = true; break; @@ -309,12 +315,12 @@ int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, } if (!link_up) { - snd_hdac_bus_stop_cmd_io(&ebus->bus); - ebus->cmd_dma_state = false; + snd_hdac_bus_stop_cmd_io(bus); + bus->cmd_dma_state = false; } } - mutex_unlock(&ebus->lock); + mutex_unlock(&bus->lock); return ret; } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_put); diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index c96d7a7a36af..1bd27576db98 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -25,7 +25,7 @@ /** * snd_hdac_ext_stream_init - initialize each stream (aka device) - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: HD-audio ext core stream object to initialize * @idx: stream index number * @direction: stream direction (SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE) @@ -34,18 +34,16 @@ * initialize the stream, if ppcap is enabled then init those and then * invoke hdac stream initialization routine */ -void snd_hdac_ext_stream_init(struct hdac_ext_bus *ebus, +void snd_hdac_ext_stream_init(struct hdac_bus *bus, struct hdac_ext_stream *stream, int idx, int direction, int tag) { - struct hdac_bus *bus = &ebus->bus; - if (bus->ppcap) { stream->pphc_addr = bus->ppcap + AZX_PPHC_BASE + AZX_PPHC_INTERVAL * idx; stream->pplc_addr = bus->ppcap + AZX_PPLC_BASE + - AZX_PPLC_MULTI * ebus->num_streams + + AZX_PPLC_MULTI * bus->num_streams + AZX_PPLC_INTERVAL * idx; } @@ -71,12 +69,12 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init); /** * snd_hdac_ext_stream_init_all - create and initialize the stream objects * for an extended hda bus - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @start_idx: start index for streams * @num_stream: number of streams to initialize * @dir: direction of streams */ -int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx, +int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx, int num_stream, int dir) { int stream_tag = 0; @@ -88,7 +86,7 @@ int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx, if (!stream) return -ENOMEM; tag = ++stream_tag; - snd_hdac_ext_stream_init(ebus, stream, idx, dir, tag); + snd_hdac_ext_stream_init(bus, stream, idx, dir, tag); idx++; } @@ -100,17 +98,16 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init_all); /** * snd_hdac_stream_free_all - free hdac extended stream objects * - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus */ -void snd_hdac_stream_free_all(struct hdac_ext_bus *ebus) +void snd_hdac_stream_free_all(struct hdac_bus *bus) { struct hdac_stream *s, *_s; struct hdac_ext_stream *stream; - struct hdac_bus *bus = ebus_to_hbus(ebus); list_for_each_entry_safe(s, _s, &bus->stream_list, list) { stream = stream_to_hdac_ext_stream(s); - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); list_del(&s->list); kfree(stream); } @@ -119,15 +116,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_stream_free_all); /** * snd_hdac_ext_stream_decouple - decouple the hdac stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: HD-audio ext core stream object to initialize * @decouple: flag to decouple */ -void snd_hdac_ext_stream_decouple(struct hdac_ext_bus *ebus, +void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, struct hdac_ext_stream *stream, bool decouple) { struct hdac_stream *hstream = &stream->hstream; - struct hdac_bus *bus = &ebus->bus; u32 val; int mask = AZX_PPCTL_PROCEN(hstream->index); @@ -251,19 +247,18 @@ void snd_hdac_ext_link_clear_stream_id(struct hdac_ext_link *link, EXPORT_SYMBOL_GPL(snd_hdac_ext_link_clear_stream_id); static struct hdac_ext_stream * -hdac_ext_link_stream_assign(struct hdac_ext_bus *ebus, +hdac_ext_link_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream) { struct hdac_ext_stream *res = NULL; struct hdac_stream *stream = NULL; - struct hdac_bus *hbus = &ebus->bus; - if (!hbus->ppcap) { - dev_err(hbus->dev, "stream type not supported\n"); + if (!bus->ppcap) { + dev_err(bus->dev, "stream type not supported\n"); return NULL; } - list_for_each_entry(stream, &hbus->stream_list, list) { + list_for_each_entry(stream, &bus->stream_list, list) { struct hdac_ext_stream *hstream = container_of(stream, struct hdac_ext_stream, hstream); @@ -277,34 +272,33 @@ hdac_ext_link_stream_assign(struct hdac_ext_bus *ebus, } if (!hstream->link_locked) { - snd_hdac_ext_stream_decouple(ebus, hstream, true); + snd_hdac_ext_stream_decouple(bus, hstream, true); res = hstream; break; } } if (res) { - spin_lock_irq(&hbus->reg_lock); + spin_lock_irq(&bus->reg_lock); res->link_locked = 1; res->link_substream = substream; - spin_unlock_irq(&hbus->reg_lock); + spin_unlock_irq(&bus->reg_lock); } return res; } static struct hdac_ext_stream * -hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus, +hdac_ext_host_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream) { struct hdac_ext_stream *res = NULL; struct hdac_stream *stream = NULL; - struct hdac_bus *hbus = &ebus->bus; - if (!hbus->ppcap) { - dev_err(hbus->dev, "stream type not supported\n"); + if (!bus->ppcap) { + dev_err(bus->dev, "stream type not supported\n"); return NULL; } - list_for_each_entry(stream, &hbus->stream_list, list) { + list_for_each_entry(stream, &bus->stream_list, list) { struct hdac_ext_stream *hstream = container_of(stream, struct hdac_ext_stream, hstream); @@ -313,17 +307,17 @@ hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus, if (!stream->opened) { if (!hstream->decoupled) - snd_hdac_ext_stream_decouple(ebus, hstream, true); + snd_hdac_ext_stream_decouple(bus, hstream, true); res = hstream; break; } } if (res) { - spin_lock_irq(&hbus->reg_lock); + spin_lock_irq(&bus->reg_lock); res->hstream.opened = 1; res->hstream.running = 0; res->hstream.substream = substream; - spin_unlock_irq(&hbus->reg_lock); + spin_unlock_irq(&bus->reg_lock); } return res; @@ -331,7 +325,7 @@ hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus, /** * snd_hdac_ext_stream_assign - assign a stream for the PCM - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @substream: PCM substream to assign * @type: type of stream (coupled, host or link stream) * @@ -346,27 +340,26 @@ hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus, * the same stream object when it's used beforehand. when a stream is * decoupled, it becomes a host stream and link stream. */ -struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_ext_bus *ebus, +struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream, int type) { struct hdac_ext_stream *hstream = NULL; struct hdac_stream *stream = NULL; - struct hdac_bus *hbus = &ebus->bus; switch (type) { case HDAC_EXT_STREAM_TYPE_COUPLED: - stream = snd_hdac_stream_assign(hbus, substream); + stream = snd_hdac_stream_assign(bus, substream); if (stream) hstream = container_of(stream, struct hdac_ext_stream, hstream); return hstream; case HDAC_EXT_STREAM_TYPE_HOST: - return hdac_ext_host_stream_assign(ebus, substream); + return hdac_ext_host_stream_assign(bus, substream); case HDAC_EXT_STREAM_TYPE_LINK: - return hdac_ext_link_stream_assign(ebus, substream); + return hdac_ext_link_stream_assign(bus, substream); default: return NULL; @@ -384,7 +377,6 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_assign); void snd_hdac_ext_stream_release(struct hdac_ext_stream *stream, int type) { struct hdac_bus *bus = stream->hstream.bus; - struct hdac_ext_bus *ebus = hbus_to_ebus(bus); switch (type) { case HDAC_EXT_STREAM_TYPE_COUPLED: @@ -393,13 +385,13 @@ void snd_hdac_ext_stream_release(struct hdac_ext_stream *stream, int type) case HDAC_EXT_STREAM_TYPE_HOST: if (stream->decoupled && !stream->link_locked) - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); snd_hdac_stream_release(&stream->hstream); break; case HDAC_EXT_STREAM_TYPE_LINK: if (stream->decoupled && !stream->hstream.opened) - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); spin_lock_irq(&bus->reg_lock); stream->link_locked = 0; stream->link_substream = NULL; @@ -415,16 +407,15 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_release); /** * snd_hdac_ext_stream_spbcap_enable - enable SPIB for a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @enable: flag to enable/disable SPIB * @index: stream index for which SPIB need to be enabled */ -void snd_hdac_ext_stream_spbcap_enable(struct hdac_ext_bus *ebus, +void snd_hdac_ext_stream_spbcap_enable(struct hdac_bus *bus, bool enable, int index) { u32 mask = 0; u32 register_mask = 0; - struct hdac_bus *bus = &ebus->bus; if (!bus->spbcap) { dev_err(bus->dev, "Address of SPB capability is NULL\n"); @@ -446,14 +437,13 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_spbcap_enable); /** * snd_hdac_ext_stream_set_spib - sets the spib value of a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: hdac_ext_stream * @value: spib value to set */ -int snd_hdac_ext_stream_set_spib(struct hdac_ext_bus *ebus, +int snd_hdac_ext_stream_set_spib(struct hdac_bus *bus, struct hdac_ext_stream *stream, u32 value) { - struct hdac_bus *bus = &ebus->bus; if (!bus->spbcap) { dev_err(bus->dev, "Address of SPB capability is NULL\n"); @@ -468,15 +458,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_set_spib); /** * snd_hdac_ext_stream_get_spbmaxfifo - gets the spib value of a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: hdac_ext_stream * * Return maxfifo for the stream */ -int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_ext_bus *ebus, +int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_bus *bus, struct hdac_ext_stream *stream) { - struct hdac_bus *bus = &ebus->bus; if (!bus->spbcap) { dev_err(bus->dev, "Address of SPB capability is NULL\n"); @@ -490,11 +479,10 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_get_spbmaxfifo); /** * snd_hdac_ext_stop_streams - stop all stream if running - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus */ -void snd_hdac_ext_stop_streams(struct hdac_ext_bus *ebus) +void snd_hdac_ext_stop_streams(struct hdac_bus *bus) { - struct hdac_bus *bus = ebus_to_hbus(ebus); struct hdac_stream *stream; if (bus->chip_init) { @@ -507,16 +495,15 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stop_streams); /** * snd_hdac_ext_stream_drsm_enable - enable DMA resume for a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @enable: flag to enable/disable DRSM * @index: stream index for which DRSM need to be enabled */ -void snd_hdac_ext_stream_drsm_enable(struct hdac_ext_bus *ebus, +void snd_hdac_ext_stream_drsm_enable(struct hdac_bus *bus, bool enable, int index) { u32 mask = 0; u32 register_mask = 0; - struct hdac_bus *bus = &ebus->bus; if (!bus->drsmcap) { dev_err(bus->dev, "Address of DRSM capability is NULL\n"); @@ -538,14 +525,13 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_drsm_enable); /** * snd_hdac_ext_stream_set_dpibr - sets the dpibr value of a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: hdac_ext_stream * @value: dpib value to set */ -int snd_hdac_ext_stream_set_dpibr(struct hdac_ext_bus *ebus, +int snd_hdac_ext_stream_set_dpibr(struct hdac_bus *bus, struct hdac_ext_stream *stream, u32 value) { - struct hdac_bus *bus = &ebus->bus; if (!bus->drsmcap) { dev_err(bus->dev, "Address of DRSM capability is NULL\n"); @@ -560,7 +546,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_set_dpibr); /** * snd_hdac_ext_stream_set_lpib - sets the lpib value of a stream - * @ebus: HD-audio ext core bus + * @bus: HD-audio core bus * @stream: hdac_ext_stream * @value: lpib value to set */ diff --git a/sound/hda/hdac_component.c b/sound/hda/hdac_component.c new file mode 100644 index 000000000000..6e46a9c73aed --- /dev/null +++ b/sound/hda/hdac_component.c @@ -0,0 +1,335 @@ +// SPDX-License-Identifier: GPL-2.0 +// hdac_component.c - routines for sync between HD-A core and DRM driver + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <linux/component.h> +#include <sound/core.h> +#include <sound/hdaudio.h> +#include <sound/hda_component.h> +#include <sound/hda_register.h> + +static void hdac_acomp_release(struct device *dev, void *res) +{ +} + +static struct drm_audio_component *hdac_get_acomp(struct device *dev) +{ + return devres_find(dev, hdac_acomp_release, NULL, NULL); +} + +/** + * snd_hdac_set_codec_wakeup - Enable / disable HDMI/DP codec wakeup + * @bus: HDA core bus + * @enable: enable or disable the wakeup + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function should be called during the chip reset, also called at + * resume for updating STATESTS register read. + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) +{ + struct drm_audio_component *acomp = bus->audio_component; + + if (!acomp || !acomp->ops) + return -ENODEV; + + if (!acomp->ops->codec_wake_override) + return 0; + + dev_dbg(bus->dev, "%s codec wakeup\n", + enable ? "enable" : "disable"); + + acomp->ops->codec_wake_override(acomp->dev, enable); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup); + +/** + * snd_hdac_display_power - Power up / down the power refcount + * @bus: HDA core bus + * @enable: power up or down + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function manages a refcount and calls the get_power() and + * put_power() ops accordingly, toggling the codec wakeup, too. + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_display_power(struct hdac_bus *bus, bool enable) +{ + struct drm_audio_component *acomp = bus->audio_component; + + if (!acomp || !acomp->ops) + return -ENODEV; + + dev_dbg(bus->dev, "display power %s\n", + enable ? "enable" : "disable"); + + if (enable) { + if (!bus->drm_power_refcount++) { + if (acomp->ops->get_power) + acomp->ops->get_power(acomp->dev); + snd_hdac_set_codec_wakeup(bus, true); + snd_hdac_set_codec_wakeup(bus, false); + } + } else { + WARN_ON(!bus->drm_power_refcount); + if (!--bus->drm_power_refcount) + if (acomp->ops->put_power) + acomp->ops->put_power(acomp->dev); + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_display_power); + +/** + * snd_hdac_sync_audio_rate - Set N/CTS based on the sample rate + * @codec: HDA codec + * @nid: the pin widget NID + * @dev_id: device identifier + * @rate: the sample rate to set + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function sets N/CTS value based on the given sample rate. + * Returns zero for success, or a negative error code. + */ +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, + int dev_id, int rate) +{ + struct hdac_bus *bus = codec->bus; + struct drm_audio_component *acomp = bus->audio_component; + int port, pipe; + + if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate) + return -ENODEV; + port = nid; + if (acomp->audio_ops && acomp->audio_ops->pin2port) { + port = acomp->audio_ops->pin2port(codec, nid); + if (port < 0) + return -EINVAL; + } + pipe = dev_id; + return acomp->ops->sync_audio_rate(acomp->dev, port, pipe, rate); +} +EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); + +/** + * snd_hdac_acomp_get_eld - Get the audio state and ELD via component + * @codec: HDA codec + * @nid: the pin widget NID + * @dev_id: device identifier + * @audio_enabled: the pointer to store the current audio state + * @buffer: the buffer pointer to store ELD bytes + * @max_bytes: the max bytes to be stored on @buffer + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function queries the current state of the audio on the given + * digital port and fetches the ELD bytes onto the given buffer. + * It returns the number of bytes for the total ELD data, zero for + * invalid ELD, or a negative error code. + * + * The return size is the total bytes required for the whole ELD bytes, + * thus it may be over @max_bytes. If it's over @max_bytes, it implies + * that only a part of ELD bytes have been fetched. + */ +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, + bool *audio_enabled, char *buffer, int max_bytes) +{ + struct hdac_bus *bus = codec->bus; + struct drm_audio_component *acomp = bus->audio_component; + int port, pipe; + + if (!acomp || !acomp->ops || !acomp->ops->get_eld) + return -ENODEV; + + port = nid; + if (acomp->audio_ops && acomp->audio_ops->pin2port) { + port = acomp->audio_ops->pin2port(codec, nid); + if (port < 0) + return -EINVAL; + } + pipe = dev_id; + return acomp->ops->get_eld(acomp->dev, port, pipe, audio_enabled, + buffer, max_bytes); +} +EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); + +static int hdac_component_master_bind(struct device *dev) +{ + struct drm_audio_component *acomp = hdac_get_acomp(dev); + int ret; + + if (WARN_ON(!acomp)) + return -EINVAL; + + ret = component_bind_all(dev, acomp); + if (ret < 0) + return ret; + + if (WARN_ON(!(acomp->dev && acomp->ops))) { + ret = -EINVAL; + goto out_unbind; + } + + /* pin the module to avoid dynamic unbinding, but only if given */ + if (!try_module_get(acomp->ops->owner)) { + ret = -ENODEV; + goto out_unbind; + } + + if (acomp->audio_ops && acomp->audio_ops->master_bind) { + ret = acomp->audio_ops->master_bind(dev, acomp); + if (ret < 0) + goto module_put; + } + + return 0; + + module_put: + module_put(acomp->ops->owner); +out_unbind: + component_unbind_all(dev, acomp); + + return ret; +} + +static void hdac_component_master_unbind(struct device *dev) +{ + struct drm_audio_component *acomp = hdac_get_acomp(dev); + + if (acomp->audio_ops && acomp->audio_ops->master_unbind) + acomp->audio_ops->master_unbind(dev, acomp); + module_put(acomp->ops->owner); + component_unbind_all(dev, acomp); + WARN_ON(acomp->ops || acomp->dev); +} + +static const struct component_master_ops hdac_component_master_ops = { + .bind = hdac_component_master_bind, + .unbind = hdac_component_master_unbind, +}; + +/** + * snd_hdac_acomp_register_notifier - Register audio component ops + * @bus: HDA core bus + * @aops: audio component ops + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function sets the given ops to be called by the graphics driver. + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_acomp_register_notifier(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *aops) +{ + if (!bus->audio_component) + return -ENODEV; + + bus->audio_component->audio_ops = aops; + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_acomp_register_notifier); + +/** + * snd_hdac_acomp_init - Initialize audio component + * @bus: HDA core bus + * @match_master: match function for finding components + * @extra_size: Extra bytes to allocate + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function initializes and sets up the audio component to communicate + * with graphics driver. + * + * Unlike snd_hdac_i915_init(), this function doesn't synchronize with the + * binding with the DRM component. Each caller needs to sync via master_bind + * audio_ops. + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_acomp_init(struct hdac_bus *bus, + const struct drm_audio_component_audio_ops *aops, + int (*match_master)(struct device *, void *), + size_t extra_size) +{ + struct component_match *match = NULL; + struct device *dev = bus->dev; + struct drm_audio_component *acomp; + int ret; + + if (WARN_ON(hdac_get_acomp(dev))) + return -EBUSY; + + acomp = devres_alloc(hdac_acomp_release, sizeof(*acomp) + extra_size, + GFP_KERNEL); + if (!acomp) + return -ENOMEM; + acomp->audio_ops = aops; + bus->audio_component = acomp; + devres_add(dev, acomp); + + component_match_add(dev, &match, match_master, bus); + ret = component_master_add_with_match(dev, &hdac_component_master_ops, + match); + if (ret < 0) + goto out_err; + + return 0; + +out_err: + bus->audio_component = NULL; + devres_destroy(dev, hdac_acomp_release, NULL, NULL); + dev_info(dev, "failed to add audio component master (%d)\n", ret); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_hdac_acomp_init); + +/** + * snd_hdac_acomp_exit - Finalize audio component + * @bus: HDA core bus + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with graphics driver. + * + * This function releases the audio component that has been used. + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_acomp_exit(struct hdac_bus *bus) +{ + struct device *dev = bus->dev; + struct drm_audio_component *acomp = bus->audio_component; + + if (!acomp) + return 0; + + WARN_ON(bus->drm_power_refcount); + if (bus->drm_power_refcount > 0 && acomp->ops) + acomp->ops->put_power(acomp->dev); + + component_master_del(dev, &hdac_component_master_ops); + + bus->audio_component = NULL; + devres_destroy(dev, hdac_acomp_release, NULL, NULL); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_acomp_exit); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 7ba100bb1c3f..dbf02a3a8d2f 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -738,7 +738,7 @@ static struct hda_rate_tbl rate_bits[] = { */ unsigned int snd_hdac_calc_stream_format(unsigned int rate, unsigned int channels, - unsigned int format, + snd_pcm_format_t format, unsigned int maxbps, unsigned short spdif_ctls) { diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index cbe818eda336..b5282cbbe489 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -15,88 +15,12 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/pci.h> -#include <linux/component.h> -#include <drm/i915_component.h> #include <sound/core.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> #include <sound/hda_register.h> -static struct i915_audio_component *hdac_acomp; - -/** - * snd_hdac_set_codec_wakeup - Enable / disable HDMI/DP codec wakeup - * @bus: HDA core bus - * @enable: enable or disable the wakeup - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function should be called during the chip reset, also called at - * resume for updating STATESTS register read. - * - * Returns zero for success or a negative error code. - */ -int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) -{ - struct i915_audio_component *acomp = bus->audio_component; - - if (!acomp || !acomp->ops) - return -ENODEV; - - if (!acomp->ops->codec_wake_override) { - dev_warn(bus->dev, - "Invalid codec wake callback\n"); - return 0; - } - - dev_dbg(bus->dev, "%s codec wakeup\n", - enable ? "enable" : "disable"); - - acomp->ops->codec_wake_override(acomp->dev, enable); - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup); - -/** - * snd_hdac_display_power - Power up / down the power refcount - * @bus: HDA core bus - * @enable: power up or down - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function manages a refcount and calls the i915 get_power() and - * put_power() ops accordingly, toggling the codec wakeup, too. - * - * Returns zero for success or a negative error code. - */ -int snd_hdac_display_power(struct hdac_bus *bus, bool enable) -{ - struct i915_audio_component *acomp = bus->audio_component; - - if (!acomp || !acomp->ops) - return -ENODEV; - - dev_dbg(bus->dev, "display power %s\n", - enable ? "enable" : "disable"); - - if (enable) { - if (!bus->i915_power_refcount++) { - acomp->ops->get_power(acomp->dev); - snd_hdac_set_codec_wakeup(bus, true); - snd_hdac_set_codec_wakeup(bus, false); - } - } else { - WARN_ON(!bus->i915_power_refcount); - if (!--bus->i915_power_refcount) - acomp->ops->put_power(acomp->dev); - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hdac_display_power); +static struct completion bind_complete; #define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ ((pci)->device == 0x0c0c) || \ @@ -119,7 +43,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_display_power); */ void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { - struct i915_audio_component *acomp = bus->audio_component; + struct drm_audio_component *acomp = bus->audio_component; struct pci_dev *pci = to_pci_dev(bus->dev); int cdclk_freq; unsigned int bclk_m, bclk_n; @@ -158,181 +82,11 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus) } EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk); -/* There is a fixed mapping between audio pin node and display port. - * on SNB, IVY, HSW, BSW, SKL, BXT, KBL: - * Pin Widget 5 - PORT B (port = 1 in i915 driver) - * Pin Widget 6 - PORT C (port = 2 in i915 driver) - * Pin Widget 7 - PORT D (port = 3 in i915 driver) - * - * on VLV, ILK: - * Pin Widget 4 - PORT B (port = 1 in i915 driver) - * Pin Widget 5 - PORT C (port = 2 in i915 driver) - * Pin Widget 6 - PORT D (port = 3 in i915 driver) - */ -static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) -{ - int base_nid; - - switch (codec->vendor_id) { - case 0x80860054: /* ILK */ - case 0x80862804: /* ILK */ - case 0x80862882: /* VLV */ - base_nid = 3; - break; - default: - base_nid = 4; - break; - } - - if (WARN_ON(pin_nid <= base_nid || pin_nid > base_nid + 3)) - return -1; - return pin_nid - base_nid; -} - -/** - * snd_hdac_sync_audio_rate - Set N/CTS based on the sample rate - * @codec: HDA codec - * @nid: the pin widget NID - * @dev_id: device identifier - * @rate: the sample rate to set - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function sets N/CTS value based on the given sample rate. - * Returns zero for success, or a negative error code. - */ -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, - int dev_id, int rate) -{ - struct hdac_bus *bus = codec->bus; - struct i915_audio_component *acomp = bus->audio_component; - int port, pipe; - - if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate) - return -ENODEV; - port = pin2port(codec, nid); - if (port < 0) - return -EINVAL; - pipe = dev_id; - return acomp->ops->sync_audio_rate(acomp->dev, port, pipe, rate); -} -EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); - -/** - * snd_hdac_acomp_get_eld - Get the audio state and ELD via component - * @codec: HDA codec - * @nid: the pin widget NID - * @dev_id: device identifier - * @audio_enabled: the pointer to store the current audio state - * @buffer: the buffer pointer to store ELD bytes - * @max_bytes: the max bytes to be stored on @buffer - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function queries the current state of the audio on the given - * digital port and fetches the ELD bytes onto the given buffer. - * It returns the number of bytes for the total ELD data, zero for - * invalid ELD, or a negative error code. - * - * The return size is the total bytes required for the whole ELD bytes, - * thus it may be over @max_bytes. If it's over @max_bytes, it implies - * that only a part of ELD bytes have been fetched. - */ -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, - bool *audio_enabled, char *buffer, int max_bytes) -{ - struct hdac_bus *bus = codec->bus; - struct i915_audio_component *acomp = bus->audio_component; - int port, pipe; - - if (!acomp || !acomp->ops || !acomp->ops->get_eld) - return -ENODEV; - - port = pin2port(codec, nid); - if (port < 0) - return -EINVAL; - - pipe = dev_id; - return acomp->ops->get_eld(acomp->dev, port, pipe, audio_enabled, - buffer, max_bytes); -} -EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); - -static int hdac_component_master_bind(struct device *dev) -{ - struct i915_audio_component *acomp = hdac_acomp; - int ret; - - ret = component_bind_all(dev, acomp); - if (ret < 0) - return ret; - - if (WARN_ON(!(acomp->dev && acomp->ops && acomp->ops->get_power && - acomp->ops->put_power && acomp->ops->get_cdclk_freq))) { - ret = -EINVAL; - goto out_unbind; - } - - /* - * Atm, we don't support dynamic unbinding initiated by the child - * component, so pin its containing module until we unbind. - */ - if (!try_module_get(acomp->ops->owner)) { - ret = -ENODEV; - goto out_unbind; - } - - return 0; - -out_unbind: - component_unbind_all(dev, acomp); - - return ret; -} - -static void hdac_component_master_unbind(struct device *dev) -{ - struct i915_audio_component *acomp = hdac_acomp; - - module_put(acomp->ops->owner); - component_unbind_all(dev, acomp); - WARN_ON(acomp->ops || acomp->dev); -} - -static const struct component_master_ops hdac_component_master_ops = { - .bind = hdac_component_master_bind, - .unbind = hdac_component_master_unbind, -}; - -static int hdac_component_master_match(struct device *dev, void *data) +static int i915_component_master_match(struct device *dev, void *data) { - /* i915 is the only supported component */ return !strcmp(dev->driver->name, "i915"); } -/** - * snd_hdac_i915_register_notifier - Register i915 audio component ops - * @aops: i915 audio component ops - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function sets the given ops to be called by the i915 graphics driver. - * - * Returns zero for success or a negative error code. - */ -int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *aops) -{ - if (!hdac_acomp) - return -ENODEV; - - hdac_acomp->audio_ops = aops; - return 0; -} -EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier); - /* check whether intel graphics is present */ static bool i915_gfx_present(void) { @@ -345,6 +99,19 @@ static bool i915_gfx_present(void) return pci_dev_present(ids); } +static int i915_master_bind(struct device *dev, + struct drm_audio_component *acomp) +{ + complete_all(&bind_complete); + /* clear audio_ops here as it was needed only for completion call */ + acomp->audio_ops = NULL; + return 0; +} + +static const struct drm_audio_component_audio_ops i915_init_ops = { + .master_bind = i915_master_bind +}; + /** * snd_hdac_i915_init - Initialize i915 audio component * @bus: HDA core bus @@ -359,83 +126,31 @@ static bool i915_gfx_present(void) */ int snd_hdac_i915_init(struct hdac_bus *bus) { - struct component_match *match = NULL; - struct device *dev = bus->dev; - struct i915_audio_component *acomp; - int ret; - - if (WARN_ON(hdac_acomp)) - return -EBUSY; + struct drm_audio_component *acomp; + int err; if (!i915_gfx_present()) return -ENODEV; - acomp = kzalloc(sizeof(*acomp), GFP_KERNEL); - if (!acomp) - return -ENOMEM; - bus->audio_component = acomp; - hdac_acomp = acomp; - - component_match_add(dev, &match, hdac_component_master_match, bus); - ret = component_master_add_with_match(dev, &hdac_component_master_ops, - match); - if (ret < 0) - goto out_err; - - /* - * Atm, we don't support deferring the component binding, so make sure - * i915 is loaded and that the binding successfully completes. - */ - request_module("i915"); + init_completion(&bind_complete); + err = snd_hdac_acomp_init(bus, &i915_init_ops, + i915_component_master_match, + sizeof(struct i915_audio_component) - sizeof(*acomp)); + if (err < 0) + return err; + acomp = bus->audio_component; + if (!acomp) + return -ENODEV; if (!acomp->ops) { - ret = -ENODEV; - goto out_master_del; + request_module("i915"); + /* 10s timeout */ + wait_for_completion_timeout(&bind_complete, 10 * 1000); + } + if (!acomp->ops) { + snd_hdac_acomp_exit(bus); + return -ENODEV; } - dev_dbg(dev, "bound to i915 component master\n"); - return 0; -out_master_del: - component_master_del(dev, &hdac_component_master_ops); -out_err: - kfree(acomp); - bus->audio_component = NULL; - hdac_acomp = NULL; - dev_info(dev, "failed to add i915 component master (%d)\n", ret); - - return ret; } EXPORT_SYMBOL_GPL(snd_hdac_i915_init); - -/** - * snd_hdac_i915_exit - Finalize i915 audio component - * @bus: HDA core bus - * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. - * - * This function releases the i915 audio component that has been used. - * - * Returns zero for success or a negative error code. - */ -int snd_hdac_i915_exit(struct hdac_bus *bus) -{ - struct device *dev = bus->dev; - struct i915_audio_component *acomp = bus->audio_component; - - if (!acomp) - return 0; - - WARN_ON(bus->i915_power_refcount); - if (bus->i915_power_refcount > 0 && acomp->ops) - acomp->ops->put_power(acomp->dev); - - component_master_del(dev, &hdac_component_master_ops); - - kfree(acomp); - bus->audio_component = NULL; - hdac_acomp = NULL; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hdac_i915_exit); diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index e1472c7ab6c1..eee422390d8e 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -621,7 +621,7 @@ int snd_hdac_dsp_prepare(struct hdac_stream *azx_dev, unsigned int format, unsigned int byte_size, struct snd_dma_buffer *bufp) { struct hdac_bus *bus = azx_dev->bus; - u32 *bdl; + __le32 *bdl; int err; snd_hdac_dsp_lock(azx_dev); @@ -651,7 +651,7 @@ int snd_hdac_dsp_prepare(struct hdac_stream *azx_dev, unsigned int format, snd_hdac_stream_writel(azx_dev, SD_BDLPU, 0); azx_dev->frags = 0; - bdl = (u32 *)azx_dev->bdl.area; + bdl = (__le32 *)azx_dev->bdl.area; err = setup_bdle(bus, bufp, azx_dev, &bdl, 0, byte_size, 0); if (err < 0) goto error; diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 7e21621e492a..2647309bc675 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -621,15 +621,3 @@ int snd_cs8427_iec958_pcm(struct snd_i2c_device *cs8427, unsigned int rate) } EXPORT_SYMBOL(snd_cs8427_iec958_pcm); - -static int __init alsa_cs8427_module_init(void) -{ - return 0; -} - -static void __exit alsa_cs8427_module_exit(void) -{ -} - -module_init(alsa_cs8427_module_init) -module_exit(alsa_cs8427_module_exit) diff --git a/sound/i2c/i2c.c b/sound/i2c/i2c.c index ef2a9afe9e19..c4a232f18a79 100644 --- a/sound/i2c/i2c.c +++ b/sound/i2c/i2c.c @@ -338,16 +338,3 @@ static int snd_i2c_bit_probeaddr(struct snd_i2c_bus *bus, unsigned short addr) snd_i2c_bit_stop(bus); return err; } - - -static int __init alsa_i2c_init(void) -{ - return 0; -} - -static void __exit alsa_i2c_exit(void) -{ -} - -module_init(alsa_i2c_init) -module_exit(alsa_i2c_exit) diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index bf377dc192aa..7f2761a2e7c8 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -911,15 +911,3 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak) return 0; } EXPORT_SYMBOL(snd_akm4xxx_build_controls); - -static int __init alsa_akm4xxx_module_init(void) -{ - return 0; -} - -static void __exit alsa_akm4xxx_module_exit(void) -{ -} - -module_init(alsa_akm4xxx_module_init) -module_exit(alsa_akm4xxx_module_exit) diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c index 2d22310dce05..239c4822427f 100644 --- a/sound/i2c/tea6330t.c +++ b/sound/i2c/tea6330t.c @@ -368,19 +368,3 @@ int snd_tea6330t_update_mixer(struct snd_card *card, EXPORT_SYMBOL(snd_tea6330t_detect); EXPORT_SYMBOL(snd_tea6330t_update_mixer); - -/* - * INIT part - */ - -static int __init alsa_tea6330t_init(void) -{ - return 0; -} - -static void __exit alsa_tea6330t_exit(void) -{ -} - -module_init(alsa_tea6330t_init) -module_exit(alsa_tea6330t_exit) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 43b35a873d78..d7db1eeebc84 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -459,7 +459,7 @@ config SND_MSND_CLASSIC Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or Monterey (not for the Pinnacle or Fiji). - See <file:Documentation/sound/oss/MultiSound> for important information + See <file:Documentation/sound/cards/multisound.sh> for important information about this driver. Note that it has been discontinued, but the Voyetra Turtle Beach knowledge base entry for it is still available at <http://www.turtlebeach.com/site/kb_ftp/790.asp>. diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index 923201414469..fba6d22f7f4b 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -85,7 +85,8 @@ static void snd_ad1816a_write_mask(struct snd_ad1816a *chip, unsigned char reg, static unsigned char snd_ad1816a_get_format(struct snd_ad1816a *chip, - unsigned int format, int channels) + snd_pcm_format_t format, + int channels) { unsigned char retval = AD1816A_FMT_LINEAR_8; diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index a826c138e7f5..3dfe7e592c25 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -260,7 +260,6 @@ static int snd_es968_pnp_detect(struct pnp_card_link *pcard, struct snd_card *card; static unsigned int dev; int error; - struct snd_es1688 *chip; if (snd_es968_pnp_is_probed) return -EBUSY; @@ -276,7 +275,6 @@ static int snd_es968_pnp_detect(struct pnp_card_link *pcard, sizeof(struct snd_es1688), &card); if (error < 0) return error; - chip = card->private_data; error = snd_card_es968_pnp(card, dev, pcard, pid); if (error < 0) { diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index f9c0662e9a22..50cdce0e8946 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -1029,19 +1029,3 @@ EXPORT_SYMBOL(snd_es1688_mixer_write); EXPORT_SYMBOL(snd_es1688_create); EXPORT_SYMBOL(snd_es1688_pcm); EXPORT_SYMBOL(snd_es1688_mixer); - -/* - * INIT part - */ - -static int __init alsa_es1688_init(void) -{ - return 0; -} - -static void __exit alsa_es1688_exit(void) -{ -} - -module_init(alsa_es1688_init) -module_exit(alsa_es1688_exit) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 2a6960c3e2a4..0d103d6f805e 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1024,6 +1024,7 @@ static int snd_es18xx_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem val = 3; } else retVal = snd_es18xx_mixer_bits(chip, 0x7a, 0x08, 0x00) != 0x00; + /* fall through */ /* 4 source chips */ case 0x1868: case 0x1878: diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c index b9994cc9f5fb..af9eea41379f 100644 --- a/sound/isa/galaxy/galaxy.c +++ b/sound/isa/galaxy/galaxy.c @@ -260,6 +260,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) break; case 2: irq[n] = 9; + /* Fall through */ case 9: wss_config[n] |= WSS_CONFIG_IRQ_9; break; @@ -304,6 +305,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) case 1: if (dma1[n] == 0) break; + /* Fall through */ default: dev_err(dev, "invalid capture DMA %d\n", dma2[n]); return 0; @@ -333,6 +335,7 @@ mpu: break; case 2: mpu_irq[n] = 9; + /* Fall through */ case 9: config[n] |= GALAXY_CONFIG_MPUIRQ_2; break; diff --git a/sound/isa/gus/gus_io.c b/sound/isa/gus/gus_io.c index ca79878d8d8c..2fd32ef22c30 100644 --- a/sound/isa/gus/gus_io.c +++ b/sound/isa/gus/gus_io.c @@ -461,7 +461,7 @@ void snd_gf1_print_voice_registers(struct snd_gus_card * gus) printk(KERN_INFO " -%i- GFA1 effect address = 0x%x\n", voice, snd_gf1_i_read_addr(gus, 0x11, ctrl & 4)); printk(KERN_INFO " -%i- GFA1 effect volume = 0x%x\n", voice, snd_gf1_i_read16(gus, 0x16)); printk(KERN_INFO " -%i- GFA1 effect volume final = 0x%x\n", voice, snd_gf1_i_read16(gus, 0x1d)); - printk(KERN_INFO " -%i- GFA1 effect acumulator = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x14)); + printk(KERN_INFO " -%i- GFA1 effect accumulator = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x14)); } if (mode & 0x20) { printk(KERN_INFO " -%i- GFA1 left offset = 0x%x (%i)\n", voice, snd_gf1_i_read16(gus, 0x13), snd_gf1_i_read16(gus, 0x13) >> 4); diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index 3cf9b13c780a..3b8a0c880db5 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -465,19 +465,3 @@ EXPORT_SYMBOL(snd_gf1_mem_alloc); EXPORT_SYMBOL(snd_gf1_mem_xfree); EXPORT_SYMBOL(snd_gf1_mem_free); EXPORT_SYMBOL(snd_gf1_mem_lock); - -/* - * INIT part - */ - -static int __init alsa_gus_init(void) -{ - return 0; -} - -static void __exit alsa_gus_exit(void) -{ -} - -module_init(alsa_gus_init) -module_exit(alsa_gus_exit) diff --git a/sound/isa/gus/gus_reset.c b/sound/isa/gus/gus_reset.c index 3d1fed0c2620..59b3f683d49b 100644 --- a/sound/isa/gus/gus_reset.c +++ b/sound/isa/gus/gus_reset.c @@ -292,7 +292,6 @@ void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice) { unsigned long flags; void (*private_free)(struct snd_gus_voice *voice); - void *private_data; if (voice == NULL || !voice->use) return; @@ -300,7 +299,6 @@ void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice) snd_gf1_clear_voices(gus, voice->number, voice->number); spin_lock_irqsave(&gus->voice_alloc, flags); private_free = voice->private_free; - private_data = voice->private_data; voice->private_free = NULL; voice->private_data = NULL; if (voice->pcm) diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c index 569897f64fda..7c3203fe4869 100644 --- a/sound/isa/msnd/msnd.c +++ b/sound/isa/msnd/msnd.c @@ -54,7 +54,7 @@ #define LOGNAME "msnd" -void snd_msnd_init_queue(void *base, int start, int size) +void snd_msnd_init_queue(void __iomem *base, int start, int size) { writew(PCTODSP_BASED(start), base + JQS_wStart); writew(PCTODSP_OFFSET(size) - 1, base + JQS_wSize); @@ -270,7 +270,7 @@ int snd_msnd_DARQ(struct snd_msnd *chip, int bank) udelay(1); if (chip->capturePeriods == 2) { - void *pDAQ = chip->mappedbase + DARQ_DATA_BUFF + + void __iomem *pDAQ = chip->mappedbase + DARQ_DATA_BUFF + bank * DAQDS__size + DAQDS_wStart; unsigned short offset = 0x3000 + chip->capturePeriodBytes; @@ -309,7 +309,7 @@ int snd_msnd_DAPQ(struct snd_msnd *chip, int start) { u16 DAPQ_tail; int protect = start, nbanks = 0; - void *DAQD; + void __iomem *DAQD; static int play_banks_submitted; /* unsigned long flags; spin_lock_irqsave(&chip->lock, flags); not necessary */ @@ -370,7 +370,7 @@ static void snd_msnd_play_reset_queue(struct snd_msnd *chip, unsigned int pcm_count) { int n; - void *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; + void __iomem *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; chip->last_playbank = -1; chip->playLimit = pcm_count * (pcm_periods - 1); @@ -398,7 +398,7 @@ static void snd_msnd_capture_reset_queue(struct snd_msnd *chip, unsigned int pcm_count) { int n; - void *pDAQ; + void __iomem *pDAQ; /* unsigned long flags; */ /* snd_msnd_init_queue(chip->DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE); */ @@ -485,7 +485,7 @@ static int snd_msnd_playback_open(struct snd_pcm_substream *substream) clear_bit(F_WRITING, &chip->flags); snd_msnd_enable_irq(chip); - runtime->dma_area = chip->mappedbase; + runtime->dma_area = (__force void *)chip->mappedbase; runtime->dma_bytes = 0x3000; chip->playback_substream = substream; @@ -508,7 +508,7 @@ static int snd_msnd_playback_hw_params(struct snd_pcm_substream *substream, { int i; struct snd_msnd *chip = snd_pcm_substream_chip(substream); - void *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; + void __iomem *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; chip->play_sample_size = snd_pcm_format_width(params_format(params)); chip->play_channels = params_channels(params); @@ -589,7 +589,7 @@ static int snd_msnd_capture_open(struct snd_pcm_substream *substream) set_bit(F_AUDIO_READ_INUSE, &chip->flags); snd_msnd_enable_irq(chip); - runtime->dma_area = chip->mappedbase + 0x3000; + runtime->dma_area = (__force void *)chip->mappedbase + 0x3000; runtime->dma_bytes = 0x3000; memset(runtime->dma_area, 0, runtime->dma_bytes); chip->capture_substream = substream; @@ -654,7 +654,7 @@ static int snd_msnd_capture_hw_params(struct snd_pcm_substream *substream, { int i; struct snd_msnd *chip = snd_pcm_substream_chip(substream); - void *pDAQ = chip->mappedbase + DARQ_DATA_BUFF; + void __iomem *pDAQ = chip->mappedbase + DARQ_DATA_BUFF; chip->capture_sample_size = snd_pcm_format_width(params_format(params)); chip->capture_channels = params_channels(params); diff --git a/sound/isa/msnd/msnd.h b/sound/isa/msnd/msnd.h index 5f3c7dcd9f9d..80c718757eef 100644 --- a/sound/isa/msnd/msnd.h +++ b/sound/isa/msnd/msnd.h @@ -283,7 +283,7 @@ struct snd_msnd { }; -void snd_msnd_init_queue(void *base, int start, int size); +void snd_msnd_init_queue(void __iomem *base, int start, int size); int snd_msnd_send_dsp_cmd(struct snd_msnd *chip, u8 cmd); int snd_msnd_send_word(struct snd_msnd *chip, diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c index 013d8d1170fe..42876b0cb68b 100644 --- a/sound/isa/msnd/msnd_midi.c +++ b/sound/isa/msnd/msnd_midi.c @@ -119,7 +119,7 @@ void snd_msndmidi_input_read(void *mpuv) { unsigned long flags; struct snd_msndmidi *mpu = mpuv; - void *pwMIDQData = mpu->dev->mappedbase + MIDQ_DATA_BUFF; + void __iomem *pwMIDQData = mpu->dev->mappedbase + MIDQ_DATA_BUFF; u16 head, tail, size; spin_lock_irqsave(&mpu->input_lock, flags); diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 6c584d9b6c42..11af9c40bc05 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -82,10 +82,10 @@ static void set_default_audio_parameters(struct snd_msnd *chip) { - chip->play_sample_size = DEFSAMPLESIZE; + chip->play_sample_size = snd_pcm_format_width(DEFSAMPLESIZE); chip->play_sample_rate = DEFSAMPLERATE; chip->play_channels = DEFCHANNELS; - chip->capture_sample_size = DEFSAMPLESIZE; + chip->capture_sample_size = snd_pcm_format_width(DEFSAMPLESIZE); chip->capture_sample_rate = DEFSAMPLERATE; chip->capture_channels = DEFCHANNELS; } @@ -169,7 +169,7 @@ static void snd_msnd_eval_dsp_msg(struct snd_msnd *chip, u16 wMessage) static irqreturn_t snd_msnd_interrupt(int irq, void *dev_id) { struct snd_msnd *chip = dev_id; - void *pwDSPQData = chip->mappedbase + DSPQ_DATA_BUFF; + void __iomem *pwDSPQData = chip->mappedbase + DSPQ_DATA_BUFF; u16 head, tail, size; /* Send ack to DSP */ @@ -810,7 +810,7 @@ module_param(calibrate_signal, int, 0444); #ifndef MSND_CLASSIC module_param_array(digital, int, NULL, 0444); module_param_hw_array(cfg, long, ioport, NULL, 0444); -module_param_array(reset, int, 0, 0444); +module_param_array(reset, int, NULL, 0444); module_param_hw_array(mpu_io, long, ioport, NULL, 0444); module_param_hw_array(mpu_irq, int, irq, NULL, 0444); module_param_hw_array(ide_io0, long, ioport, NULL, 0444); diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 8894c7c18ad6..c6136c6b0214 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -176,10 +176,13 @@ static int aci_busy_wait(struct snd_miro_aci *aci) switch (timeout-ACI_MINTIME) { case 0 ... 9: out /= 10; + /* fall through */ case 10 ... 19: out /= 10; + /* fall through */ case 20 ... 30: out /= 10; + /* fall through */ default: set_current_state(TASK_UNINTERRUPTIBLE); schedule_timeout(out); @@ -834,6 +837,7 @@ static unsigned char snd_miro_read(struct snd_miro *chip, retval = inb(chip->mc_base + 9); break; } + /* fall through */ case OPTi9XX_HW_82C929: retval = inb(chip->mc_base + reg); @@ -863,6 +867,7 @@ static void snd_miro_write(struct snd_miro *chip, unsigned char reg, outb(value, chip->mc_base + 9); break; } + /* fall through */ case OPTi9XX_HW_82C929: outb(value, chip->mc_base + reg); diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 505cd81e19fa..ac0ab6eb40f0 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -261,6 +261,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip, retval = inb(chip->mc_base + 9); break; } + /* Fall through */ case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: @@ -303,6 +304,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, outb(value, chip->mc_base + 9); break; } + /* Fall through */ case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: @@ -350,6 +352,7 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0xf0, 0xfc); /* enable wave audio */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02); + /* Fall through */ case OPTi9XX_HW_82C925: /* enable WSS mode */ diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c index c2e41d2762f7..d45a6b9d6437 100644 --- a/sound/isa/sb/emu8000_patch.c +++ b/sound/isa/sb/emu8000_patch.c @@ -165,11 +165,8 @@ snd_emu8000_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, return 0; /* be sure loop points start < end */ - if (sp->v.loopstart > sp->v.loopend) { - int tmp = sp->v.loopstart; - sp->v.loopstart = sp->v.loopend; - sp->v.loopend = tmp; - } + if (sp->v.loopstart > sp->v.loopend) + swap(sp->v.loopstart, sp->v.loopend); /* compute true data size to be loaded */ truesize = sp->v.size; diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c index bc5af71d3bdb..f46f6ec3ea0c 100644 --- a/sound/isa/sb/emu8000_pcm.c +++ b/sound/isa/sb/emu8000_pcm.c @@ -470,7 +470,7 @@ static int emu8k_pcm_copy(struct snd_pcm_substream *subs, /* convert to word unit */ pos = (pos << 1) + rec->loop_start[voice]; count <<= 1; - LOOP_WRITE(rec, pos, src, count, COPY_UESR); + LOOP_WRITE(rec, pos, src, count, COPY_USER); return 0; } diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index fa5780bb0c68..bf3db0d2ea12 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -60,18 +60,18 @@ MODULE_FIRMWARE("sb16/ima_adpcm_capture.csp"); * RIFF data format */ struct riff_header { - __u32 name; - __u32 len; + __le32 name; + __le32 len; }; struct desc_header { struct riff_header info; - __u16 func_nr; - __u16 VOC_type; - __u16 flags_play_rec; - __u16 flags_16bit_8bit; - __u16 flags_stereo_mono; - __u16 flags_rates; + __le16 func_nr; + __le16 VOC_type; + __le16 flags_play_rec; + __le16 flags_16bit_8bit; + __le16 flags_stereo_mono; + __le16 flags_rates; }; /* @@ -93,7 +93,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, struct snd_sb_csp_microcode __user * code); static int snd_sb_csp_unload(struct snd_sb_csp * p); static int snd_sb_csp_load_user(struct snd_sb_csp * p, const unsigned char __user *buf, int size, int load_flags); -static int snd_sb_csp_autoload(struct snd_sb_csp * p, int pcm_sfmt, int play_rec_mode); +static int snd_sb_csp_autoload(struct snd_sb_csp * p, snd_pcm_format_t pcm_sfmt, int play_rec_mode); static int snd_sb_csp_check_version(struct snd_sb_csp * p); static int snd_sb_csp_use(struct snd_sb_csp * p); @@ -314,7 +314,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, unsigned short func_nr = 0; struct riff_header file_h, item_h, code_h; - __u32 item_type; + __le32 item_type; struct desc_header funcdesc_h; unsigned long flags; @@ -326,7 +326,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, if (copy_from_user(&file_h, data_ptr, sizeof(file_h))) return -EFAULT; - if ((file_h.name != RIFF_HEADER) || + if ((le32_to_cpu(file_h.name) != RIFF_HEADER) || (le32_to_cpu(file_h.len) >= SNDRV_SB_CSP_MAX_MICROCODE_FILE_SIZE - sizeof(file_h))) { snd_printd("%s: Invalid RIFF header\n", __func__); return -EINVAL; @@ -336,7 +336,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, if (copy_from_user(&item_type, data_ptr, sizeof(item_type))) return -EFAULT; - if (item_type != CSP__HEADER) { + if (le32_to_cpu(item_type) != CSP__HEADER) { snd_printd("%s: Invalid RIFF file type\n", __func__); return -EINVAL; } @@ -346,12 +346,12 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, if (copy_from_user(&item_h, data_ptr, sizeof(item_h))) return -EFAULT; data_ptr += sizeof(item_h); - if (item_h.name != LIST_HEADER) + if (le32_to_cpu(item_h.name) != LIST_HEADER) continue; if (copy_from_user(&item_type, data_ptr, sizeof(item_type))) return -EFAULT; - switch (item_type) { + switch (le32_to_cpu(item_type)) { case FUNC_HEADER: if (copy_from_user(&funcdesc_h, data_ptr + sizeof(item_type), sizeof(funcdesc_h))) return -EFAULT; @@ -378,7 +378,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, return -EFAULT; /* init microcode blocks */ - if (code_h.name != INIT_HEADER) + if (le32_to_cpu(code_h.name) != INIT_HEADER) break; data_ptr += sizeof(code_h); err = snd_sb_csp_load_user(p, data_ptr, le32_to_cpu(code_h.len), @@ -391,7 +391,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, if (copy_from_user(&code_h, data_ptr, sizeof(code_h))) return -EFAULT; - if (code_h.name != MAIN_HEADER) { + if (le32_to_cpu(code_h.name) != MAIN_HEADER) { snd_printd("%s: Missing 'main' microcode\n", __func__); return -EINVAL; } @@ -726,7 +726,7 @@ static int snd_sb_csp_firmware_load(struct snd_sb_csp *p, int index, int flags) * autoload hardware codec if necessary * return 0 if CSP is loaded and ready to run (p->running != 0) */ -static int snd_sb_csp_autoload(struct snd_sb_csp * p, int pcm_sfmt, int play_rec_mode) +static int snd_sb_csp_autoload(struct snd_sb_csp * p, snd_pcm_format_t pcm_sfmt, int play_rec_mode) { unsigned long flags; int err = 0; @@ -736,7 +736,7 @@ static int snd_sb_csp_autoload(struct snd_sb_csp * p, int pcm_sfmt, int play_rec return -EBUSY; /* autoload microcode only if requested hardware codec is not already loaded */ - if (((1 << pcm_sfmt) & p->acc_format) && (play_rec_mode & p->mode)) { + if (((1U << (__force int)pcm_sfmt) & p->acc_format) && (play_rec_mode & p->mode)) { p->running = SNDRV_SB_CSP_ST_AUTO; } else { switch (pcm_sfmt) { @@ -1185,19 +1185,3 @@ static void info_read(struct snd_info_entry *entry, struct snd_info_buffer *buff /* */ EXPORT_SYMBOL(snd_sb_csp_new); - -/* - * INIT part - */ - -static int __init alsa_sb_csp_init(void) -{ - return 0; -} - -static void __exit alsa_sb_csp_exit(void) -{ -} - -module_init(alsa_sb_csp_init) -module_exit(alsa_sb_csp_exit) diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index 3e39ba220c39..37e6ce7b0b13 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -49,6 +49,9 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Routines for control of 16-bit SoundBlaster cards and clones"); MODULE_LICENSE("GPL"); +#define runtime_format_bits(runtime) \ + ((unsigned int)pcm_format_to_bits((runtime)->format)) + #ifdef CONFIG_SND_SB16_CSP static void snd_sb16_csp_playback_prepare(struct snd_sb *chip, struct snd_pcm_runtime *runtime) { @@ -58,7 +61,7 @@ static void snd_sb16_csp_playback_prepare(struct snd_sb *chip, struct snd_pcm_ru if (csp->running & SNDRV_SB_CSP_ST_LOADED) { /* manually loaded codec */ if ((csp->mode & SNDRV_SB_CSP_MODE_DSP_WRITE) && - ((1U << runtime->format) == csp->acc_format)) { + (runtime_format_bits(runtime) == csp->acc_format)) { /* Supported runtime PCM format for playback */ if (csp->ops.csp_use(csp) == 0) { /* If CSP was successfully acquired */ @@ -66,7 +69,7 @@ static void snd_sb16_csp_playback_prepare(struct snd_sb *chip, struct snd_pcm_ru } } else if ((csp->mode & SNDRV_SB_CSP_MODE_QSOUND) && (csp->q_enabled)) { /* QSound decoder is loaded and enabled */ - if ((1 << runtime->format) & (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + if (runtime_format_bits(runtime) & (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE)) { /* Only for simple PCM formats */ if (csp->ops.csp_use(csp) == 0) { @@ -106,7 +109,7 @@ static void snd_sb16_csp_capture_prepare(struct snd_sb *chip, struct snd_pcm_run if (csp->running & SNDRV_SB_CSP_ST_LOADED) { /* manually loaded codec */ if ((csp->mode & SNDRV_SB_CSP_MODE_DSP_READ) && - ((1U << runtime->format) == csp->acc_format)) { + (runtime_format_bits(runtime) == csp->acc_format)) { /* Supported runtime PCM format for capture */ if (csp->ops.csp_use(csp) == 0) { /* If CSP was successfully acquired */ @@ -897,19 +900,3 @@ EXPORT_SYMBOL(snd_sb16dsp_pcm); EXPORT_SYMBOL(snd_sb16dsp_get_pcm_ops); EXPORT_SYMBOL(snd_sb16dsp_configure); EXPORT_SYMBOL(snd_sb16dsp_interrupt); - -/* - * INIT part - */ - -static int __init alsa_sb16_init(void) -{ - return 0; -} - -static void __exit alsa_sb16_exit(void) -{ -} - -module_init(alsa_sb16_init) -module_exit(alsa_sb16_exit) diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index d45df5c54423..481797744b3c 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -381,7 +381,6 @@ static int snd_sb8_capture_trigger(struct snd_pcm_substream *substream, irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) { struct snd_pcm_substream *substream; - struct snd_pcm_runtime *runtime; snd_sb_ack_8bit(chip); switch (chip->mode) { @@ -391,7 +390,6 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) /* fallthru */ case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; - runtime = substream->runtime; if (chip->playback_format == SB_DSP_OUTPUT) snd_sb8_playback_trigger(substream, SNDRV_PCM_TRIGGER_START); snd_pcm_period_elapsed(substream); @@ -402,7 +400,6 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) /* fallthru */ case SB_MODE_CAPTURE_8: substream = chip->capture_substream; - runtime = substream->runtime; if (chip->capture_format == SB_DSP_INPUT) snd_sb8_capture_trigger(substream, SNDRV_PCM_TRIGGER_START); snd_pcm_period_elapsed(substream); @@ -624,19 +621,3 @@ EXPORT_SYMBOL(snd_sb8dsp_interrupt); /* sb8_midi.c */ EXPORT_SYMBOL(snd_sb8dsp_midi_interrupt); EXPORT_SYMBOL(snd_sb8dsp_midi); - -/* - * INIT part - */ - -static int __init alsa_sb8_init(void) -{ - return 0; -} - -static void __exit alsa_sb8_exit(void) -{ -} - -module_init(alsa_sb8_init) -module_exit(alsa_sb8_exit) diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 787a4ade4afd..90b254aaef74 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -305,19 +305,3 @@ EXPORT_SYMBOL(snd_sbmixer_add_ctl); EXPORT_SYMBOL(snd_sbmixer_suspend); EXPORT_SYMBOL(snd_sbmixer_resume); #endif - -/* - * INIT part - */ - -static int __init alsa_sb_common_init(void) -{ - return 0; -} - -static void __exit alsa_sb_common_exit(void) -{ -} - -module_init(alsa_sb_common_init) -module_exit(alsa_sb_common_exit) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 8a852042a066..32453f81b95a 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -541,7 +541,7 @@ static unsigned char snd_wss_get_rate(unsigned int rate) } static unsigned char snd_wss_get_format(struct snd_wss *chip, - int format, + snd_pcm_format_t format, int channels) { unsigned char rformat; @@ -2279,19 +2279,3 @@ const struct snd_pcm_ops *snd_wss_get_pcm_ops(int direction) &snd_wss_playback_ops : &snd_wss_capture_ops; } EXPORT_SYMBOL(snd_wss_get_pcm_ops); - -/* - * INIT part - */ - -static int __init alsa_wss_init(void) -{ - return 0; -} - -static void __exit alsa_wss_exit(void) -{ -} - -module_init(alsa_wss_init); -module_exit(alsa_wss_exit); diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 9fb68b35de5a..3ec9391a4736 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -685,7 +685,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = { @@ -698,7 +697,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops snd_sgio2audio_capture_ops = { @@ -711,7 +709,6 @@ static const struct snd_pcm_ops snd_sgio2audio_capture_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; /* diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 1ef7cdf1d3e8..f4459d1a9d67 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2941,19 +2941,3 @@ int snd_ac97_tune_hardware(struct snd_ac97 *ac97, } EXPORT_SYMBOL(snd_ac97_tune_hardware); - -/* - * INIT part - */ - -static int __init alsa_ac97_init(void) -{ - return 0; -} - -static void __exit alsa_ac97_exit(void) -{ -} - -module_init(alsa_ac97_init) -module_exit(alsa_ac97_exit) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 39547e32e584..9f569379b77e 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1484,12 +1484,9 @@ static struct snd_pcm_hardware snd_ali_capture = static void snd_ali_pcm_free_substream(struct snd_pcm_runtime *runtime) { struct snd_ali_voice *pvoice = runtime->private_data; - struct snd_ali *codec; - if (pvoice) { - codec = pvoice->codec; + if (pvoice) snd_ali_free_voice(pvoice->codec, pvoice); - } } static int snd_ali_open(struct snd_pcm_substream *substream, int rec, diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 64e0961f93ba..a31fe1550903 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -311,27 +311,29 @@ static void print_hwparams(struct snd_pcm_substream *substream, snd_pcm_format_width(params_format(p)) / 8); } +#define INVALID_FORMAT (__force snd_pcm_format_t)(-1) + static snd_pcm_format_t hpi_to_alsa_formats[] = { - -1, /* INVALID */ + INVALID_FORMAT, /* INVALID */ SNDRV_PCM_FORMAT_U8, /* HPI_FORMAT_PCM8_UNSIGNED 1 */ SNDRV_PCM_FORMAT_S16, /* HPI_FORMAT_PCM16_SIGNED 2 */ - -1, /* HPI_FORMAT_MPEG_L1 3 */ + INVALID_FORMAT, /* HPI_FORMAT_MPEG_L1 3 */ SNDRV_PCM_FORMAT_MPEG, /* HPI_FORMAT_MPEG_L2 4 */ SNDRV_PCM_FORMAT_MPEG, /* HPI_FORMAT_MPEG_L3 5 */ - -1, /* HPI_FORMAT_DOLBY_AC2 6 */ - -1, /* HPI_FORMAT_DOLBY_AC3 7 */ + INVALID_FORMAT, /* HPI_FORMAT_DOLBY_AC2 6 */ + INVALID_FORMAT, /* HPI_FORMAT_DOLBY_AC3 7 */ SNDRV_PCM_FORMAT_S16_BE,/* HPI_FORMAT_PCM16_BIGENDIAN 8 */ - -1, /* HPI_FORMAT_AA_TAGIT1_HITS 9 */ - -1, /* HPI_FORMAT_AA_TAGIT1_INSERTS 10 */ + INVALID_FORMAT, /* HPI_FORMAT_AA_TAGIT1_HITS 9 */ + INVALID_FORMAT, /* HPI_FORMAT_AA_TAGIT1_INSERTS 10 */ SNDRV_PCM_FORMAT_S32, /* HPI_FORMAT_PCM32_SIGNED 11 */ - -1, /* HPI_FORMAT_RAW_BITSTREAM 12 */ - -1, /* HPI_FORMAT_AA_TAGIT1_HITS_EX1 13 */ + INVALID_FORMAT, /* HPI_FORMAT_RAW_BITSTREAM 12 */ + INVALID_FORMAT, /* HPI_FORMAT_AA_TAGIT1_HITS_EX1 13 */ SNDRV_PCM_FORMAT_FLOAT, /* HPI_FORMAT_PCM32_FLOAT 14 */ #if 1 /* ALSA can't handle 3 byte sample size together with power-of-2 * constraint on buffer_bytes, so disable this format */ - -1 + INVALID_FORMAT #else /* SNDRV_PCM_FORMAT_S24_3LE */ /* HPI_FORMAT_PCM24_SIGNED 15 */ #endif @@ -1023,7 +1025,7 @@ static u64 snd_card_asihpi_playback_formats(struct snd_card_asihpi *asihpi, format, sample_rate, 128000, 0); if (!err) err = hpi_outstream_query_format(h_stream, &hpi_format); - if (!err && (hpi_to_alsa_formats[format] != -1)) + if (!err && (hpi_to_alsa_formats[format] != INVALID_FORMAT)) formats |= pcm_format_to_bits(hpi_to_alsa_formats[format]); } return formats; @@ -1205,7 +1207,7 @@ static u64 snd_card_asihpi_capture_formats(struct snd_card_asihpi *asihpi, format, sample_rate, 128000, 0); if (!err) err = hpi_instream_query_format(h_stream, &hpi_format); - if (!err && (hpi_to_alsa_formats[format] != -1)) + if (!err && (hpi_to_alsa_formats[format] != INVALID_FORMAT)) formats |= pcm_format_to_bits(hpi_to_alsa_formats[format]); } return formats; diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 8d5abfa4e24b..2864698436a5 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -635,7 +635,6 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao, { struct hpi_message hm; struct hpi_response hr; - u32 max_streams; HPI_DEBUG_LOG(VERBOSE, "init ADAPTER_GET_INFO\n"); memset(&hm, 0, sizeof(hm)); @@ -660,10 +659,6 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao, pao->type = hr.u.ax.info.adapter_type; pao->index = hr.u.ax.info.adapter_index; - max_streams = - hr.u.ax.info.num_outstreams + - hr.u.ax.info.num_instreams; - HPI_DEBUG_LOG(VERBOSE, "got adapter info type %x index %d serial %d\n", hr.u.ax.info.adapter_type, hr.u.ax.info.adapter_index, diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 7ae63d452bba..a1e4944dcfe8 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -207,10 +207,10 @@ struct atiixp; */ struct atiixp_dma_desc { - u32 addr; /* DMA buffer address */ + __le32 addr; /* DMA buffer address */ u16 status; /* status bits */ u16 size; /* size of the packet in dwords */ - u32 next; /* address of the next packet descriptor */ + __le32 next; /* address of the next packet descriptor */ }; /* diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index a586635664e0..dc1de860cedf 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -183,10 +183,10 @@ struct atiixp_modem; */ struct atiixp_dma_desc { - u32 addr; /* DMA buffer address */ + __le32 addr; /* DMA buffer address */ u16 status; /* status bits */ u16 size; /* size of the packet in dwords */ - u32 next; /* address of the next packet descriptor */ + __le32 next; /* address of the next packet descriptor */ }; /* diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index bcc648bf6478..e3e31f07d766 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -241,7 +241,7 @@ static int vortex_core_init(vortex_t * card); static int vortex_core_shutdown(vortex_t * card); static void vortex_enable_int(vortex_t * card); static irqreturn_t vortex_interrupt(int irq, void *dev_id); -static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v); +static int vortex_alsafmt_aspfmt(snd_pcm_format_t alsafmt, vortex_t *v); /* Connection stuff. */ static void vortex_connect_default(vortex_t * vortex, int en); diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 4083c8b01619..2e5b460a847c 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2770,7 +2770,7 @@ static int vortex_core_shutdown(vortex_t * vortex) /* Alsa support. */ -static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v) +static int vortex_alsafmt_aspfmt(snd_pcm_format_t alsafmt, vortex_t *v) { int fmt; diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index d8ade8771a32..ba971042f871 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -228,14 +228,14 @@ static int snd_bt87x_create_risc(struct snd_bt87x *chip, struct snd_pcm_substrea unsigned int periods, unsigned int period_bytes) { unsigned int i, offset; - u32 *risc; + __le32 *risc; if (chip->dma_risc.area == NULL) { if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), PAGE_ALIGN(MAX_RISC_SIZE), &chip->dma_risc) < 0) return -ENOMEM; } - risc = (u32 *)chip->dma_risc.area; + risc = (__le32 *)chip->dma_risc.area; offset = 0; *risc++ = cpu_to_le32(RISC_SYNC | RISC_SYNC_FM1); *risc++ = cpu_to_le32(0); diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index abb01ce66983..8d0a3d357345 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -73,13 +73,10 @@ static void cs46xx_dsp_proc_scb_info_read (struct snd_info_entry *entry, { struct proc_scb_info * scb_info = entry->private_data; struct dsp_scb_descriptor * scb = scb_info->scb_desc; - struct dsp_spos_instance * ins; struct snd_cs46xx *chip = scb_info->chip; int j,col; void __iomem *dst = chip->region.idx[1].remap_addr + DSP_PARAMETER_BYTE_OFFSET; - ins = chip->dsp_spos_instance; - mutex_lock(&chip->spos_mutex); snd_iprintf(buffer,"%04x %s:\n",scb->address,scb->scb_name); diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index de409cda50aa..4590086d9cd8 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -192,8 +192,6 @@ static void process_bm0_irq(struct cs5535audio *cs5535au) bm_stat = cs_readb(cs5535au, ACC_BM0_STATUS); spin_unlock(&cs5535au->reg_lock); if (bm_stat & EOP) { - struct cs5535audio_dma *dma; - dma = cs5535au->playback_substream->runtime->private_data; snd_pcm_period_elapsed(cs5535au->playback_substream); } else { dev_err(cs5535au->card->dev, @@ -208,11 +206,8 @@ static void process_bm1_irq(struct cs5535audio *cs5535au) spin_lock(&cs5535au->reg_lock); bm_stat = cs_readb(cs5535au, ACC_BM1_STATUS); spin_unlock(&cs5535au->reg_lock); - if (bm_stat & EOP) { - struct cs5535audio_dma *dma; - dma = cs5535au->capture_substream->runtime->private_data; + if (bm_stat & EOP) snd_pcm_period_elapsed(cs5535au->capture_substream); - } } static irqreturn_t snd_cs5535audio_interrupt(int irq, void *dev_id) diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h index f4fcdf93f3c8..d84620a0c26c 100644 --- a/sound/pci/cs5535audio/cs5535audio.h +++ b/sound/pci/cs5535audio/cs5535audio.h @@ -67,9 +67,9 @@ struct cs5535audio_dma_ops { }; struct cs5535audio_dma_desc { - u32 addr; - u16 size; - u16 ctlreserved; + __le32 addr; + __le16 size; + __le16 ctlreserved; }; struct cs5535audio_dma { diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index ee7065f6e162..326caec854e1 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -158,8 +158,8 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, lastdesc->addr = cpu_to_le32((u32) dma->desc_buf.addr); lastdesc->size = 0; lastdesc->ctlreserved = cpu_to_le16(PRD_JMP); - jmpprd_addr = cpu_to_le32(lastdesc->addr + - (sizeof(struct cs5535audio_dma_desc)*periods)); + jmpprd_addr = (u32)dma->desc_buf.addr + + sizeof(struct cs5535audio_dma_desc) * periods; dma->substream = substream; dma->period_bytes = period_bytes; diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 8e6eb9d7984b..6a051a1c3724 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1319,7 +1319,7 @@ static int hw_pll_init(struct hw *hw, unsigned int rsr) break; hw_write_20kx(hw, PLLCTL, pllctl); - mdelay(40); + msleep(40); } if (i >= 3) { dev_alert(hw->card->dev, "PLL initialization failed!!!\n"); @@ -1407,7 +1407,7 @@ static int hw_reset_dac(struct hw *hw) /* To be effective, need to reset the DAC twice. */ for (i = 0; i < 2; i++) { /* set gpio */ - mdelay(100); + msleep(100); gpioorg = (u16)hw_read_20kx(hw, GPIO); gpioorg &= 0xfffd; hw_write_20kx(hw, GPIO, gpioorg); @@ -2030,7 +2030,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) hw_write_20kx(hw, GIE, 0); /* Reset all SRC pending interrupts */ hw_write_20kx(hw, SRCIP, 0); - mdelay(30); + msleep(30); /* Detect the card ID and configure GPIO accordingly. */ switch (hw->model) { diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index b866d6b2c923..3c966fafc754 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -1316,12 +1316,12 @@ static int hw_pll_init(struct hw *hw, unsigned int rsr) set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 4 : 147 - 4); set_field(&pllctl, PLLCTL_RD, 48000 == rsr ? 1 - 1 : 10 - 1); hw_write_20kx(hw, PLL_CTL, pllctl); - mdelay(40); + msleep(40); pllctl = hw_read_20kx(hw, PLL_CTL); set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 2 : 147 - 2); hw_write_20kx(hw, PLL_CTL, pllctl); - mdelay(40); + msleep(40); for (i = 0; i < 1000; i++) { pllstat = hw_read_20kx(hw, PLL_STAT); @@ -1584,7 +1584,7 @@ static void hw_dac_stop(struct hw *hw) data = hw_read_20kx(hw, GPIO_DATA); data &= 0xFFFFFFFD; hw_write_20kx(hw, GPIO_DATA, data); - mdelay(10); + usleep_range(10000, 11000); } static void hw_dac_start(struct hw *hw) @@ -1593,7 +1593,7 @@ static void hw_dac_start(struct hw *hw) data = hw_read_20kx(hw, GPIO_DATA); data |= 0x2; hw_write_20kx(hw, GPIO_DATA, data); - mdelay(50); + msleep(50); } static void hw_dac_reset(struct hw *hw) @@ -1864,11 +1864,11 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info) hw_write_20kx(hw, GPIO_DATA, data); } - mdelay(10); + usleep_range(10000, 11000); /* Return the ADC to normal operation. */ data |= (0x1 << 15); hw_write_20kx(hw, GPIO_DATA, data); - mdelay(50); + msleep(50); /* I2C write to register offset 0x0B to set ADC LRCLK polarity */ /* invert bit, interface format to I2S, word length to 24-bit, */ diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c index db710d0a609f..4777d50fbbf8 100644 --- a/sound/pci/ctxfi/ctmixer.c +++ b/sound/pci/ctxfi/ctmixer.c @@ -938,17 +938,18 @@ static int ct_mixer_topology_build(struct ct_mixer *mixer) struct sum *sum; struct amixer *amix_d, *amix_s; enum CT_AMIXER_CTL i, j; + enum CT_SUM_CTL k; /* Build topology from destination to source */ /* Set up Master mixer */ - for (i = AMIXER_MASTER_F, j = SUM_IN_F; - i <= AMIXER_MASTER_S; i++, j++) { + for (i = AMIXER_MASTER_F, k = SUM_IN_F; + i <= AMIXER_MASTER_S; i++, k++) { amix_d = mixer->amixers[i*CHN_NUM]; - sum = mixer->sums[j*CHN_NUM]; + sum = mixer->sums[k*CHN_NUM]; amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL); amix_d = mixer->amixers[i*CHN_NUM+1]; - sum = mixer->sums[j*CHN_NUM+1]; + sum = mixer->sums[k*CHN_NUM+1]; amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL); } @@ -972,12 +973,12 @@ static int ct_mixer_topology_build(struct ct_mixer *mixer) amix_d->ops->setup(amix_d, &amix_s->rsc, INIT_VOL, NULL); /* Set up PCM-in mixer */ - for (i = AMIXER_PCM_F, j = SUM_IN_F; i <= AMIXER_PCM_S; i++, j++) { + for (i = AMIXER_PCM_F, k = SUM_IN_F; i <= AMIXER_PCM_S; i++, k++) { amix_d = mixer->amixers[i*CHN_NUM]; - sum = mixer->sums[j*CHN_NUM]; + sum = mixer->sums[k*CHN_NUM]; amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); amix_d = mixer->amixers[i*CHN_NUM+1]; - sum = mixer->sums[j*CHN_NUM+1]; + sum = mixer->sums[k*CHN_NUM+1]; amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); } diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 358ef7dcf410..907cf1a46712 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -713,6 +713,7 @@ static int pcm_prepare(struct snd_pcm_substream *substream) break; case SNDRV_PCM_FORMAT_S32_BE: format.data_are_bigendian = 1; + /* fall through */ case SNDRV_PCM_FORMAT_S32_LE: format.bits_per_sample = 32; break; @@ -764,6 +765,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) pipe->last_counter = 0; pipe->position = 0; *pipe->dma_counter = 0; + /* fall through */ case PIPE_STATE_PAUSED: pipe->state = PIPE_STATE_STARTED; break; diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 44b390a667d5..be4d0489394a 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -294,7 +294,7 @@ struct audiopipe { - volatile u32 *dma_counter; /* Commpage register that contains + volatile __le32 *dma_counter; /* Commpage register that contains * the current dma position * (lower 32 bits only) */ diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index 22c786b8a889..cc3c79387194 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -73,19 +73,21 @@ register. write_control_reg sends the new control register value to the DSP. */ static int write_control_reg(struct echoaudio *chip, u32 ctl, u32 frq, char force) { + __le32 ctl_reg, frq_reg; + if (wait_handshake(chip)) return -EIO; dev_dbg(chip->card->dev, "WriteControlReg: Setting 0x%x, 0x%x\n", ctl, frq); - ctl = cpu_to_le32(ctl); - frq = cpu_to_le32(frq); + ctl_reg = cpu_to_le32(ctl); + frq_reg = cpu_to_le32(frq); - if (ctl != chip->comm_page->control_register || - frq != chip->comm_page->e3g_frq_register || force) { - chip->comm_page->e3g_frq_register = frq; - chip->comm_page->control_register = ctl; + if (ctl_reg != chip->comm_page->control_register || + frq_reg != chip->comm_page->e3g_frq_register || force) { + chip->comm_page->e3g_frq_register = frq_reg; + chip->comm_page->control_register = ctl_reg; clear_handshake(chip); return send_vector(chip, DSP_VC_WRITE_CONTROL_REG); } diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 15aae2fad8e4..b181752b8481 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -679,7 +679,7 @@ static int restore_dsp_rettings(struct echoaudio *chip) /* Gina20/Darla20 only. Should be harmless for other cards. */ chip->comm_page->gd_clock_state = GD_CLOCK_UNDEF; chip->comm_page->gd_spdif_status = GD_SPDIF_STATUS_UNDEF; - chip->comm_page->handshake = 0xffffffff; + chip->comm_page->handshake = cpu_to_le32(0xffffffff); /* Restore output busses */ for (i = 0; i < num_busses_out(chip); i++) { @@ -989,7 +989,7 @@ static int init_dsp_comm_page(struct echoaudio *chip) /* Init the comm page */ chip->comm_page->comm_size = cpu_to_le32(sizeof(struct comm_page)); - chip->comm_page->handshake = 0xffffffff; + chip->comm_page->handshake = cpu_to_le32(0xffffffff); chip->comm_page->midi_out_free_count = cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE); chip->comm_page->sample_rate = cpu_to_le32(44100); @@ -1087,7 +1087,7 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe, /* The counter register is where the DSP writes the 32 bit DMA position for a pipe. The DSP is constantly updating this value as it moves data. The DMA counter is in units of bytes, not samples. */ - pipe->dma_counter = &chip->comm_page->position[pipe_index]; + pipe->dma_counter = (__le32 *)&chip->comm_page->position[pipe_index]; *pipe->dma_counter = 0; return pipe_index; } diff --git a/sound/pci/echoaudio/echoaudio_dsp.h b/sound/pci/echoaudio/echoaudio_dsp.h index cb7d75a0a503..aa9129519795 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.h +++ b/sound/pci/echoaudio/echoaudio_dsp.h @@ -627,8 +627,8 @@ sg_entry struct is read by the DSP, so all values must be little-endian. */ #define MAX_SGLIST_ENTRIES 512 struct sg_entry { - u32 addr; - u32 size; + __le32 addr; + __le32 size; }; @@ -643,18 +643,18 @@ struct sg_entry { ****************************************************************************/ struct comm_page { /* Base Length*/ - u32 comm_size; /* size of this object 0x000 4 */ - u32 flags; /* See Appendix A below 0x004 4 */ - u32 unused; /* Unused entry 0x008 4 */ - u32 sample_rate; /* Card sample rate in Hz 0x00c 4 */ - u32 handshake; /* DSP command handshake 0x010 4 */ - u32 cmd_start; /* Chs. to start mask 0x014 4 */ - u32 cmd_stop; /* Chs. to stop mask 0x018 4 */ - u32 cmd_reset; /* Chs. to reset mask 0x01c 4 */ - u16 audio_format[DSP_MAXPIPES]; /* Chs. audio format 0x020 32*2 */ + __le32 comm_size; /* size of this object 0x000 4 */ + __le32 flags; /* See Appendix A below 0x004 4 */ + __le32 unused; /* Unused entry 0x008 4 */ + __le32 sample_rate; /* Card sample rate in Hz 0x00c 4 */ + __le32 handshake; /* DSP command handshake 0x010 4 */ + __le32 cmd_start; /* Chs. to start mask 0x014 4 */ + __le32 cmd_stop; /* Chs. to stop mask 0x018 4 */ + __le32 cmd_reset; /* Chs. to reset mask 0x01c 4 */ + __le16 audio_format[DSP_MAXPIPES]; /* Chs. audio format 0x020 32*2 */ struct sg_entry sglist_addr[DSP_MAXPIPES]; /* Chs. Physical sglist addrs 0x060 32*8 */ - u32 position[DSP_MAXPIPES]; + __le32 position[DSP_MAXPIPES]; /* Positions for ea. ch. 0x160 32*4 */ s8 vu_meter[DSP_MAXPIPES]; /* VU meters 0x1e0 32*1 */ @@ -666,28 +666,28 @@ struct comm_page { /* Base Length*/ /* Input gain 0x230 16*1 */ s8 monitors[MONITOR_ARRAY_SIZE]; /* Monitor map 0x240 0x180 */ - u32 play_coeff[MAX_PLAY_TAPS]; + __le32 play_coeff[MAX_PLAY_TAPS]; /* Gina/Darla play filters - obsolete 0x3c0 168*4 */ - u32 rec_coeff[MAX_REC_TAPS]; + __le32 rec_coeff[MAX_REC_TAPS]; /* Gina/Darla record filters - obsolete 0x660 192*4 */ - u16 midi_input[MIDI_IN_BUFFER_SIZE]; + __le16 midi_input[MIDI_IN_BUFFER_SIZE]; /* MIDI input data transfer buffer 0x960 256*2 */ u8 gd_clock_state; /* Chg Gina/Darla clock state 0xb60 1 */ u8 gd_spdif_status; /* Chg. Gina/Darla S/PDIF state 0xb61 1 */ u8 gd_resampler_state; /* Should always be 3 0xb62 1 */ u8 filler2; /* 0xb63 1 */ - u32 nominal_level_mask; /* -10 level enable mask 0xb64 4 */ - u16 input_clock; /* Chg. Input clock state 0xb68 2 */ - u16 output_clock; /* Chg. Output clock state 0xb6a 2 */ - u32 status_clocks; /* Current Input clock state 0xb6c 4 */ - u32 ext_box_status; /* External box status 0xb70 4 */ - u32 cmd_add_buffer; /* Pipes to add (obsolete) 0xb74 4 */ - u32 midi_out_free_count; + __le32 nominal_level_mask; /* -10 level enable mask 0xb64 4 */ + __le16 input_clock; /* Chg. Input clock state 0xb68 2 */ + __le16 output_clock; /* Chg. Output clock state 0xb6a 2 */ + __le32 status_clocks; /* Current Input clock state 0xb6c 4 */ + __le32 ext_box_status; /* External box status 0xb70 4 */ + __le32 cmd_add_buffer; /* Pipes to add (obsolete) 0xb74 4 */ + __le32 midi_out_free_count; /* # of bytes free in MIDI output FIFO 0xb78 4 */ - u32 unused2; /* Cyclic pipes 0xb7c 4 */ - u32 control_register; + __le32 unused2; /* Cyclic pipes 0xb7c 4 */ + __le32 control_register; /* Mona, Gina24, Layla24, 3G ctrl reg 0xb80 4 */ - u32 e3g_frq_register; /* 3G frequency register 0xb84 4 */ + __le32 e3g_frq_register; /* 3G frequency register 0xb84 4 */ u8 filler[24]; /* filler 0xb88 24*1 */ s8 vmixer[VMIXER_ARRAY_SIZE]; /* Vmixer levels 0xba0 64*1 */ diff --git a/sound/pci/echoaudio/echoaudio_gml.c b/sound/pci/echoaudio/echoaudio_gml.c index 834b39e97db7..eea6fe530ab4 100644 --- a/sound/pci/echoaudio/echoaudio_gml.c +++ b/sound/pci/echoaudio/echoaudio_gml.c @@ -63,6 +63,8 @@ the control register. write_control_reg sends the new control register value to the DSP. */ static int write_control_reg(struct echoaudio *chip, u32 value, char force) { + __le32 reg_value; + /* Handle the digital input auto-mute */ if (chip->digital_in_automute) value |= GML_DIGITAL_IN_AUTO_MUTE; @@ -72,11 +74,11 @@ static int write_control_reg(struct echoaudio *chip, u32 value, char force) dev_dbg(chip->card->dev, "write_control_reg: 0x%x\n", value); /* Write the control register */ - value = cpu_to_le32(value); - if (value != chip->comm_page->control_register || force) { + reg_value = cpu_to_le32(value); + if (reg_value != chip->comm_page->control_register || force) { if (wait_handshake(chip)) return -EIO; - chip->comm_page->control_register = value; + chip->comm_page->control_register = reg_value; clear_handshake(chip); return send_vector(chip, DSP_VC_WRITE_CONTROL_REG); } diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 0e069aeab86d..c32eb7053715 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -70,11 +70,8 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, loopend = sampleend; /* be sure loop points start < end */ - if (sp->v.loopstart >= sp->v.loopend) { - int tmp = sp->v.loopstart; - sp->v.loopstart = sp->v.loopend; - sp->v.loopend = tmp; - } + if (sp->v.loopstart >= sp->v.loopend) + swap(sp->v.loopstart, sp->v.loopend); /* compute true data size to be loaded */ truesize = sp->v.size + BLANK_HEAD_SIZE; diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index de2ecbe95d6c..90713741c2dc 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -526,7 +526,7 @@ static int snd_emu10k1_gpr_poke(struct snd_emu10k1 *emu, if (!test_bit(gpr, icode->gpr_valid)) continue; if (in_kernel) - val = *(u32 *)&icode->gpr_map[gpr]; + val = *(__force u32 *)&icode->gpr_map[gpr]; else if (get_user(val, &icode->gpr_map[gpr])) return -EFAULT; snd_emu10k1_ptr_write(emu, emu->gpr_base + gpr, 0, val); @@ -560,8 +560,8 @@ static int snd_emu10k1_tram_poke(struct snd_emu10k1 *emu, if (!test_bit(tram, icode->tram_valid)) continue; if (in_kernel) { - val = *(u32 *)&icode->tram_data_map[tram]; - addr = *(u32 *)&icode->tram_addr_map[tram]; + val = *(__force u32 *)&icode->tram_data_map[tram]; + addr = *(__force u32 *)&icode->tram_addr_map[tram]; } else { if (get_user(val, &icode->tram_data_map[tram]) || get_user(addr, &icode->tram_addr_map[tram])) @@ -611,8 +611,8 @@ static int snd_emu10k1_code_poke(struct snd_emu10k1 *emu, if (!test_bit(pc / 2, icode->code_valid)) continue; if (in_kernel) { - lo = *(u32 *)&icode->code[pc + 0]; - hi = *(u32 *)&icode->code[pc + 1]; + lo = *(__force u32 *)&icode->code[pc + 0]; + hi = *(__force u32 *)&icode->code[pc + 1]; } else { if (get_user(lo, &icode->code[pc + 0]) || get_user(hi, &icode->code[pc + 1])) @@ -666,7 +666,7 @@ static unsigned int *copy_tlv(const unsigned int __user *_tlv, bool in_kernel) if (!_tlv) return NULL; if (in_kernel) - memcpy(data, (void *)_tlv, sizeof(data)); + memcpy(data, (__force void *)_tlv, sizeof(data)); else if (copy_from_user(data, _tlv, sizeof(data))) return NULL; if (data[1] >= MAX_TLV_SIZE) @@ -676,7 +676,7 @@ static unsigned int *copy_tlv(const unsigned int __user *_tlv, bool in_kernel) return NULL; memcpy(tlv, data, sizeof(data)); if (in_kernel) { - memcpy(tlv + 2, (void *)(_tlv + 2), data[1]); + memcpy(tlv + 2, (__force void *)(_tlv + 2), data[1]); } else if (copy_from_user(tlv + 2, _tlv + 2, data[1])) { kfree(tlv); return NULL; @@ -693,7 +693,7 @@ static int copy_gctl(struct snd_emu10k1 *emu, if (emu->support_tlv) { if (in_kernel) - memcpy(gctl, (void *)&_gctl[idx], sizeof(*gctl)); + memcpy(gctl, (__force void *)&_gctl[idx], sizeof(*gctl)); else if (copy_from_user(gctl, &_gctl[idx], sizeof(*gctl))) return -EFAULT; return 0; @@ -701,7 +701,7 @@ static int copy_gctl(struct snd_emu10k1 *emu, octl = (struct snd_emu10k1_fx8010_control_old_gpr __user *)_gctl; if (in_kernel) - memcpy(gctl, (void *)&octl[idx], sizeof(*octl)); + memcpy(gctl, (__force void *)&octl[idx], sizeof(*octl)); else if (copy_from_user(gctl, &octl[idx], sizeof(*octl))) return -EFAULT; gctl->tlv = NULL; @@ -735,7 +735,7 @@ static int snd_emu10k1_verify_controls(struct snd_emu10k1 *emu, for (i = 0, _id = icode->gpr_del_controls; i < icode->gpr_del_control_count; i++, _id++) { if (in_kernel) - id = *(struct snd_ctl_elem_id *)_id; + id = *(__force struct snd_ctl_elem_id *)_id; else if (copy_from_user(&id, _id, sizeof(id))) return -EFAULT; if (snd_emu10k1_look_for_ctl(emu, &id) == NULL) @@ -833,7 +833,7 @@ static int snd_emu10k1_add_controls(struct snd_emu10k1 *emu, knew.device = gctl->id.device; knew.subdevice = gctl->id.subdevice; knew.info = snd_emu10k1_gpr_ctl_info; - knew.tlv.p = copy_tlv(gctl->tlv, in_kernel); + knew.tlv.p = copy_tlv((__force const unsigned int __user *)gctl->tlv, in_kernel); if (knew.tlv.p) knew.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ; @@ -897,7 +897,7 @@ static int snd_emu10k1_del_controls(struct snd_emu10k1 *emu, for (i = 0, _id = icode->gpr_del_controls; i < icode->gpr_del_control_count; i++, _id++) { if (in_kernel) - id = *(struct snd_ctl_elem_id *)_id; + id = *(__force struct snd_ctl_elem_id *)_id; else if (copy_from_user(&id, _id, sizeof(id))) return -EFAULT; down_write(&card->controls_rwsem); diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 69f9b100bd24..9f2b6097f486 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -290,7 +290,7 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, struct snd_pcm_runtime *runtime = substream->runtime; unsigned int silent_page, tmp; int voice, stereo, w_16; - unsigned char attn, send_amount[8]; + unsigned char send_amount[8]; unsigned char send_routing[8]; unsigned long flags; unsigned int pitch_target; @@ -313,7 +313,6 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, /* volume parameters */ if (extra) { - attn = 0; memset(send_routing, 0, sizeof(send_routing)); send_routing[0] = 0; send_routing[1] = 1; @@ -779,7 +778,7 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); /* do we need this? */ snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[0]); - /* follow thru */ + /* fall through */ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE) @@ -929,7 +928,7 @@ static int snd_emu10k1_efx_playback_trigger(struct snd_pcm_substream *substream, } snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); - /* follow thru */ + /* fall through */ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: snd_emu10k1_playback_prepare_voice(emu, epcm->extra, 1, 1, NULL); diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 39f79a6b5283..727eb3da1fda 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -2392,7 +2392,7 @@ static int snd_audiopci_probe(struct pci_dev *pci, static int dev; struct snd_card *card; struct ensoniq *ensoniq; - int err, pcm_devs[2]; + int err; if (dev >= SNDRV_CARDS) return -ENODEV; @@ -2412,7 +2412,6 @@ static int snd_audiopci_probe(struct pci_dev *pci, } card->private_data = ensoniq; - pcm_devs[0] = 0; pcm_devs[1] = 1; #ifdef CHIP1370 if ((err = snd_ensoniq_1370_mixer(ensoniq)) < 0) { snd_card_free(card); diff --git a/sound/pci/hda/dell_wmi_helper.c b/sound/pci/hda/dell_wmi_helper.c index 1b48a8c19d28..8a7dbd1a7fbf 100644 --- a/sound/pci/hda/dell_wmi_helper.c +++ b/sound/pci/hda/dell_wmi_helper.c @@ -6,111 +6,18 @@ #if IS_ENABLED(CONFIG_DELL_LAPTOP) #include <linux/dell-led.h> -enum { - MICMUTE_LED_ON, - MICMUTE_LED_OFF, - MICMUTE_LED_FOLLOW_CAPTURE, - MICMUTE_LED_FOLLOW_MUTE, -}; - -static int dell_led_mode = MICMUTE_LED_FOLLOW_MUTE; -static int dell_capture; -static int dell_led_value; static int (*dell_micmute_led_set_func)(int); -static void (*dell_old_cap_hook)(struct hda_codec *, - struct snd_kcontrol *, - struct snd_ctl_elem_value *); - -static void call_micmute_led_update(void) -{ - int val; - - switch (dell_led_mode) { - case MICMUTE_LED_ON: - val = 1; - break; - case MICMUTE_LED_OFF: - val = 0; - break; - case MICMUTE_LED_FOLLOW_CAPTURE: - val = dell_capture; - break; - case MICMUTE_LED_FOLLOW_MUTE: - default: - val = !dell_capture; - break; - } - - if (val == dell_led_value) - return; - dell_led_value = val; - dell_micmute_led_set_func(dell_led_value); -} - -static void update_dell_wmi_micmute_led(struct hda_codec *codec, - struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - if (dell_old_cap_hook) - dell_old_cap_hook(codec, kcontrol, ucontrol); - - if (!ucontrol || !dell_micmute_led_set_func) - return; - if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { - /* TODO: How do I verify if it's a mono or stereo here? */ - dell_capture = (ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]); - call_micmute_led_update(); - } -} -static int dell_mic_mute_led_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) +static void dell_micmute_update(struct hda_codec *codec) { - static const char * const texts[] = { - "On", "Off", "Follow Capture", "Follow Mute", - }; - - return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); -} + struct hda_gen_spec *spec = codec->spec; -static int dell_mic_mute_led_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.enumerated.item[0] = dell_led_mode; - return 0; + dell_micmute_led_set_func(spec->micmute_led.led_value); } -static int dell_mic_mute_led_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - unsigned int mode; - - mode = ucontrol->value.enumerated.item[0]; - if (mode > MICMUTE_LED_FOLLOW_MUTE) - mode = MICMUTE_LED_FOLLOW_MUTE; - if (mode == dell_led_mode) - return 0; - dell_led_mode = mode; - call_micmute_led_update(); - return 1; -} - -static const struct snd_kcontrol_new dell_mic_mute_mode_ctls[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mic Mute-LED Mode", - .info = dell_mic_mute_led_mode_info, - .get = dell_mic_mute_led_mode_get, - .put = dell_mic_mute_led_mode_put, - }, - {} -}; - static void alc_fixup_dell_wmi(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct alc_spec *spec = codec->spec; bool removefunc = false; if (action == HDA_FIXUP_ACT_PROBE) { @@ -121,25 +28,14 @@ static void alc_fixup_dell_wmi(struct hda_codec *codec, return; } - removefunc = true; - if (dell_micmute_led_set_func(false) >= 0) { - dell_led_value = 0; - if (spec->gen.num_adc_nids > 1 && !spec->gen.dyn_adc_switch) - codec_dbg(codec, "Skipping micmute LED control due to several ADCs"); - else { - dell_old_cap_hook = spec->gen.cap_sync_hook; - spec->gen.cap_sync_hook = update_dell_wmi_micmute_led; - removefunc = false; - add_mixer(spec, dell_mic_mute_mode_ctls); - } - } - + removefunc = (dell_micmute_led_set_func(false) < 0) || + (snd_hda_gen_add_micmute_led(codec, + dell_micmute_update) <= 0); } if (dell_micmute_led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { symbol_put(dell_micmute_led_set); dell_micmute_led_set_func = NULL; - dell_old_cap_hook = NULL; } } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 20a171ac4bb2..0a5085537034 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -37,15 +37,8 @@ #include "hda_jack.h" #include <sound/hda_hwdep.h> -#ifdef CONFIG_PM -#define codec_in_pm(codec) atomic_read(&(codec)->core.in_pm) -#define hda_codec_is_power_on(codec) \ - (!pm_runtime_suspended(hda_codec_dev(codec))) -#else -#define codec_in_pm(codec) 0 -#define hda_codec_is_power_on(codec) 1 -#endif - +#define codec_in_pm(codec) snd_hdac_is_in_pm(&codec->core) +#define hda_codec_is_power_on(codec) snd_hdac_is_power_on(&codec->core) #define codec_has_epss(codec) \ ((codec)->core.power_caps & AC_PWRST_EPSS) #define codec_has_clkstop(codec) \ @@ -858,6 +851,39 @@ static void snd_hda_codec_dev_release(struct device *dev) kfree(codec); } +#define DEV_NAME_LEN 31 + +static int snd_hda_codec_device_init(struct hda_bus *bus, struct snd_card *card, + unsigned int codec_addr, struct hda_codec **codecp) +{ + char name[DEV_NAME_LEN]; + struct hda_codec *codec; + int err; + + dev_dbg(card->dev, "%s: entry\n", __func__); + + if (snd_BUG_ON(!bus)) + return -EINVAL; + if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) + return -EINVAL; + + codec = kzalloc(sizeof(*codec), GFP_KERNEL); + if (!codec) + return -ENOMEM; + + sprintf(name, "hdaudioC%dD%d", card->number, codec_addr); + err = snd_hdac_device_init(&codec->core, &bus->core, name, codec_addr); + if (err < 0) { + kfree(codec); + return err; + } + + codec->core.type = HDA_DEV_LEGACY; + *codecp = codec; + + return err; +} + /** * snd_hda_codec_new - create a HDA codec * @bus: the bus to assign @@ -869,7 +895,19 @@ static void snd_hda_codec_dev_release(struct device *dev) int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, unsigned int codec_addr, struct hda_codec **codecp) { - struct hda_codec *codec; + int ret; + + ret = snd_hda_codec_device_init(bus, card, codec_addr, codecp); + if (ret < 0) + return ret; + + return snd_hda_codec_device_new(bus, card, codec_addr, *codecp); +} +EXPORT_SYMBOL_GPL(snd_hda_codec_new); + +int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, + unsigned int codec_addr, struct hda_codec *codec) +{ char component[31]; hda_nid_t fg; int err; @@ -879,25 +917,14 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, .dev_free = snd_hda_codec_dev_free, }; + dev_dbg(card->dev, "%s: entry\n", __func__); + if (snd_BUG_ON(!bus)) return -EINVAL; if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) return -EINVAL; - codec = kzalloc(sizeof(*codec), GFP_KERNEL); - if (!codec) - return -ENOMEM; - - sprintf(component, "hdaudioC%dD%d", card->number, codec_addr); - err = snd_hdac_device_init(&codec->core, &bus->core, component, - codec_addr); - if (err < 0) { - kfree(codec); - return err; - } - codec->core.dev.release = snd_hda_codec_dev_release; - codec->core.type = HDA_DEV_LEGACY; codec->core.exec_verb = codec_exec_verb; codec->bus = bus; @@ -957,15 +984,13 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, if (err < 0) goto error; - if (codecp) - *codecp = codec; return 0; error: put_device(hda_codec_dev(codec)); return err; } -EXPORT_SYMBOL_GPL(snd_hda_codec_new); +EXPORT_SYMBOL_GPL(snd_hda_codec_device_new); /** * snd_hda_codec_update_widgets - Refresh widget caps and pin defaults @@ -2846,14 +2871,13 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec) { unsigned int state; - atomic_inc(&codec->core.in_pm); - + snd_hdac_enter_pm(&codec->core); if (codec->patch_ops.suspend) codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); state = hda_set_power_state(codec, AC_PWRST_D3); update_power_acct(codec, true); - atomic_dec(&codec->core.in_pm); + snd_hdac_leave_pm(&codec->core); return state; } @@ -2862,8 +2886,7 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec) */ static void hda_call_codec_resume(struct hda_codec *codec) { - atomic_inc(&codec->core.in_pm); - + snd_hdac_enter_pm(&codec->core); if (codec->core.regmap) regcache_mark_dirty(codec->core.regmap); @@ -2886,7 +2909,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) hda_jackpoll_work(&codec->jackpoll_work.work); else snd_hda_jack_report_sync(codec); - atomic_dec(&codec->core.in_pm); + snd_hdac_leave_pm(&codec->core); } static int hda_codec_runtime_suspend(struct device *dev) @@ -2992,6 +3015,7 @@ int snd_hda_codec_build_controls(struct hda_codec *codec) sync_power_up_states(codec); return 0; } +EXPORT_SYMBOL_GPL(snd_hda_codec_build_controls); /* * PCM stuff @@ -3197,6 +3221,7 @@ int snd_hda_codec_parse_pcms(struct hda_codec *codec) return 0; } +EXPORT_SYMBOL_GPL(snd_hda_codec_parse_pcms); /* assign all PCMs of the given codec */ int snd_hda_codec_build_pcms(struct hda_codec *codec) @@ -3252,8 +3277,8 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, for (; knew->name; knew++) { struct snd_kcontrol *kctl; int addr = 0, idx = 0; - if (knew->iface == -1) /* skip this codec private value */ - continue; + if (knew->iface == (__force snd_ctl_elem_iface_t)-1) + continue; /* skip this codec private value */ for (;;) { kctl = snd_ctl_new1(knew, codec); if (!kctl) @@ -3843,7 +3868,7 @@ EXPORT_SYMBOL_GPL(snd_hda_correct_pin_ctl); * This function is a helper to set a pin ctl value more safely. * It corrects the pin ctl value via snd_hda_correct_pin_ctl(), stores the * value in pin target array via snd_hda_codec_set_pin_target(), then - * actually writes the value via either snd_hda_codec_update_cache() or + * actually writes the value via either snd_hda_codec_write_cache() or * snd_hda_codec_write() depending on @cached flag. */ int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, @@ -3852,7 +3877,7 @@ int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, val = snd_hda_correct_pin_ctl(codec, pin, val); snd_hda_codec_set_pin_target(codec, pin, val); if (cached) - return snd_hda_codec_update_cache(codec, pin, 0, + return snd_hda_codec_write_cache(codec, pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val); else return snd_hda_codec_write(codec, pin, 0, diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index a8b1b31f161c..0d98bb9068b1 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -84,6 +84,7 @@ struct hda_bus { */ typedef int (*hda_codec_patch_t)(struct hda_codec *); +#define HDA_CODEC_ID_SKIP_PROBE 0x00000001 #define HDA_CODEC_ID_GENERIC_HDMI 0x00000101 #define HDA_CODEC_ID_GENERIC 0x00000201 @@ -308,6 +309,8 @@ struct hda_codec { */ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, unsigned int codec_addr, struct hda_codec **codecp); +int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, + unsigned int codec_addr, struct hda_codec *codec); int snd_hda_codec_configure(struct hda_codec *codec); int snd_hda_codec_update_widgets(struct hda_codec *codec); @@ -382,9 +385,6 @@ snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, return snd_hdac_regmap_write(&codec->core, nid, verb, parm); } -#define snd_hda_codec_update_cache(codec, nid, flags, verb, parm) \ - snd_hda_codec_write_cache(codec, nid, flags, verb, parm) - /* the struct for codec->pin_configs */ struct hda_pincfg { hda_nid_t nid; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index db773e219aaa..579984ecdec3 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -209,7 +209,7 @@ static void parse_user_hints(struct hda_codec *codec) */ #define update_pin_ctl(codec, pin, val) \ - snd_hda_codec_update_cache(codec, pin, 0, \ + snd_hda_codec_write_cache(codec, pin, 0, \ AC_VERB_SET_PIN_WIDGET_CONTROL, val) /* restore the pinctl based on the cached value */ @@ -898,7 +898,7 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path, hda_nid_t nid = path->path[i]; if (enable && path->multi[i]) - snd_hda_codec_update_cache(codec, nid, 0, + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, path->idx[i]); if (has_amp_in(codec, path, i)) @@ -930,7 +930,7 @@ static void set_pin_eapd(struct hda_codec *codec, hda_nid_t pin, bool enable) return; if (codec->inv_eapd) enable = !enable; - snd_hda_codec_update_cache(codec, pin, 0, + snd_hda_codec_write_cache(codec, pin, 0, AC_VERB_SET_EAPD_BTLENABLE, enable ? 0x02 : 0x00); } @@ -3900,6 +3900,142 @@ static int parse_mic_boost(struct hda_codec *codec) } /* + * mic mute LED hook helpers + */ +enum { + MICMUTE_LED_ON, + MICMUTE_LED_OFF, + MICMUTE_LED_FOLLOW_CAPTURE, + MICMUTE_LED_FOLLOW_MUTE, +}; + +static void call_micmute_led_update(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + unsigned int val; + + switch (spec->micmute_led.led_mode) { + case MICMUTE_LED_ON: + val = 1; + break; + case MICMUTE_LED_OFF: + val = 0; + break; + case MICMUTE_LED_FOLLOW_CAPTURE: + val = !!spec->micmute_led.capture; + break; + case MICMUTE_LED_FOLLOW_MUTE: + default: + val = !spec->micmute_led.capture; + break; + } + + if (val == spec->micmute_led.led_value) + return; + spec->micmute_led.led_value = val; + if (spec->micmute_led.update) + spec->micmute_led.update(codec); +} + +static void update_micmute_led(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_gen_spec *spec = codec->spec; + unsigned int mask; + + if (spec->micmute_led.old_hook) + spec->micmute_led.old_hook(codec, kcontrol, ucontrol); + + if (!ucontrol) + return; + mask = 1U << snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + if (!strcmp("Capture Switch", ucontrol->id.name)) { + /* TODO: How do I verify if it's a mono or stereo here? */ + if (ucontrol->value.integer.value[0] || + ucontrol->value.integer.value[1]) + spec->micmute_led.capture |= mask; + else + spec->micmute_led.capture &= ~mask; + call_micmute_led_update(codec); + } +} + +static int micmute_led_mode_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[] = { + "On", "Off", "Follow Capture", "Follow Mute", + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static int micmute_led_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_gen_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->micmute_led.led_mode; + return 0; +} + +static int micmute_led_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_gen_spec *spec = codec->spec; + unsigned int mode; + + mode = ucontrol->value.enumerated.item[0]; + if (mode > MICMUTE_LED_FOLLOW_MUTE) + mode = MICMUTE_LED_FOLLOW_MUTE; + if (mode == spec->micmute_led.led_mode) + return 0; + spec->micmute_led.led_mode = mode; + call_micmute_led_update(codec); + return 1; +} + +static const struct snd_kcontrol_new micmute_led_mode_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Mute-LED Mode", + .info = micmute_led_mode_info, + .get = micmute_led_mode_get, + .put = micmute_led_mode_put, +}; + +/** + * snd_hda_gen_add_micmute_led - helper for setting up mic mute LED hook + * @codec: the HDA codec + * @hook: the callback for updating LED + * + * Called from the codec drivers for offering the mic mute LED controls. + * When established, it sets up cap_sync_hook and triggers the callback at + * each time when the capture mixer switch changes. The callback is supposed + * to update the LED accordingly. + * + * Returns 0 if the hook is established or a negative error code. + */ +int snd_hda_gen_add_micmute_led(struct hda_codec *codec, + void (*hook)(struct hda_codec *)) +{ + struct hda_gen_spec *spec = codec->spec; + + spec->micmute_led.led_mode = MICMUTE_LED_FOLLOW_MUTE; + spec->micmute_led.capture = 0; + spec->micmute_led.led_value = 0; + spec->micmute_led.old_hook = spec->cap_sync_hook; + spec->micmute_led.update = hook; + spec->cap_sync_hook = update_micmute_led; + if (!snd_hda_gen_add_kctl(spec, NULL, &micmute_led_mode_ctl)) + return -ENOMEM; + return 0; +} +EXPORT_SYMBOL_GPL(snd_hda_gen_add_micmute_led); + +/* * parse digital I/Os and set up NIDs in BIOS auto-parse mode */ static void parse_digital(struct hda_codec *codec) @@ -5837,7 +5973,7 @@ static void clear_unsol_on_unused_pins(struct hda_codec *codec) hda_nid_t nid = pin->nid; if (is_jack_detectable(codec, nid) && !snd_hda_jack_tbl_get(codec, nid)) - snd_hda_codec_update_cache(codec, nid, 0, + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, 0); } } diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 61772317de46..10123664fa61 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -86,6 +86,16 @@ struct badness_table { extern const struct badness_table hda_main_out_badness; extern const struct badness_table hda_extra_out_badness; +struct hda_micmute_hook { + unsigned int led_mode; + unsigned int capture; + unsigned int led_value; + void (*update)(struct hda_codec *codec); + void (*old_hook)(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +}; + struct hda_gen_spec { char stream_name_analog[32]; /* analog PCM stream */ const struct hda_pcm_stream *stream_analog_playback; @@ -276,6 +286,9 @@ struct hda_gen_spec { struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); + /* mic mute LED hook; called via cap_sync_hook */ + struct hda_micmute_hook micmute_led; + /* PCM hooks */ void (*pcm_playback_hook)(struct hda_pcm_stream *hinfo, struct hda_codec *codec, @@ -342,4 +355,7 @@ unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on); int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin); +int snd_hda_gen_add_micmute_led(struct hda_codec *codec, + void (*hook)(struct hda_codec *)); + #endif /* __SOUND_HDA_GENERIC_H */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1ae1850b3bfd..1b2ce304152a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1319,15 +1319,16 @@ static const struct vga_switcheroo_client_ops azx_vs_ops = { static int register_vga_switcheroo(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct pci_dev *p; int err; if (!hda->use_vga_switcheroo) return 0; - /* FIXME: currently only handling DIS controller - * is there any machine with two switchable HDMI audio controllers? - */ - err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, - VGA_SWITCHEROO_DIS); + + p = get_bound_vga(chip->pci); + err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, p); + pci_dev_put(p); + if (err < 0) return err; hda->vga_switcheroo_registered = 1; @@ -1429,7 +1430,7 @@ static struct pci_dev *get_bound_vga(struct pci_dev *pci) p = pci_get_domain_bus_and_slot(pci_domain_nr(pci->bus), pci->bus->number, 0); if (p) { - if ((p->class >> 8) == PCI_CLASS_DISPLAY_VGA) + if ((p->class >> 16) == PCI_BASE_CLASS_DISPLAY) return p; pci_dev_put(p); } @@ -2207,7 +2208,7 @@ out_free: */ static struct snd_pci_quirk power_save_blacklist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ - SND_PCI_QUIRK(0x1849, 0x0c0c, "Asrock B85M-ITX", 0), + SND_PCI_QUIRK(0x1849, 0xc892, "Asrock B85M-ITX", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1849, 0x7662, "Asrock H81M-HDS", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ @@ -2535,7 +2536,8 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), - .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | + AZX_DCAPS_PM_RUNTIME }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x0002), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 757857313426..fd476fb40e1b 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -148,7 +148,7 @@ static void ad_vmaster_eapd_hook(void *private_data, int enabled) return; if (codec->inv_eapd) enabled = !enabled; - snd_hda_codec_update_cache(codec, spec->eapd_nid, 0, + snd_hda_codec_write_cache(codec, spec->eapd_nid, 0, AC_VERB_SET_EAPD_BTLENABLE, enabled ? 0x02 : 0x00); } @@ -991,7 +991,7 @@ static void ad1884_vmaster_hp_gpio_hook(void *private_data, int enabled) if (spec->eapd_nid) ad_vmaster_eapd_hook(private_data, enabled); - snd_hda_codec_update_cache(codec, 0x01, 0, + snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, enabled ? 0x00 : 0x02); } diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 321e95c409c1..0166a3d7cd55 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -897,7 +897,7 @@ struct ca0132_spec { const struct hda_verb *base_init_verbs; const struct hda_verb *base_exit_verbs; const struct hda_verb *chip_init_verbs; - const struct hda_verb *sbz_init_verbs; + const struct hda_verb *desktop_init_verbs; struct hda_verb *spec_init_verbs; struct auto_pin_cfg autocfg; @@ -965,9 +965,11 @@ struct ca0132_spec { long cur_ctl_vals[TUNING_CTLS_COUNT]; #endif /* - * Sound Blaster Z PCI region 2 iomem, used for input and output - * switching, and other unknown commands. + * The Recon3D, Sound Blaster Z, Sound Blaster ZxR, and Sound Blaster + * AE-5 all use PCI region 2 to toggle GPIO and other currently unknown + * things. */ + bool use_pci_mmio; void __iomem *mem_base; /* @@ -994,6 +996,7 @@ enum { QUIRK_ALIENWARE_M17XR4, QUIRK_SBZ, QUIRK_R3DI, + QUIRK_R3D, }; static const struct hda_pintbl alienware_pincfgs[] = { @@ -1025,6 +1028,21 @@ static const struct hda_pintbl sbz_pincfgs[] = { {} }; +/* Recon3D pin configs taken from Windows Driver */ +static const struct hda_pintbl r3d_pincfgs[] = { + { 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */ + { 0x0c, 0x014510f0 }, /* SPDIF Out 1 */ + { 0x0d, 0x014510f0 }, /* Digital Out */ + { 0x0e, 0x01c520f0 }, /* SPDIF In */ + { 0x0f, 0x0221401f }, /* Port A -- BackPanel HP */ + { 0x10, 0x01016011 }, /* Port D -- Center/LFE or FP Hp */ + { 0x11, 0x01011014 }, /* Port B -- LineMicIn2 / Rear L/R */ + { 0x12, 0x02a090f0 }, /* Port C -- LineIn1 */ + { 0x13, 0x908700f0 }, /* What U Hear In*/ + { 0x18, 0x50d000f0 }, /* N/A */ + {} +}; + /* Recon3D integrated pin configs taken from Windows Driver */ static const struct hda_pintbl r3di_pincfgs[] = { { 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */ @@ -1050,6 +1068,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI), + SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D), {} }; @@ -3073,6 +3092,24 @@ static bool dspload_wait_loaded(struct hda_codec *codec) */ /* + * For cards with PCI-E region2 (Sound Blaster Z/ZxR, Recon3D, and AE-5) + * the mmio address 0x320 is used to set GPIO pins. The format for the data + * The first eight bits are just the number of the pin. So far, I've only seen + * this number go to 7. + */ +static void ca0132_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin, + bool enable) +{ + struct ca0132_spec *spec = codec->spec; + unsigned short gpio_data; + + gpio_data = gpio_pin & 0xF; + gpio_data |= ((enable << 8) & 0x100); + + writew(gpio_data, spec->mem_base + 0x320); +} + +/* * Sets up the GPIO pins so that they are discoverable. If this isn't done, * the card shows as having no GPIO pins. */ @@ -3947,15 +3984,19 @@ static int ca0132_alt_select_out(struct hda_codec *codec) /*speaker out config*/ switch (spec->quirk) { case QUIRK_SBZ: - writew(0x0007, spec->mem_base + 0x320); - writew(0x0104, spec->mem_base + 0x320); - writew(0x0101, spec->mem_base + 0x320); + ca0132_mmio_gpio_set(codec, 7, false); + ca0132_mmio_gpio_set(codec, 4, true); + ca0132_mmio_gpio_set(codec, 1, true); chipio_set_control_param(codec, 0x0D, 0x18); break; case QUIRK_R3DI: chipio_set_control_param(codec, 0x0D, 0x24); r3di_gpio_out_set(codec, R3DI_LINE_OUT); break; + case QUIRK_R3D: + chipio_set_control_param(codec, 0x0D, 0x24); + ca0132_mmio_gpio_set(codec, 1, true); + break; } /* disable headphone node */ @@ -3983,15 +4024,19 @@ static int ca0132_alt_select_out(struct hda_codec *codec) /* Headphone out config*/ switch (spec->quirk) { case QUIRK_SBZ: - writew(0x0107, spec->mem_base + 0x320); - writew(0x0104, spec->mem_base + 0x320); - writew(0x0001, spec->mem_base + 0x320); + ca0132_mmio_gpio_set(codec, 7, true); + ca0132_mmio_gpio_set(codec, 4, true); + ca0132_mmio_gpio_set(codec, 1, false); chipio_set_control_param(codec, 0x0D, 0x12); break; case QUIRK_R3DI: chipio_set_control_param(codec, 0x0D, 0x21); r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT); break; + case QUIRK_R3D: + chipio_set_control_param(codec, 0x0D, 0x21); + ca0132_mmio_gpio_set(codec, 0x1, false); + break; } snd_hda_codec_write(codec, spec->out_pins[0], 0, @@ -4025,15 +4070,19 @@ static int ca0132_alt_select_out(struct hda_codec *codec) /* Surround out config*/ switch (spec->quirk) { case QUIRK_SBZ: - writew(0x0007, spec->mem_base + 0x320); - writew(0x0104, spec->mem_base + 0x320); - writew(0x0101, spec->mem_base + 0x320); + ca0132_mmio_gpio_set(codec, 7, false); + ca0132_mmio_gpio_set(codec, 4, true); + ca0132_mmio_gpio_set(codec, 1, true); chipio_set_control_param(codec, 0x0D, 0x18); break; case QUIRK_R3DI: chipio_set_control_param(codec, 0x0D, 0x24); r3di_gpio_out_set(codec, R3DI_LINE_OUT); break; + case QUIRK_R3D: + ca0132_mmio_gpio_set(codec, 1, true); + chipio_set_control_param(codec, 0x0D, 0x24); + break; } /* enable line out node */ pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, @@ -4291,7 +4340,8 @@ static int ca0132_alt_select_in(struct hda_codec *codec) case REAR_MIC: switch (spec->quirk) { case QUIRK_SBZ: - writew(0x0000, spec->mem_base + 0x320); + case QUIRK_R3D: + ca0132_mmio_gpio_set(codec, 0, false); tmp = FLOAT_THREE; break; case QUIRK_R3DI: @@ -4323,7 +4373,8 @@ static int ca0132_alt_select_in(struct hda_codec *codec) ca0132_mic_boost_set(codec, 0); switch (spec->quirk) { case QUIRK_SBZ: - writew(0x0000, spec->mem_base + 0x320); + case QUIRK_R3D: + ca0132_mmio_gpio_set(codec, 0, false); break; case QUIRK_R3DI: r3di_gpio_mic_set(codec, R3DI_REAR_MIC); @@ -4349,8 +4400,9 @@ static int ca0132_alt_select_in(struct hda_codec *codec) case FRONT_MIC: switch (spec->quirk) { case QUIRK_SBZ: - writew(0x0100, spec->mem_base + 0x320); - writew(0x0005, spec->mem_base + 0x320); + case QUIRK_R3D: + ca0132_mmio_gpio_set(codec, 0, true); + ca0132_mmio_gpio_set(codec, 5, false); tmp = FLOAT_THREE; break; case QUIRK_R3DI: @@ -5516,8 +5568,7 @@ static int ca0132_alt_add_effect_slider(struct hda_codec *codec, hda_nid_t nid, sprintf(namestr, "FX: %s %s Volume", pfx, dirstr[dir]); - knew.tlv.c = 0; - knew.tlv.p = 0; + knew.tlv.c = NULL; switch (nid) { case XBASS_XOVER: @@ -5729,11 +5780,11 @@ static const struct snd_kcontrol_new ca0132_mixer[] = { }; /* - * SBZ specific control mixer. Removes auto-detect for mic, and adds surround - * controls. Also sets both the Front Playback and Capture Volume controls to - * alt so they set the DSP's decibel level. + * Desktop specific control mixer. Removes auto-detect for mic, and adds + * surround controls. Also sets both the Front Playback and Capture Volume + * controls to alt so they set the DSP's decibel level. */ -static const struct snd_kcontrol_new sbz_mixer[] = { +static const struct snd_kcontrol_new desktop_mixer[] = { CA0132_ALT_CODEC_VOL("Front Playback Volume", 0x02, HDA_OUTPUT), CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x04, 0, HDA_OUTPUT), @@ -5804,8 +5855,8 @@ static int ca0132_build_controls(struct hda_codec *codec) */ num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT; for (i = 0; i < num_fx; i++) { - /* SBZ breaks if Echo Cancellation is used */ - if (spec->quirk == QUIRK_SBZ) { + /* SBZ and R3D break if Echo Cancellation is used. */ + if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3D) { if (i == (ECHO_CANCELLATION - IN_EFFECT_START_NID + OUT_EFFECTS_COUNT)) continue; @@ -6187,10 +6238,10 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec) } /* - * Recon3Di r3di_setup_defaults sub functions. + * Recon3D r3d_setup_defaults sub functions. */ -static void r3di_dsp_scp_startup(struct hda_codec *codec) +static void r3d_dsp_scp_startup(struct hda_codec *codec) { unsigned int tmp; @@ -6211,7 +6262,7 @@ static void r3di_dsp_scp_startup(struct hda_codec *codec) } -static void r3di_dsp_initial_mic_setup(struct hda_codec *codec) +static void r3d_dsp_initial_mic_setup(struct hda_codec *codec) { unsigned int tmp; @@ -6421,10 +6472,10 @@ static void ca0132_setup_defaults(struct hda_codec *codec) } /* - * Setup default parameters for Recon3Di DSP. + * Setup default parameters for Recon3D/Recon3Di DSP. */ -static void r3di_setup_defaults(struct hda_codec *codec) +static void r3d_setup_defaults(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; unsigned int tmp; @@ -6434,9 +6485,9 @@ static void r3di_setup_defaults(struct hda_codec *codec) if (spec->dsp_state != DSP_DOWNLOADED) return; - r3di_dsp_scp_startup(codec); + r3d_dsp_scp_startup(codec); - r3di_dsp_initial_mic_setup(codec); + r3d_dsp_initial_mic_setup(codec); /*remove DSP headroom*/ tmp = FLOAT_ZERO; @@ -6450,7 +6501,8 @@ static void r3di_setup_defaults(struct hda_codec *codec) /* Set speaker source? */ dspio_set_uint_param(codec, 0x32, 0x00, tmp); - r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED); + if (spec->quirk == QUIRK_R3DI) + r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED); /* Setup effect defaults */ num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1; @@ -6462,7 +6514,6 @@ static void r3di_setup_defaults(struct hda_codec *codec) ca0132_effects[idx].def_vals[i]); } } - } /* @@ -6727,7 +6778,12 @@ static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb) static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb) { - ca0132_select_mic(codec); + struct ca0132_spec *spec = codec->spec; + + if (spec->use_alt_functions) + ca0132_alt_select_in(codec); + else + ca0132_select_mic(codec); } static void ca0132_init_unsol(struct hda_codec *codec) @@ -6798,8 +6854,8 @@ static struct hda_verb ca0132_init_verbs0[] = { {} }; -/* Extra init verbs for SBZ */ -static struct hda_verb sbz_init_verbs[] = { +/* Extra init verbs for desktop cards. */ +static struct hda_verb ca0132_init_verbs1[] = { {0x15, 0x70D, 0x20}, {0x15, 0x70E, 0x19}, {0x15, 0x707, 0x00}, @@ -6891,16 +6947,12 @@ static void sbz_region2_exit(struct hda_codec *codec) writeb(0x0, spec->mem_base + 0x100); for (i = 0; i < 8; i++) writeb(0xb3, spec->mem_base + 0x304); - /* - * I believe these are GPIO, with the right most hex digit being the - * gpio pin, and the second digit being on or off. We see this more in - * the input/output select functions. - */ - writew(0x0000, spec->mem_base + 0x320); - writew(0x0001, spec->mem_base + 0x320); - writew(0x0104, spec->mem_base + 0x320); - writew(0x0005, spec->mem_base + 0x320); - writew(0x0007, spec->mem_base + 0x320); + + ca0132_mmio_gpio_set(codec, 0, false); + ca0132_mmio_gpio_set(codec, 1, false); + ca0132_mmio_gpio_set(codec, 4, true); + ca0132_mmio_gpio_set(codec, 5, false); + ca0132_mmio_gpio_set(codec, 7, false); } static void sbz_set_pin_ctl_default(struct hda_codec *codec) @@ -6916,7 +6968,7 @@ static void sbz_set_pin_ctl_default(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00); } -static void sbz_clear_unsolicited(struct hda_codec *codec) +static void ca0132_clear_unsolicited(struct hda_codec *codec) { hda_nid_t pins[7] = {0x0B, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13}; unsigned int i; @@ -6969,21 +7021,22 @@ static void sbz_exit_chip(struct hda_codec *codec) chipio_set_control_param(codec, 0x0D, 0x24); - sbz_clear_unsolicited(codec); + ca0132_clear_unsolicited(codec); sbz_set_pin_ctl_default(codec); snd_hda_codec_write(codec, 0x0B, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x00); - if (dspload_is_loaded(codec)) - dsp_reset(codec); - - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE, 0x00); - sbz_region2_exit(codec); } +static void r3d_exit_chip(struct hda_codec *codec) +{ + ca0132_clear_unsolicited(codec); + snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); + snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5b); +} + static void ca0132_exit_chip(struct hda_codec *codec) { /* put any chip cleanup stuffs here. */ @@ -7098,9 +7151,27 @@ static void sbz_pre_dsp_setup(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, 0x44); } -/* - * Extra commands that don't really fit anywhere else. - */ +static void r3d_pre_dsp_setup(struct hda_codec *codec) +{ + + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfc); + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfd); + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfe); + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xff); + + chipio_write(codec, 0x18b0a4, 0x000000c2); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x1E); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0x1C); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_DATA_WRITE, 0x5B); + + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x44); +} + static void r3di_pre_dsp_setup(struct hda_codec *codec) { chipio_write(codec, 0x18b0a4, 0x000000c2); @@ -7125,13 +7196,12 @@ static void r3di_pre_dsp_setup(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, 0x04); } - /* * These are sent before the DSP is downloaded. Not sure * what they do, or if they're necessary. Could possibly * be removed. Figure they're better to leave in. */ -static void sbz_region2_startup(struct hda_codec *codec) +static void ca0132_mmio_init(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; @@ -7171,7 +7241,7 @@ static void ca0132_alt_init(struct hda_codec *codec) ca0132_gpio_init(codec); sbz_pre_dsp_setup(codec); snd_hda_sequence_write(codec, spec->chip_init_verbs); - snd_hda_sequence_write(codec, spec->sbz_init_verbs); + snd_hda_sequence_write(codec, spec->desktop_init_verbs); break; case QUIRK_R3DI: codec_dbg(codec, "R3DI alt_init"); @@ -7182,6 +7252,11 @@ static void ca0132_alt_init(struct hda_codec *codec) snd_hda_sequence_write(codec, spec->chip_init_verbs); snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x6FF, 0xC4); break; + case QUIRK_R3D: + r3d_pre_dsp_setup(codec); + snd_hda_sequence_write(codec, spec->chip_init_verbs); + snd_hda_sequence_write(codec, spec->desktop_init_verbs); + break; } } @@ -7218,8 +7293,8 @@ static int ca0132_init(struct hda_codec *codec) spec->dsp_state = DSP_DOWNLOAD_INIT; spec->curr_chip_addx = INVALID_CHIP_ADDRESS; - if (spec->quirk == QUIRK_SBZ) - sbz_region2_startup(codec); + if (spec->use_pci_mmio) + ca0132_mmio_init(codec); snd_hda_power_up_pm(codec); @@ -7236,14 +7311,13 @@ static int ca0132_init(struct hda_codec *codec) ca0132_refresh_widget_caps(codec); - if (spec->quirk == QUIRK_SBZ) - writew(0x0107, spec->mem_base + 0x320); - switch (spec->quirk) { case QUIRK_R3DI: - r3di_setup_defaults(codec); + case QUIRK_R3D: + r3d_setup_defaults(codec); break; case QUIRK_SBZ: + sbz_setup_defaults(codec); break; default: ca0132_setup_defaults(codec); @@ -7274,20 +7348,12 @@ static int ca0132_init(struct hda_codec *codec) ca0132_gpio_setup(codec); snd_hda_sequence_write(codec, spec->spec_init_verbs); - switch (spec->quirk) { - case QUIRK_SBZ: - sbz_setup_defaults(codec); - ca0132_alt_select_out(codec); - ca0132_alt_select_in(codec); - break; - case QUIRK_R3DI: + if (spec->use_alt_functions) { ca0132_alt_select_out(codec); ca0132_alt_select_in(codec); - break; - default: + } else { ca0132_select_out(codec); ca0132_select_mic(codec); - break; } snd_hda_jack_report_sync(codec); @@ -7316,16 +7382,17 @@ static void ca0132_free(struct hda_codec *codec) case QUIRK_SBZ: sbz_exit_chip(codec); break; + case QUIRK_R3D: + r3d_exit_chip(codec); + break; case QUIRK_R3DI: r3di_gpio_shutdown(codec); - snd_hda_sequence_write(codec, spec->base_exit_verbs); - ca0132_exit_chip(codec); - break; - default: - snd_hda_sequence_write(codec, spec->base_exit_verbs); - ca0132_exit_chip(codec); break; } + + snd_hda_sequence_write(codec, spec->base_exit_verbs); + ca0132_exit_chip(codec); + snd_hda_power_down(codec); if (spec->mem_base) iounmap(spec->mem_base); @@ -7386,8 +7453,15 @@ static void ca0132_config(struct hda_codec *codec) spec->unsol_tag_amic1 = 0x11; break; case QUIRK_SBZ: - codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__); - snd_hda_apply_pincfgs(codec, sbz_pincfgs); + case QUIRK_R3D: + if (spec->quirk == QUIRK_SBZ) { + codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__); + snd_hda_apply_pincfgs(codec, sbz_pincfgs); + } + if (spec->quirk == QUIRK_R3D) { + codec_dbg(codec, "%s: QUIRK_R3D applied.\n", __func__); + snd_hda_apply_pincfgs(codec, r3d_pincfgs); + } spec->num_outputs = 2; spec->out_pins[0] = 0x0B; /* Line out */ @@ -7473,8 +7547,8 @@ static int ca0132_prepare_verbs(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; spec->chip_init_verbs = ca0132_init_verbs0; - if (spec->quirk == QUIRK_SBZ) - spec->sbz_init_verbs = sbz_init_verbs; + if (spec->quirk == QUIRK_SBZ || spec->quirk == QUIRK_R3D) + spec->desktop_init_verbs = ca0132_init_verbs1; spec->spec_init_verbs = kcalloc(NUM_SPEC_VERBS, sizeof(struct hda_verb), GFP_KERNEL); @@ -7530,25 +7604,19 @@ static int patch_ca0132(struct hda_codec *codec) else spec->quirk = QUIRK_NONE; - /* Setup BAR Region 2 for Sound Blaster Z */ - if (spec->quirk == QUIRK_SBZ) { - spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); - if (spec->mem_base == NULL) { - codec_warn(codec, "pci_iomap failed!"); - codec_info(codec, "perhaps this is not an SBZ?"); - spec->quirk = QUIRK_NONE; - } - } - spec->dsp_state = DSP_DOWNLOAD_INIT; spec->num_mixers = 1; /* Set which mixers each quirk uses. */ switch (spec->quirk) { case QUIRK_SBZ: - spec->mixers[0] = sbz_mixer; + spec->mixers[0] = desktop_mixer; snd_hda_codec_set_name(codec, "Sound Blaster Z"); break; + case QUIRK_R3D: + spec->mixers[0] = desktop_mixer; + snd_hda_codec_set_name(codec, "Recon3D"); + break; case QUIRK_R3DI: spec->mixers[0] = r3di_mixer; snd_hda_codec_set_name(codec, "Recon3Di"); @@ -7558,19 +7626,34 @@ static int patch_ca0132(struct hda_codec *codec) break; } - /* Setup whether or not to use alt functions/controls */ + /* Setup whether or not to use alt functions/controls/pci_mmio */ switch (spec->quirk) { case QUIRK_SBZ: + case QUIRK_R3D: + spec->use_alt_controls = true; + spec->use_alt_functions = true; + spec->use_pci_mmio = true; + break; case QUIRK_R3DI: spec->use_alt_controls = true; spec->use_alt_functions = true; + spec->use_pci_mmio = false; break; default: spec->use_alt_controls = false; spec->use_alt_functions = false; + spec->use_pci_mmio = false; break; } + if (spec->use_pci_mmio) { + spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); + if (spec->mem_base == NULL) { + codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE."); + spec->quirk = QUIRK_NONE; + } + } + spec->base_init_verbs = ca0132_base_init_verbs; spec->base_exit_verbs = ca0132_base_exit_verbs; diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index d6e079f4ec09..a7f91be45194 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1096,25 +1096,6 @@ static int cs421x_init(struct hda_codec *codec) return 0; } -static int cs421x_build_controls(struct hda_codec *codec) -{ - struct cs_spec *spec = codec->spec; - int err; - - err = snd_hda_gen_build_controls(codec); - if (err < 0) - return err; - - if (spec->gen.autocfg.speaker_outs && - spec->vendor_nid == CS4210_VENDOR_NID) { - err = snd_hda_ctl_add(codec, 0, - snd_ctl_new1(&cs421x_speaker_boost_ctl, codec)); - if (err < 0) - return err; - } - return 0; -} - static void fix_volume_caps(struct hda_codec *codec, hda_nid_t dac) { unsigned int caps; @@ -1144,6 +1125,14 @@ static int cs421x_parse_auto_config(struct hda_codec *codec) return err; parse_cs421x_digital(codec); + + if (spec->gen.autocfg.speaker_outs && + spec->vendor_nid == CS4210_VENDOR_NID) { + if (!snd_hda_gen_add_kctl(&spec->gen, NULL, + &cs421x_speaker_boost_ctl)) + return -ENOMEM; + } + return 0; } @@ -1175,7 +1164,7 @@ static int cs421x_suspend(struct hda_codec *codec) #endif static const struct hda_codec_ops cs421x_patch_ops = { - .build_controls = cs421x_build_controls, + .build_controls = snd_hda_gen_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = cs421x_init, .free = cs_free, diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f641c20095f7..cfd4e4f97f8f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -37,8 +37,6 @@ struct conexant_spec { struct hda_gen_spec gen; - unsigned int beep_amp; - /* extra EAPD pins */ unsigned int num_eapds; hda_nid_t eapds[4]; @@ -62,65 +60,48 @@ struct conexant_spec { #ifdef CONFIG_SND_HDA_INPUT_BEEP -static inline void set_beep_amp(struct conexant_spec *spec, hda_nid_t nid, - int idx, int dir) -{ - spec->gen.beep_nid = nid; - spec->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir); -} -/* additional beep mixers; the actual parameters are overwritten at build */ +/* additional beep mixers; private_value will be overwritten */ static const struct snd_kcontrol_new cxt_beep_mixer[] = { HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0, 1, 0, HDA_OUTPUT), HDA_CODEC_MUTE_BEEP_MONO("Beep Playback Switch", 0, 1, 0, HDA_OUTPUT), - { } /* end */ }; -/* create beep controls if needed */ -static int add_beep_ctls(struct hda_codec *codec) +static int set_beep_amp(struct conexant_spec *spec, hda_nid_t nid, + int idx, int dir) { - struct conexant_spec *spec = codec->spec; - int err; + struct snd_kcontrol_new *knew; + unsigned int beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir); + int i; - if (spec->beep_amp) { - const struct snd_kcontrol_new *knew; - for (knew = cxt_beep_mixer; knew->name; knew++) { - struct snd_kcontrol *kctl; - kctl = snd_ctl_new1(knew, codec); - if (!kctl) - return -ENOMEM; - kctl->private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, 0, kctl); - if (err < 0) - return err; - } + spec->gen.beep_nid = nid; + for (i = 0; i < ARRAY_SIZE(cxt_beep_mixer); i++) { + knew = snd_hda_gen_add_kctl(&spec->gen, NULL, + &cxt_beep_mixer[i]); + if (!knew) + return -ENOMEM; + knew->private_value = beep_amp; } return 0; } -#else -#define set_beep_amp(spec, nid, idx, dir) /* NOP */ -#define add_beep_ctls(codec) 0 -#endif - -/* - * Automatic parser for CX20641 & co - */ -#ifdef CONFIG_SND_HDA_INPUT_BEEP -static void cx_auto_parse_beep(struct hda_codec *codec) +static int cx_auto_parse_beep(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; hda_nid_t nid; for_each_hda_codec_node(nid, codec) - if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_BEEP) { - set_beep_amp(spec, nid, 0, HDA_OUTPUT); - break; - } + if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_BEEP) + return set_beep_amp(spec, nid, 0, HDA_OUTPUT); + return 0; } #else -#define cx_auto_parse_beep(codec) +#define cx_auto_parse_beep(codec) 0 #endif +/* + * Automatic parser for CX20641 & co + */ + /* parse EAPDs */ static void cx_auto_parse_eapd(struct hda_codec *codec) { @@ -179,21 +160,6 @@ static void cx_auto_vmaster_hook_mute_led(void *private_data, int enabled) enabled ? 0x00 : 0x02); } -static int cx_auto_build_controls(struct hda_codec *codec) -{ - int err; - - err = snd_hda_gen_build_controls(codec); - if (err < 0) - return err; - - err = add_beep_ctls(codec); - if (err < 0) - return err; - - return 0; -} - static int cx_auto_init(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -211,6 +177,7 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; switch (codec->core.vendor_id) { + case 0x14f12008: /* CX8200 */ case 0x14f150f2: /* CX20722 */ case 0x14f150f4: /* CX20724 */ break; @@ -218,13 +185,14 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) return; } - /* Turn the CX20722 codec into D3 to avoid spurious noises + /* Turn the problematic codec into D3 to avoid spurious noises from the internal speaker during (and after) reboot */ cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false); snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + msleep(10); } static void cx_auto_free(struct hda_codec *codec) @@ -234,7 +202,7 @@ static void cx_auto_free(struct hda_codec *codec) } static const struct hda_codec_ops cx_auto_patch_ops = { - .build_controls = cx_auto_build_controls, + .build_controls = snd_hda_gen_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = cx_auto_init, .reboot_notify = cx_auto_reboot_notify, @@ -343,6 +311,7 @@ static void cxt_fixup_headphone_mic(struct hda_codec *codec, snd_hdac_regmap_add_vendor_verb(&codec->core, 0x410); break; case HDA_FIXUP_ACT_PROBE: + WARN_ON(spec->gen.cap_sync_hook); spec->gen.cap_sync_hook = cxt_update_headset_mode_hook; spec->gen.automute_hook = cxt_update_headset_mode; break; @@ -374,7 +343,7 @@ static void cxt_fixup_headset_mic(struct hda_codec *codec, * control. */ #define update_mic_pin(codec, nid, val) \ - snd_hda_codec_update_cache(codec, nid, 0, \ + snd_hda_codec_write_cache(codec, nid, 0, \ AC_VERB_SET_PIN_WIDGET_CONTROL, val) static const struct hda_input_mux olpc_xo_dc_bias = { @@ -695,16 +664,12 @@ static void cxt_fixup_gpio_mute_hook(void *private_data, int enabled) } /* turn on/off mic-mute LED via GPIO per capture hook */ -static void cxt_fixup_gpio_mic_mute_hook(struct hda_codec *codec, - struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void cxt_gpio_micmute_update(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - if (ucontrol) - cxt_update_gpio_led(codec, spec->gpio_mic_led_mask, - ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]); + cxt_update_gpio_led(codec, spec->gpio_mic_led_mask, + spec->gen.micmute_led.led_value); } @@ -721,11 +686,11 @@ static void cxt_fixup_mute_led_gpio(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = cxt_fixup_gpio_mute_hook; - spec->gen.cap_sync_hook = cxt_fixup_gpio_mic_mute_hook; spec->gpio_led = 0; spec->mute_led_polarity = 0; spec->gpio_mute_led_mask = 0x01; spec->gpio_mic_led_mask = 0x02; + snd_hda_gen_add_micmute_led(codec, cxt_gpio_micmute_update); } snd_hda_add_verbs(codec, gpio_init); if (spec->gpio_led) @@ -1037,7 +1002,6 @@ static int patch_conexant_auto(struct hda_codec *codec) codec->spec = spec; codec->patch_ops = cx_auto_patch_ops; - cx_auto_parse_beep(codec); cx_auto_parse_eapd(codec); spec->gen.own_eapd_ctl = 1; if (spec->dynamic_eapd) @@ -1097,6 +1061,10 @@ static int patch_conexant_auto(struct hda_codec *codec) if (err < 0) goto error; + err = cx_auto_parse_beep(codec); + if (err < 0) + goto error; + /* Some laptops with Conexant chips show stalls in S3 resume, * which falls into the single-cmd mode. * Better to make reset, then. diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8a49415aebac..cb587dce67a9 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -177,13 +177,13 @@ struct hdmi_spec { /* i915/powerwell (Haswell+/Valleyview+) specific */ bool use_acomp_notifier; /* use i915 eld_notify callback for hotplug */ - struct i915_audio_component_audio_ops i915_audio_ops; + struct drm_audio_component_audio_ops drm_audio_ops; struct hdac_chmap chmap; hda_nid_t vendor_nid; }; -#ifdef CONFIG_SND_HDA_I915 +#ifdef CONFIG_SND_HDA_COMPONENT static inline bool codec_has_acomp(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -339,13 +339,13 @@ static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, if (!per_pin) { /* no pin is bound to the pcm */ uinfo->count = 0; - mutex_unlock(&spec->pcm_lock); - return 0; + goto unlock; } eld = &per_pin->sink_eld; uinfo->count = eld->eld_valid ? eld->eld_size : 0; - mutex_unlock(&spec->pcm_lock); + unlock: + mutex_unlock(&spec->pcm_lock); return 0; } @@ -357,6 +357,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; int pcm_idx; + int err = 0; pcm_idx = kcontrol->private_value; mutex_lock(&spec->pcm_lock); @@ -365,16 +366,15 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, /* no pin is bound to the pcm */ memset(ucontrol->value.bytes.data, 0, ARRAY_SIZE(ucontrol->value.bytes.data)); - mutex_unlock(&spec->pcm_lock); - return 0; + goto unlock; } - eld = &per_pin->sink_eld; + eld = &per_pin->sink_eld; if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data) || eld->eld_size > ELD_MAX_SIZE) { - mutex_unlock(&spec->pcm_lock); snd_BUG(); - return -EINVAL; + err = -EINVAL; + goto unlock; } memset(ucontrol->value.bytes.data, 0, @@ -382,9 +382,10 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, if (eld->eld_valid) memcpy(ucontrol->value.bytes.data, eld->eld_buffer, eld->eld_size); - mutex_unlock(&spec->pcm_lock); - return 0; + unlock: + mutex_unlock(&spec->pcm_lock); + return err; } static const struct snd_kcontrol_new eld_bytes_ctl = { @@ -1209,8 +1210,8 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, pin_idx = hinfo_to_pin_index(codec, hinfo); if (!spec->dyn_pcm_assign) { if (snd_BUG_ON(pin_idx < 0)) { - mutex_unlock(&spec->pcm_lock); - return -EINVAL; + err = -EINVAL; + goto unlock; } } else { /* no pin is assigned to the PCM @@ -1218,16 +1219,13 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, */ if (pin_idx < 0) { err = hdmi_pcm_open_no_pin(hinfo, codec, substream); - mutex_unlock(&spec->pcm_lock); - return err; + goto unlock; } } err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx); - if (err < 0) { - mutex_unlock(&spec->pcm_lock); - return err; - } + if (err < 0) + goto unlock; per_cvt = get_cvt(spec, cvt_idx); /* Claim converter */ @@ -1264,12 +1262,11 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, per_cvt->assigned = 0; hinfo->nid = 0; snd_hda_spdif_ctls_unassign(codec, pcm_idx); - mutex_unlock(&spec->pcm_lock); - return -ENODEV; + err = -ENODEV; + goto unlock; } } - mutex_unlock(&spec->pcm_lock); /* Store the updated parameters */ runtime->hw.channels_min = hinfo->channels_min; runtime->hw.channels_max = hinfo->channels_max; @@ -1278,7 +1275,9 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); - return 0; + unlock: + mutex_unlock(&spec->pcm_lock); + return err; } /* @@ -1867,7 +1866,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_runtime *runtime = substream->runtime; bool non_pcm; int pinctl; - int err; + int err = 0; mutex_lock(&spec->pcm_lock); pin_idx = hinfo_to_pin_index(codec, hinfo); @@ -1879,13 +1878,12 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, pin_cvt_fixup(codec, NULL, cvt_nid); snd_hda_codec_setup_stream(codec, cvt_nid, stream_tag, 0, format); - mutex_unlock(&spec->pcm_lock); - return 0; + goto unlock; } if (snd_BUG_ON(pin_idx < 0)) { - mutex_unlock(&spec->pcm_lock); - return -EINVAL; + err = -EINVAL; + goto unlock; } per_pin = get_pin(spec, pin_idx); pin_nid = per_pin->pin_nid; @@ -1924,6 +1922,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, /* snd_hda_set_dev_select() has been called before */ err = spec->ops.setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); + unlock: mutex_unlock(&spec->pcm_lock); return err; } @@ -1945,6 +1944,7 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, struct hdmi_spec_per_cvt *per_cvt; struct hdmi_spec_per_pin *per_pin; int pinctl; + int err = 0; if (hinfo->nid) { pcm_idx = hinfo_to_pcm_index(codec, hinfo); @@ -1963,14 +1963,12 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, snd_hda_spdif_ctls_unassign(codec, pcm_idx); clear_bit(pcm_idx, &spec->pcm_in_use); pin_idx = hinfo_to_pin_index(codec, hinfo); - if (spec->dyn_pcm_assign && pin_idx < 0) { - mutex_unlock(&spec->pcm_lock); - return 0; - } + if (spec->dyn_pcm_assign && pin_idx < 0) + goto unlock; if (snd_BUG_ON(pin_idx < 0)) { - mutex_unlock(&spec->pcm_lock); - return -EINVAL; + err = -EINVAL; + goto unlock; } per_pin = get_pin(spec, pin_idx); @@ -1989,10 +1987,11 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, per_pin->setup = false; per_pin->channels = 0; mutex_unlock(&per_pin->lock); + unlock: mutex_unlock(&spec->pcm_lock); } - return 0; + return err; } static const struct hda_pcm_ops generic_ops = { @@ -2288,7 +2287,7 @@ static void generic_hdmi_free(struct hda_codec *codec) int pin_idx, pcm_idx; if (codec_has_acomp(codec)) - snd_hdac_i915_register_notifier(NULL); + snd_hdac_acomp_register_notifier(&codec->bus->core, NULL); for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); @@ -2471,6 +2470,38 @@ static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, snd_hda_codec_set_power_to_all(codec, fg, power_state); } +/* There is a fixed mapping between audio pin node and display port. + * on SNB, IVY, HSW, BSW, SKL, BXT, KBL: + * Pin Widget 5 - PORT B (port = 1 in i915 driver) + * Pin Widget 6 - PORT C (port = 2 in i915 driver) + * Pin Widget 7 - PORT D (port = 3 in i915 driver) + * + * on VLV, ILK: + * Pin Widget 4 - PORT B (port = 1 in i915 driver) + * Pin Widget 5 - PORT C (port = 2 in i915 driver) + * Pin Widget 6 - PORT D (port = 3 in i915 driver) + */ +static int intel_base_nid(struct hda_codec *codec) +{ + switch (codec->core.vendor_id) { + case 0x80860054: /* ILK */ + case 0x80862804: /* ILK */ + case 0x80862882: /* VLV */ + return 4; + default: + return 5; + } +} + +static int intel_pin2port(void *audio_ptr, int pin_nid) +{ + int base_nid = intel_base_nid(audio_ptr); + + if (WARN_ON(pin_nid < base_nid || pin_nid >= base_nid + 3)) + return -1; + return pin_nid - base_nid + 1; /* intel port is 1-based */ +} + static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) { struct hda_codec *codec = audio_ptr; @@ -2481,16 +2512,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) if (port < 1 || port > 3) return; - switch (codec->core.vendor_id) { - case 0x80860054: /* ILK */ - case 0x80862804: /* ILK */ - case 0x80862882: /* VLV */ - pin_nid = port + 0x03; - break; - default: - pin_nid = port + 0x04; - break; - } + pin_nid = port + intel_base_nid(codec) - 1; /* intel port is 1-based */ /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume @@ -2498,7 +2520,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) return; /* ditto during suspend/resume process itself */ - if (atomic_read(&(codec)->core.in_pm)) + if (snd_hdac_is_in_pm(&codec->core)) return; snd_hdac_i915_set_bclk(&codec->bus->core); @@ -2511,14 +2533,16 @@ static void register_i915_notifier(struct hda_codec *codec) struct hdmi_spec *spec = codec->spec; spec->use_acomp_notifier = true; - spec->i915_audio_ops.audio_ptr = codec; + spec->drm_audio_ops.audio_ptr = codec; /* intel_audio_codec_enable() or intel_audio_codec_disable() * will call pin_eld_notify with using audio_ptr pointer * We need make sure audio_ptr is really setup */ wmb(); - spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; - snd_hdac_i915_register_notifier(&spec->i915_audio_ops); + spec->drm_audio_ops.pin2port = intel_pin2port; + spec->drm_audio_ops.pin_eld_notify = intel_pin_eld_notify; + snd_hdac_acomp_register_notifier(&codec->bus->core, + &spec->drm_audio_ops); } /* setup_stream ops override for HSW+ */ @@ -2551,6 +2575,8 @@ static int alloc_intel_hdmi(struct hda_codec *codec) /* requires i915 binding */ if (!codec->bus->core.audio_component) { codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n"); + /* set probe_id here to prevent generic fallback binding */ + codec->probe_id = HDA_CODEC_ID_SKIP_PROBE; return -ENODEV; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f6af3e1c2b93..b20974ef1e13 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -43,11 +43,9 @@ /* extra amp-initialization sequence types */ enum { + ALC_INIT_UNDEFINED, ALC_INIT_NONE, ALC_INIT_DEFAULT, - ALC_INIT_GPIO1, - ALC_INIT_GPIO2, - ALC_INIT_GPIO3, }; enum { @@ -85,19 +83,20 @@ struct alc_spec { struct hda_gen_spec gen; /* must be at head */ /* codec parameterization */ - const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ - unsigned int num_mixers; - unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ - struct alc_customize_define cdefine; unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */ + /* GPIO bits */ + unsigned int gpio_mask; + unsigned int gpio_dir; + unsigned int gpio_data; + bool gpio_write_delay; /* add a delay before writing gpio_data */ + /* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */ int mute_led_polarity; hda_nid_t mute_led_nid; hda_nid_t cap_mute_led_nid; - unsigned int gpio_led; /* used for alc269_fixup_hp_gpio_led() */ unsigned int gpio_mute_led_mask; unsigned int gpio_mic_led_mask; @@ -205,41 +204,87 @@ static void alc_process_coef_fw(struct hda_codec *codec, } /* - * Append the given mixer and verb elements for the later use - * The mixer array is referred in build_controls(), and init_verbs are - * called in init(). + * GPIO setup tables, used in initialization */ -static void add_mixer(struct alc_spec *spec, const struct snd_kcontrol_new *mix) + +/* Enable GPIO mask and set output */ +static void alc_setup_gpio(struct hda_codec *codec, unsigned int mask) +{ + struct alc_spec *spec = codec->spec; + + spec->gpio_mask |= mask; + spec->gpio_dir |= mask; + spec->gpio_data |= mask; +} + +static void alc_write_gpio_data(struct hda_codec *codec) { - if (snd_BUG_ON(spec->num_mixers >= ARRAY_SIZE(spec->mixers))) + struct alc_spec *spec = codec->spec; + + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_data); +} + +static void alc_update_gpio_data(struct hda_codec *codec, unsigned int mask, + bool on) +{ + struct alc_spec *spec = codec->spec; + unsigned int oldval = spec->gpio_data; + + if (on) + spec->gpio_data |= mask; + else + spec->gpio_data &= ~mask; + if (oldval != spec->gpio_data) + alc_write_gpio_data(codec); +} + +static void alc_write_gpio(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->gpio_mask) return; - spec->mixers[spec->num_mixers++] = mix; + + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_MASK, spec->gpio_mask); + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DIRECTION, spec->gpio_dir); + if (spec->gpio_write_delay) + msleep(1); + alc_write_gpio_data(codec); } -/* - * GPIO setup tables, used in initialization - */ -/* Enable GPIO mask and set output */ -static const struct hda_verb alc_gpio1_init_verbs[] = { - {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, - { } -}; +static void alc_fixup_gpio(struct hda_codec *codec, int action, + unsigned int mask) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + alc_setup_gpio(codec, mask); +} -static const struct hda_verb alc_gpio2_init_verbs[] = { - {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, - { } -}; +static void alc_fixup_gpio1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_gpio(codec, action, 0x01); +} -static const struct hda_verb alc_gpio3_init_verbs[] = { - {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x03}, - { } -}; +static void alc_fixup_gpio2(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_gpio(codec, action, 0x02); +} + +static void alc_fixup_gpio3(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_gpio(codec, action, 0x03); +} + +static void alc_fixup_gpio4(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_gpio(codec, action, 0x04); +} /* * Fix hardware PLL issue @@ -447,16 +492,8 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) { alc_fill_eapd_coef(codec); alc_auto_setup_eapd(codec, true); + alc_write_gpio(codec); switch (type) { - case ALC_INIT_GPIO1: - snd_hda_sequence_write(codec, alc_gpio1_init_verbs); - break; - case ALC_INIT_GPIO2: - snd_hda_sequence_write(codec, alc_gpio2_init_verbs); - break; - case ALC_INIT_GPIO3: - snd_hda_sequence_write(codec, alc_gpio3_init_verbs); - break; case ALC_INIT_DEFAULT: switch (codec->core.vendor_id) { case 0x10ec0260: @@ -656,20 +693,22 @@ do_sku: * 7~6 : Reserved */ tmp = (ass & 0x38) >> 3; /* external Amp control */ - switch (tmp) { - case 1: - spec->init_amp = ALC_INIT_GPIO1; - break; - case 3: - spec->init_amp = ALC_INIT_GPIO2; - break; - case 7: - spec->init_amp = ALC_INIT_GPIO3; - break; - case 5: - default: - spec->init_amp = ALC_INIT_DEFAULT; - break; + if (spec->init_amp == ALC_INIT_UNDEFINED) { + switch (tmp) { + case 1: + alc_setup_gpio(codec, 0x01); + break; + case 3: + alc_setup_gpio(codec, 0x02); + break; + case 7: + alc_setup_gpio(codec, 0x03); + break; + case 5: + default: + spec->init_amp = ALC_INIT_DEFAULT; + break; + } } /* is laptop or Desktop and enable the function "Mute internal speaker @@ -722,47 +761,14 @@ static void alc_fixup_inv_dmic(struct hda_codec *codec, } -#ifdef CONFIG_SND_HDA_INPUT_BEEP -/* additional beep mixers; the actual parameters are overwritten at build */ -static const struct snd_kcontrol_new alc_beep_mixer[] = { - HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT), - HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_INPUT), - { } /* end */ -}; -#endif - static int alc_build_controls(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; - int i, err; + int err; err = snd_hda_gen_build_controls(codec); if (err < 0) return err; - for (i = 0; i < spec->num_mixers; i++) { - err = snd_hda_add_new_ctls(codec, spec->mixers[i]); - if (err < 0) - return err; - } - -#ifdef CONFIG_SND_HDA_INPUT_BEEP - /* create beep controls if needed */ - if (spec->beep_amp) { - const struct snd_kcontrol_new *knew; - for (knew = alc_beep_mixer; knew->name; knew++) { - struct snd_kcontrol *kctl; - kctl = snd_ctl_new1(knew, codec); - if (!kctl) - return -ENOMEM; - kctl->private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, 0, kctl); - if (err < 0) - return err; - } - } -#endif - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_BUILD); return 0; } @@ -973,8 +979,30 @@ static int alc_codec_rename_from_preset(struct hda_codec *codec) * Digital-beep handlers */ #ifdef CONFIG_SND_HDA_INPUT_BEEP -#define set_beep_amp(spec, nid, idx, dir) \ - ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir)) + +/* additional beep mixers; private_value will be overwritten */ +static const struct snd_kcontrol_new alc_beep_mixer[] = { + HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT), + HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_INPUT), +}; + +/* set up and create beep controls */ +static int set_beep_amp(struct alc_spec *spec, hda_nid_t nid, + int idx, int dir) +{ + struct snd_kcontrol_new *knew; + unsigned int beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir); + int i; + + for (i = 0; i < ARRAY_SIZE(alc_beep_mixer); i++) { + knew = snd_hda_gen_add_kctl(&spec->gen, NULL, + &alc_beep_mixer[i]); + if (!knew) + return -ENOMEM; + knew->private_value = beep_amp; + } + return 0; +} static const struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x103c, "ASUS", 1), @@ -999,7 +1027,7 @@ static inline int has_cdefine_beep(struct hda_codec *codec) return spec->cdefine.enable_pcbeep; } #else -#define set_beep_amp(spec, nid, idx, dir) /* NOP */ +#define set_beep_amp(spec, nid, idx, dir) 0 #define has_cdefine_beep(codec) 0 #endif @@ -1104,12 +1132,12 @@ static void alc880_fixup_vol_knob(struct hda_codec *codec, static const struct hda_fixup alc880_fixups[] = { [ALC880_FIXUP_GPIO1] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = alc_gpio1_init_verbs, + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio1, }, [ALC880_FIXUP_GPIO2] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = alc_gpio2_init_verbs, + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio2, }, [ALC880_FIXUP_MEDION_RIM] = { .type = HDA_FIXUP_VERBS, @@ -1501,8 +1529,11 @@ static int patch_alc880(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog) - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (!spec->gen.no_analog) { + err = set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (err < 0) + goto error; + } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -1544,8 +1575,8 @@ enum { static void alc260_gpio1_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - spec->gen.hp_jack_present); + + alc_update_gpio_data(codec, 0x01, spec->gen.hp_jack_present); } static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, @@ -1562,7 +1593,7 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, spec->gen.autocfg.hp_pins[0] = 0x0f; /* copy it for automute */ snd_hda_jack_detect_enable_callback(codec, 0x0f, snd_hda_gen_hp_automute); - snd_hda_add_verbs(codec, alc_gpio1_init_verbs); + alc_setup_gpio(codec, 0x01); } } @@ -1589,8 +1620,6 @@ static void alc260_fixup_kn1(struct hda_codec *codec, switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: snd_hda_apply_pincfgs(codec, pincfgs); - break; - case HDA_FIXUP_ACT_PROBE: spec->init_amp = ALC_INIT_NONE; break; } @@ -1600,7 +1629,7 @@ static void alc260_fixup_fsc_s7020(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PROBE) + if (action == HDA_FIXUP_ACT_PRE_PROBE) spec->init_amp = ALC_INIT_NONE; } @@ -1638,8 +1667,8 @@ static const struct hda_fixup alc260_fixups[] = { }, }, [ALC260_FIXUP_GPIO1] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = alc_gpio1_init_verbs, + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio1, }, [ALC260_FIXUP_GPIO1_TOGGLE] = { .type = HDA_FIXUP_FUNC, @@ -1751,8 +1780,11 @@ static int patch_alc260(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog) - set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); + if (!spec->gen.no_analog) { + err = set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); + if (err < 0) + goto error; + } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -1824,47 +1856,14 @@ static void alc889_fixup_coef(struct hda_codec *codec, alc_update_coef_idx(codec, 7, 0, 0x2030); } -/* toggle speaker-output according to the hp-jack state */ -static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted) -{ - unsigned int gpiostate, gpiomask, gpiodir; - - gpiostate = snd_hda_codec_read(codec, codec->core.afg, 0, - AC_VERB_GET_GPIO_DATA, 0); - - if (!muted) - gpiostate |= (1 << pin); - else - gpiostate &= ~(1 << pin); - - gpiomask = snd_hda_codec_read(codec, codec->core.afg, 0, - AC_VERB_GET_GPIO_MASK, 0); - gpiomask |= (1 << pin); - - gpiodir = snd_hda_codec_read(codec, codec->core.afg, 0, - AC_VERB_GET_GPIO_DIRECTION, 0); - gpiodir |= (1 << pin); - - - snd_hda_codec_write(codec, codec->core.afg, 0, - AC_VERB_SET_GPIO_MASK, gpiomask); - snd_hda_codec_write(codec, codec->core.afg, 0, - AC_VERB_SET_GPIO_DIRECTION, gpiodir); - - msleep(1); - - snd_hda_codec_write(codec, codec->core.afg, 0, - AC_VERB_SET_GPIO_DATA, gpiostate); -} - /* set up GPIO at initialization */ static void alc885_fixup_macpro_gpio(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - if (action != HDA_FIXUP_ACT_INIT) - return; - alc882_gpio_mute(codec, 0, 0); - alc882_gpio_mute(codec, 1, 0); + struct alc_spec *spec = codec->spec; + + spec->gpio_write_delay = true; + alc_fixup_gpio3(codec, fix, action); } /* Fix the connection of some pins for ALC889: @@ -2143,20 +2142,20 @@ static const struct hda_fixup alc882_fixups[] = { } }, [ALC882_FIXUP_GPIO1] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = alc_gpio1_init_verbs, + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio1, }, [ALC882_FIXUP_GPIO2] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = alc_gpio2_init_verbs, + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio2, }, [ALC882_FIXUP_GPIO3] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = alc_gpio3_init_verbs, + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio3, }, [ALC882_FIXUP_ASUS_W2JC] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = alc_gpio1_init_verbs, + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio1, .chained = true, .chain_id = ALC882_FIXUP_EAPD, }, @@ -2376,12 +2375,37 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { }; static const struct hda_model_fixup alc882_fixup_models[] = { + {.id = ALC882_FIXUP_ABIT_AW9D_MAX, .name = "abit-aw9d"}, + {.id = ALC882_FIXUP_LENOVO_Y530, .name = "lenovo-y530"}, + {.id = ALC882_FIXUP_ACER_ASPIRE_7736, .name = "acer-aspire-7736"}, + {.id = ALC882_FIXUP_ASUS_W90V, .name = "asus-w90v"}, + {.id = ALC889_FIXUP_CD, .name = "cd"}, + {.id = ALC889_FIXUP_FRONT_HP_NO_PRESENCE, .name = "no-front-hp"}, + {.id = ALC889_FIXUP_VAIO_TT, .name = "vaio-tt"}, + {.id = ALC888_FIXUP_EEE1601, .name = "eee1601"}, + {.id = ALC882_FIXUP_EAPD, .name = "alc882-eapd"}, + {.id = ALC883_FIXUP_EAPD, .name = "alc883-eapd"}, + {.id = ALC882_FIXUP_GPIO1, .name = "gpio1"}, + {.id = ALC882_FIXUP_GPIO2, .name = "gpio2"}, + {.id = ALC882_FIXUP_GPIO3, .name = "gpio3"}, + {.id = ALC889_FIXUP_COEF, .name = "alc889-coef"}, + {.id = ALC882_FIXUP_ASUS_W2JC, .name = "asus-w2jc"}, {.id = ALC882_FIXUP_ACER_ASPIRE_4930G, .name = "acer-aspire-4930g"}, {.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"}, {.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"}, + {.id = ALC885_FIXUP_MACPRO_GPIO, .name = "macpro-gpio"}, + {.id = ALC889_FIXUP_DAC_ROUTE, .name = "dac-route"}, + {.id = ALC889_FIXUP_MBP_VREF, .name = "mbp-vref"}, + {.id = ALC889_FIXUP_IMAC91_VREF, .name = "imac91-vref"}, + {.id = ALC889_FIXUP_MBA11_VREF, .name = "mba11-vref"}, + {.id = ALC889_FIXUP_MBA21_VREF, .name = "mba21-vref"}, + {.id = ALC889_FIXUP_MP11_VREF, .name = "mp11-vref"}, + {.id = ALC889_FIXUP_MP41_VREF, .name = "mp41-vref"}, {.id = ALC882_FIXUP_INV_DMIC, .name = "inv-dmic"}, {.id = ALC882_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"}, + {.id = ALC887_FIXUP_ASUS_BASS, .name = "asus-bass"}, {.id = ALC1220_FIXUP_GB_DUAL_CODECS, .name = "dual-codecs"}, + {.id = ALC1220_FIXUP_CLEVO_P950, .name = "clevo-p950"}, {} }; @@ -2435,8 +2459,11 @@ static int patch_alc882(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog && spec->gen.beep_nid) - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (!spec->gen.no_analog && spec->gen.beep_nid) { + err = set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (err < 0) + goto error; + } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -2557,6 +2584,14 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = { static const struct hda_model_fixup alc262_fixup_models[] = { {.id = ALC262_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {.id = ALC262_FIXUP_FSC_H270, .name = "fsc-h270"}, + {.id = ALC262_FIXUP_FSC_S7110, .name = "fsc-s7110"}, + {.id = ALC262_FIXUP_HP_Z200, .name = "hp-z200"}, + {.id = ALC262_FIXUP_TYAN, .name = "tyan"}, + {.id = ALC262_FIXUP_LENOVO_3000, .name = "lenovo-3000"}, + {.id = ALC262_FIXUP_BENQ, .name = "benq"}, + {.id = ALC262_FIXUP_BENQ_T31, .name = "benq-t31"}, + {.id = ALC262_FIXUP_INTEL_BAYLEYBAY, .name = "bayleybay"}, {} }; @@ -2598,8 +2633,11 @@ static int patch_alc262(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog && spec->gen.beep_nid) - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (!spec->gen.no_analog && spec->gen.beep_nid) { + err = set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (err < 0) + goto error; + } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -2645,7 +2683,6 @@ static const struct snd_kcontrol_new alc268_beep_mixer[] = { .put = alc268_beep_switch_put, .private_value = HDA_COMPOSE_AMP_VAL(0x0f, 3, 1, HDA_INPUT) }, - { } }; /* set PCBEEP vol = 0, mute connections */ @@ -2686,6 +2723,7 @@ static const struct hda_fixup alc268_fixups[] = { static const struct hda_model_fixup alc268_fixup_models[] = { {.id = ALC268_FIXUP_INV_DMIC, .name = "inv-dmic"}, {.id = ALC268_FIXUP_HP_EAPD, .name = "hp-eapd"}, + {.id = ALC268_FIXUP_SPDIF, .name = "spdif"}, {} }; @@ -2713,7 +2751,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) static int patch_alc268(struct hda_codec *codec) { struct alc_spec *spec; - int err; + int i, err; /* ALC268 has no aa-loopback mixer */ err = alc_alloc_spec(codec, 0); @@ -2735,7 +2773,13 @@ static int patch_alc268(struct hda_codec *codec) if (err > 0 && !spec->gen.no_analog && spec->gen.autocfg.speaker_pins[0] != 0x1d) { - add_mixer(spec, alc268_beep_mixer); + for (i = 0; i < ARRAY_SIZE(alc268_beep_mixer); i++) { + if (!snd_hda_gen_add_kctl(&spec->gen, NULL, + &alc268_beep_mixer[i])) { + err = -ENOMEM; + goto error; + } + } snd_hda_add_verbs(codec, alc268_beep_init_verbs); if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) /* override the amp caps for beep generator */ @@ -3454,9 +3498,8 @@ static int alc269_resume(struct hda_codec *codec) * suspend, and won't restore the data after resume, so we restore it * in the driver. */ - if (spec->gpio_led) - snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_DATA, - spec->gpio_led); + if (spec->gpio_data) + alc_write_gpio_data(codec); if (spec->has_alc5505_dsp) alc5505_dsp_resume(codec); @@ -3696,18 +3739,10 @@ static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask, bool enabled) { struct alc_spec *spec = codec->spec; - unsigned int oldval = spec->gpio_led; if (spec->mute_led_polarity) enabled = !enabled; - - if (enabled) - spec->gpio_led &= ~mask; - else - spec->gpio_led |= mask; - if (spec->gpio_led != oldval) - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - spec->gpio_led); + alc_update_gpio_data(codec, mask, !enabled); /* muted -> LED on */ } /* turn on/off mute LED via GPIO per vmaster hook */ @@ -3720,104 +3755,79 @@ static void alc_fixup_gpio_mute_hook(void *private_data, int enabled) } /* turn on/off mic-mute LED via GPIO per capture hook */ -static void alc_fixup_gpio_mic_mute_hook(struct hda_codec *codec, - struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void alc_gpio_micmute_update(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (ucontrol) - alc_update_gpio_led(codec, spec->gpio_mic_led_mask, - ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]); + alc_update_gpio_led(codec, spec->gpio_mic_led_mask, + spec->gen.micmute_led.led_value); } -static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, - const struct hda_fixup *fix, int action) +/* setup mute and mic-mute GPIO bits, add hooks appropriately */ +static void alc_fixup_hp_gpio_led(struct hda_codec *codec, + int action, + unsigned int mute_mask, + unsigned int micmute_mask) { struct alc_spec *spec = codec->spec; - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x18 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x18 }, - {} - }; - if (action == HDA_FIXUP_ACT_PRE_PROBE) { + alc_fixup_gpio(codec, action, mute_mask | micmute_mask); + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + if (mute_mask) { + spec->gpio_mute_led_mask = mute_mask; spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook; - spec->gpio_led = 0; - spec->mute_led_polarity = 0; - spec->gpio_mute_led_mask = 0x08; - spec->gpio_mic_led_mask = 0x10; - snd_hda_add_verbs(codec, gpio_init); + } + if (micmute_mask) { + spec->gpio_mic_led_mask = micmute_mask; + snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); } } -static void alc286_fixup_hp_gpio_led(struct hda_codec *codec, +static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct alc_spec *spec = codec->spec; - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x22 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x22 }, - {} - }; + alc_fixup_hp_gpio_led(codec, action, 0x08, 0x10); +} - if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook; - spec->gpio_led = 0; - spec->mute_led_polarity = 0; - spec->gpio_mute_led_mask = 0x02; - spec->gpio_mic_led_mask = 0x20; - snd_hda_add_verbs(codec, gpio_init); - } +static void alc286_fixup_hp_gpio_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_hp_gpio_led(codec, action, 0x02, 0x20); } /* turn on/off mic-mute LED per capture hook */ -static void alc269_fixup_hp_cap_mic_mute_hook(struct hda_codec *codec, - struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void alc_cap_micmute_update(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int pinval, enable, disable; + unsigned int pinval; + if (!spec->cap_mute_led_nid) + return; pinval = snd_hda_codec_get_pin_target(codec, spec->cap_mute_led_nid); pinval &= ~AC_PINCTL_VREFEN; - enable = pinval | AC_PINCTL_VREF_80; - disable = pinval | AC_PINCTL_VREF_HIZ; - - if (!ucontrol) - return; - - if (ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]) - pinval = disable; + if (spec->gen.micmute_led.led_value) + pinval |= AC_PINCTL_VREF_80; else - pinval = enable; - - if (spec->cap_mute_led_nid) - snd_hda_set_pin_ctl_cache(codec, spec->cap_mute_led_nid, pinval); + pinval |= AC_PINCTL_VREF_HIZ; + snd_hda_set_pin_ctl_cache(codec, spec->cap_mute_led_nid, pinval); } static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x08 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x08 }, - {} - }; + alc_fixup_hp_gpio_led(codec, action, 0x08, 0); if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook; - spec->gpio_led = 0; - spec->mute_led_polarity = 0; - spec->gpio_mute_led_mask = 0x08; + /* Like hp_gpio_mic1_led, but also needs GPIO4 low to + * enable headphone amp + */ + spec->gpio_mask |= 0x10; + spec->gpio_dir |= 0x10; spec->cap_mute_led_nid = 0x18; - snd_hda_add_verbs(codec, gpio_init); + snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update); codec->power_filter = led_power_filter; } } @@ -3825,22 +3835,12 @@ static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec, static void alc280_fixup_hp_gpio4(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - /* Like hp_gpio_mic1_led, but also needs GPIO4 low to enable headphone amp */ struct alc_spec *spec = codec->spec; - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x18 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x18 }, - {} - }; + alc_fixup_hp_gpio_led(codec, action, 0x08, 0); if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook; - spec->gpio_led = 0; - spec->mute_led_polarity = 0; - spec->gpio_mute_led_mask = 0x08; spec->cap_mute_led_nid = 0x18; - snd_hda_add_verbs(codec, gpio_init); + snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update); codec->power_filter = led_power_filter; } } @@ -3890,38 +3890,29 @@ static int alc_register_micmute_input_device(struct hda_codec *codec) return 0; } +/* GPIO1 = set according to SKU external amp + * GPIO2 = mic mute hotkey + * GPIO3 = mute LED + * GPIO4 = mic mute LED + */ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - /* GPIO1 = set according to SKU external amp - GPIO2 = mic mute hotkey - GPIO3 = mute LED - GPIO4 = mic mute LED */ - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x1e }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x1a }, - { 0x01, AC_VERB_SET_GPIO_DATA, 0x02 }, - {} - }; - struct alc_spec *spec = codec->spec; + alc_fixup_hp_gpio_led(codec, action, 0x08, 0x10); if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->init_amp = ALC_INIT_DEFAULT; if (alc_register_micmute_input_device(codec) != 0) return; - snd_hda_add_verbs(codec, gpio_init); + spec->gpio_mask |= 0x06; + spec->gpio_dir |= 0x02; + spec->gpio_data |= 0x02; snd_hda_codec_write_cache(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x04); snd_hda_jack_detect_enable_callback(codec, codec->core.afg, gpio2_mic_hotkey_event); - - spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook; - spec->gpio_led = 0; - spec->mute_led_polarity = 0; - spec->gpio_mute_led_mask = 0x08; - spec->gpio_mic_led_mask = 0x10; return; } @@ -3929,40 +3920,28 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, return; switch (action) { - case HDA_FIXUP_ACT_PROBE: - spec->init_amp = ALC_INIT_DEFAULT; - break; case HDA_FIXUP_ACT_FREE: input_unregister_device(spec->kb_dev); spec->kb_dev = NULL; } } +/* Line2 = mic mute hotkey + * GPIO2 = mic mute LED + */ static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - /* Line2 = mic mute hotkey - GPIO2 = mic mute LED */ - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x04 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 }, - {} - }; - struct alc_spec *spec = codec->spec; + alc_fixup_hp_gpio_led(codec, action, 0, 0x04); if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->init_amp = ALC_INIT_DEFAULT; if (alc_register_micmute_input_device(codec) != 0) return; - snd_hda_add_verbs(codec, gpio_init); snd_hda_jack_detect_enable_callback(codec, 0x1b, gpio2_mic_hotkey_event); - - spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook; - spec->gpio_led = 0; - spec->mute_led_polarity = 0; - spec->gpio_mic_led_mask = 0x04; return; } @@ -3970,9 +3949,6 @@ static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec, return; switch (action) { - case HDA_FIXUP_ACT_PROBE: - spec->init_amp = ALC_INIT_DEFAULT; - break; case HDA_FIXUP_ACT_FREE: input_unregister_device(spec->kb_dev); spec->kb_dev = NULL; @@ -3988,14 +3964,10 @@ static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec, { struct alc_spec *spec = codec->spec; + alc269_fixup_hp_mute_led_micx(codec, fix, action, 0x1a); if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; - spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook; - spec->mute_led_polarity = 0; - spec->mute_led_nid = 0x1a; spec->cap_mute_led_nid = 0x18; - spec->gen.vmaster_mute_enum = 1; - codec->power_filter = led_power_filter; + snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update); } } @@ -4843,6 +4815,7 @@ static void alc_probe_headset_mode(struct hda_codec *codec) spec->headphone_mic_pin = cfg->inputs[i].pin; } + WARN_ON(spec->gen.cap_sync_hook); spec->gen.cap_sync_hook = alc_update_headset_mode_hook; spec->gen.automute_hook = alc_update_headset_mode; spec->gen.hp_automute_hook = alc_update_headset_jack_cb; @@ -4934,13 +4907,10 @@ static void alc288_update_headset_jack_cb(struct hda_codec *codec, struct hda_jack_callback *jack) { struct alc_spec *spec = codec->spec; - int present; alc_update_headset_jack_cb(codec, jack); /* Headset Mic enable or disable, only for Dell Dino */ - present = spec->gen.hp_jack_present ? 0x40 : 0; - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - present); + alc_update_gpio_data(codec, 0x40, spec->gen.hp_jack_present); } static void alc_fixup_headset_mode_dell_alc288(struct hda_codec *codec, @@ -4949,6 +4919,9 @@ static void alc_fixup_headset_mode_dell_alc288(struct hda_codec *codec, alc_fixup_headset_mode(codec, fix, action); if (action == HDA_FIXUP_ACT_PROBE) { struct alc_spec *spec = codec->spec; + /* toggled via hp_automute_hook */ + spec->gpio_mask |= 0x40; + spec->gpio_dir |= 0x40; spec->gen.hp_automute_hook = alc288_update_headset_jack_cb; } } @@ -4969,7 +4942,7 @@ static void alc_no_shutup(struct hda_codec *codec) static void alc_fixup_no_shutup(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - if (action == HDA_FIXUP_ACT_PROBE) { + if (action == HDA_FIXUP_ACT_PRE_PROBE) { struct alc_spec *spec = codec->spec; spec->shutup = alc_no_shutup; } @@ -5051,10 +5024,9 @@ static void alc_fixup_dell_xps13(struct hda_codec *codec, * it causes a click noise at start up */ snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); + spec->shutup = alc_shutup_dell_xps13; break; case HDA_FIXUP_ACT_PROBE: - spec->shutup = alc_shutup_dell_xps13; - /* Make the internal mic the default input source. */ for (i = 0; i < imux->num_items; i++) { if (spec->gen.imux_pins[i] == 0x12) { @@ -5231,13 +5203,6 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - /* TX300 needs to set up GPIO2 for the speaker amp */ - static const struct hda_verb gpio2_verbs[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x04 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 }, - { 0x01, AC_VERB_SET_GPIO_DATA, 0x04 }, - {} - }; static const struct hda_pintbl dock_pins[] = { { 0x1b, 0x21114000 }, /* dock speaker pin */ {} @@ -5245,13 +5210,18 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec, switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: - snd_hda_add_verbs(codec, gpio2_verbs); + spec->init_amp = ALC_INIT_DEFAULT; + /* TX300 needs to set up GPIO2 for the speaker amp */ + alc_setup_gpio(codec, 0x04); snd_hda_apply_pincfgs(codec, dock_pins); spec->gen.auto_mute_via_amp = 1; spec->gen.automute_hook = asus_tx300_automute; snd_hda_jack_detect_enable_callback(codec, 0x1b, snd_hda_gen_hp_automute); break; + case HDA_FIXUP_ACT_PROBE: + spec->init_amp = ALC_INIT_DEFAULT; + break; case HDA_FIXUP_ACT_BUILD: /* this is a bit tricky; give more sane names for the main * (tablet) speaker and the dock speaker, respectively @@ -5325,30 +5295,26 @@ static void alc280_fixup_hp_9480m(struct hda_codec *codec, int action) { struct alc_spec *spec = codec->spec; - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x18 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x18 }, - {} - }; + alc_fixup_hp_gpio_led(codec, action, 0x08, 0); if (action == HDA_FIXUP_ACT_PRE_PROBE) { - /* Set the hooks to turn the headphone amp on/off - * as needed - */ - spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; + /* amp at GPIO4; toggled via alc280_hp_gpio4_automute_hook() */ + spec->gpio_mask |= 0x10; + spec->gpio_dir |= 0x10; spec->gen.hp_automute_hook = alc280_hp_gpio4_automute_hook; + } +} - /* The GPIOs are currently off */ - spec->gpio_led = 0; - - /* GPIO3 is connected to the output mute LED, - * high is on, low is off - */ - spec->mute_led_polarity = 0; - spec->gpio_mute_led_mask = 0x08; +static void alc275_fixup_gpio4_off(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; - /* Initialize GPIO configuration */ - snd_hda_add_verbs(codec, gpio_init); + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gpio_mask |= 0x04; + spec->gpio_dir |= 0x04; + /* set data bit low */ } } @@ -5492,7 +5458,6 @@ enum { ALC280_FIXUP_HP_9480M, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC288_FIXUP_DELL_XPS_13_GPIO6, ALC288_FIXUP_DELL_XPS_13, ALC288_FIXUP_DISABLE_AAMIX, ALC292_FIXUP_DELL_E7X, @@ -5540,13 +5505,8 @@ static const struct hda_fixup alc269_fixups[] = { } }, [ALC275_FIXUP_SONY_VAIO_GPIO2] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - {0x01, AC_VERB_SET_GPIO_MASK, 0x04}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, - { } - }, + .type = HDA_FIXUP_FUNC, + .v.func = alc275_fixup_gpio4_off, .chained = true, .chain_id = ALC269_FIXUP_SONY_VAIO }, @@ -6113,22 +6073,11 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC288_FIXUP_DELL_HEADSET_MODE }, - [ALC288_FIXUP_DELL_XPS_13_GPIO6] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - {0x01, AC_VERB_SET_GPIO_MASK, 0x40}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x40}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, - { } - }, - .chained = true, - .chain_id = ALC288_FIXUP_DELL1_MIC_NO_PRESENCE - }, [ALC288_FIXUP_DISABLE_AAMIX] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, .chained = true, - .chain_id = ALC288_FIXUP_DELL_XPS_13_GPIO6 + .chain_id = ALC288_FIXUP_DELL1_MIC_NO_PRESENCE }, [ALC288_FIXUP_DELL_XPS_13] = { .type = HDA_FIXUP_FUNC, @@ -6291,14 +6240,9 @@ static const struct hda_fixup alc269_fixups[] = { .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE }, [ALC256_FIXUP_ASUS_AIO_GPIO2] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - /* Set up GPIO2 for the speaker amp */ - { 0x01, AC_VERB_SET_GPIO_MASK, 0x04 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 }, - { 0x01, AC_VERB_SET_GPIO_DATA, 0x04 }, - {} - }, + .type = HDA_FIXUP_FUNC, + /* Set up GPIO2 for the speaker amp */ + .v.func = alc_fixup_gpio4, }, [ALC233_FIXUP_ASUS_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, @@ -6530,6 +6474,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -6713,13 +6658,95 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, .name = "hp-dock-gpio-mic1-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"}, + {.id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, .name = "dell-headset3"}, + {.id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, .name = "dell-headset4"}, {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"}, {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, {.id = ALC292_FIXUP_TPT460, .name = "tpt460"}, + {.id = ALC298_FIXUP_TPT470_DOCK, .name = "tpt470-dock"}, {.id = ALC233_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, {.id = ALC700_FIXUP_INTEL_REFERENCE, .name = "alc700-ref"}, + {.id = ALC269_FIXUP_SONY_VAIO, .name = "vaio"}, + {.id = ALC269_FIXUP_DELL_M101Z, .name = "dell-m101z"}, + {.id = ALC269_FIXUP_ASUS_G73JW, .name = "asus-g73jw"}, + {.id = ALC269_FIXUP_LENOVO_EAPD, .name = "lenovo-eapd"}, + {.id = ALC275_FIXUP_SONY_HWEQ, .name = "sony-hweq"}, + {.id = ALC269_FIXUP_PCM_44K, .name = "pcm44k"}, + {.id = ALC269_FIXUP_LIFEBOOK, .name = "lifebook"}, + {.id = ALC269_FIXUP_LIFEBOOK_EXTMIC, .name = "lifebook-extmic"}, + {.id = ALC269_FIXUP_LIFEBOOK_HP_PIN, .name = "lifebook-hp-pin"}, + {.id = ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC, .name = "lifebook-u7x7"}, + {.id = ALC269VB_FIXUP_AMIC, .name = "alc269vb-amic"}, + {.id = ALC269VB_FIXUP_DMIC, .name = "alc269vb-dmic"}, + {.id = ALC269_FIXUP_HP_MUTE_LED_MIC1, .name = "hp-mute-led-mic1"}, + {.id = ALC269_FIXUP_HP_MUTE_LED_MIC2, .name = "hp-mute-led-mic2"}, + {.id = ALC269_FIXUP_HP_MUTE_LED_MIC3, .name = "hp-mute-led-mic3"}, + {.id = ALC269_FIXUP_HP_GPIO_MIC1_LED, .name = "hp-gpio-mic1"}, + {.id = ALC269_FIXUP_HP_LINE1_MIC1_LED, .name = "hp-line1-mic1"}, + {.id = ALC269_FIXUP_NO_SHUTUP, .name = "noshutup"}, + {.id = ALC286_FIXUP_SONY_MIC_NO_PRESENCE, .name = "sony-nomic"}, + {.id = ALC269_FIXUP_ASPIRE_HEADSET_MIC, .name = "aspire-headset-mic"}, + {.id = ALC269_FIXUP_ASUS_X101, .name = "asus-x101"}, + {.id = ALC271_FIXUP_HP_GATE_MIC_JACK, .name = "acer-ao7xx"}, + {.id = ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572, .name = "acer-aspire-e1"}, + {.id = ALC269_FIXUP_ACER_AC700, .name = "acer-ac700"}, + {.id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST, .name = "limit-mic-boost"}, + {.id = ALC269VB_FIXUP_ASUS_ZENBOOK, .name = "asus-zenbook"}, + {.id = ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A, .name = "asus-zenbook-ux31a"}, + {.id = ALC269VB_FIXUP_ORDISSIMO_EVE2, .name = "ordissimo"}, + {.id = ALC282_FIXUP_ASUS_TX300, .name = "asus-tx300"}, + {.id = ALC283_FIXUP_INT_MIC, .name = "alc283-int-mic"}, + {.id = ALC290_FIXUP_MONO_SPEAKERS_HSJACK, .name = "mono-speakers"}, + {.id = ALC290_FIXUP_SUBWOOFER_HSJACK, .name = "alc290-subwoofer"}, + {.id = ALC269_FIXUP_THINKPAD_ACPI, .name = "thinkpad"}, + {.id = ALC269_FIXUP_DMIC_THINKPAD_ACPI, .name = "dmic-thinkpad"}, + {.id = ALC255_FIXUP_ACER_MIC_NO_PRESENCE, .name = "alc255-acer"}, + {.id = ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, .name = "alc255-asus"}, + {.id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc255-dell1"}, + {.id = ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "alc255-dell2"}, + {.id = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc293-dell1"}, + {.id = ALC283_FIXUP_HEADSET_MIC, .name = "alc283-headset"}, + {.id = ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, .name = "alc255-dell-mute"}, + {.id = ALC282_FIXUP_ASPIRE_V5_PINS, .name = "aspire-v5"}, + {.id = ALC280_FIXUP_HP_GPIO4, .name = "hp-gpio4"}, + {.id = ALC286_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, + {.id = ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, .name = "hp-gpio2-hotkey"}, + {.id = ALC280_FIXUP_HP_DOCK_PINS, .name = "hp-dock-pins"}, + {.id = ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, .name = "hp-dock-gpio-mic"}, + {.id = ALC280_FIXUP_HP_9480M, .name = "hp-9480m"}, + {.id = ALC288_FIXUP_DELL_HEADSET_MODE, .name = "alc288-dell-headset"}, + {.id = ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc288-dell1"}, + {.id = ALC288_FIXUP_DELL_XPS_13, .name = "alc288-dell-xps13"}, + {.id = ALC292_FIXUP_DELL_E7X, .name = "dell-e7x"}, + {.id = ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK, .name = "alc293-dell"}, + {.id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc298-dell1"}, + {.id = ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, .name = "alc298-dell-aio"}, + {.id = ALC275_FIXUP_DELL_XPS, .name = "alc275-dell-xps"}, + {.id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, .name = "alc256-dell-xps13"}, + {.id = ALC293_FIXUP_LENOVO_SPK_NOISE, .name = "lenovo-spk-noise"}, + {.id = ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, .name = "lenovo-hotkey"}, + {.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"}, + {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc255-dell1"}, + {.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"}, + {.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"}, + {.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"}, + {.id = ALC298_FIXUP_SPK_VOLUME, .name = "alc298-spk-volume"}, + {.id = ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER, .name = "dell-inspiron-7559"}, + {.id = ALC269_FIXUP_ATIV_BOOK_8, .name = "ativ-book"}, + {.id = ALC221_FIXUP_HP_MIC_NO_PRESENCE, .name = "alc221-hp-mic"}, + {.id = ALC256_FIXUP_ASUS_HEADSET_MODE, .name = "alc256-asus-headset"}, + {.id = ALC256_FIXUP_ASUS_MIC, .name = "alc256-asus-mic"}, + {.id = ALC256_FIXUP_ASUS_AIO_GPIO2, .name = "alc256-asus-aio"}, + {.id = ALC233_FIXUP_ASUS_MIC_NO_PRESENCE, .name = "alc233-asus"}, + {.id = ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE, .name = "alc233-eapd"}, + {.id = ALC294_FIXUP_LENOVO_MIC_LOCATION, .name = "alc294-lenovo-mic"}, + {.id = ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE, .name = "alc225-wyse"}, + {.id = ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, .name = "alc274-dell-aio"}, + {.id = ALC255_FIXUP_DUMMY_LINEOUT_VERB, .name = "alc255-dummy-lineout"}, + {.id = ALC255_FIXUP_DELL_HEADSET_MIC, .name = "alc255-dell-headset"}, + {.id = ALC295_FIXUP_HP_X360, .name = "alc295-hp-x360"}, {} }; #define ALC225_STANDARD_PINS \ @@ -6983,7 +7010,7 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60130}, {0x19, 0x03a11020}, {0x21, 0x0321101f}), - SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL_XPS_13_GPIO6, + SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60120}, {0x14, 0x90170110}, {0x21, 0x0321101f}), @@ -7140,18 +7167,6 @@ static int patch_alc269(struct hda_codec *codec) spec->shutup = alc_default_shutup; spec->init_hook = alc_default_init; - snd_hda_pick_fixup(codec, alc269_fixup_models, - alc269_fixup_tbl, alc269_fixups); - snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups); - snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl, - alc269_fixups); - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - - alc_auto_parse_customize_define(codec); - - if (has_cdefine_beep(codec)) - spec->gen.beep_nid = 0x01; - switch (codec->core.vendor_id) { case 0x10ec0269: spec->codec_variant = ALC269_TYPE_ALC269VA; @@ -7271,13 +7286,28 @@ static int patch_alc269(struct hda_codec *codec) spec->init_hook = alc5505_dsp_init; } + snd_hda_pick_fixup(codec, alc269_fixup_models, + alc269_fixup_tbl, alc269_fixups); + snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups); + snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl, + alc269_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + + alc_auto_parse_customize_define(codec); + + if (has_cdefine_beep(codec)) + spec->gen.beep_nid = 0x01; + /* automatic parse from the BIOS config */ err = alc269_parse_auto_config(codec); if (err < 0) goto error; - if (!spec->gen.no_analog && spec->gen.beep_nid && spec->gen.mixer_nid) - set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT); + if (!spec->gen.no_analog && spec->gen.beep_nid && spec->gen.mixer_nid) { + err = set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT); + if (err < 0) + goto error; + } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -7406,8 +7436,11 @@ static int patch_alc861(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog) - set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); + if (!spec->gen.no_analog) { + err = set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); + if (err < 0) + goto error; + } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -7447,16 +7480,21 @@ static void alc861vd_fixup_dallas(struct hda_codec *codec, } } +/* reset GPIO1 */ +static void alc660vd_fixup_asus_gpio1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gpio_mask |= 0x02; + alc_fixup_gpio(codec, action, 0x01); +} + static const struct hda_fixup alc861vd_fixups[] = { [ALC660VD_FIX_ASUS_GPIO1] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - /* reset GPIO1 */ - {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, - { } - } + .type = HDA_FIXUP_FUNC, + .v.func = alc660vd_fixup_asus_gpio1, }, [ALC861VD_FIX_DALLAS] = { .type = HDA_FIXUP_FUNC, @@ -7495,8 +7533,11 @@ static int patch_alc861vd(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog) - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (!spec->gen.no_analog) { + err = set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (err < 0) + goto error; + } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -7577,7 +7618,7 @@ static unsigned int gpio_led_power_filter(struct hda_codec *codec, unsigned int power_state) { struct alc_spec *spec = codec->spec; - if (nid == codec->core.afg && power_state == AC_PWRST_D3 && spec->gpio_led) + if (nid == codec->core.afg && power_state == AC_PWRST_D3 && spec->gpio_data) return AC_PWRST_D0; return power_state; } @@ -7586,18 +7627,10 @@ static void alc662_fixup_led_gpio1(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - static const struct hda_verb gpio_init[] = { - { 0x01, AC_VERB_SET_GPIO_MASK, 0x01 }, - { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01 }, - {} - }; + alc_fixup_hp_gpio_led(codec, action, 0x01, 0); if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; - spec->gpio_led = 0; spec->mute_led_polarity = 1; - spec->gpio_mute_led_mask = 0x01; - snd_hda_add_verbs(codec, gpio_init); codec->power_filter = gpio_led_power_filter; } } @@ -8110,7 +8143,10 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { }; static const struct hda_model_fixup alc662_fixup_models[] = { + {.id = ALC662_FIXUP_ASPIRE, .name = "aspire"}, + {.id = ALC662_FIXUP_IDEAPAD, .name = "ideapad"}, {.id = ALC272_FIXUP_MARIO, .name = "mario"}, + {.id = ALC662_FIXUP_HP_RP5800, .name = "hp-rp5800"}, {.id = ALC662_FIXUP_ASUS_MODE1, .name = "asus-mode1"}, {.id = ALC662_FIXUP_ASUS_MODE2, .name = "asus-mode2"}, {.id = ALC662_FIXUP_ASUS_MODE3, .name = "asus-mode3"}, @@ -8119,8 +8155,23 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_ASUS_MODE6, .name = "asus-mode6"}, {.id = ALC662_FIXUP_ASUS_MODE7, .name = "asus-mode7"}, {.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"}, + {.id = ALC662_FIXUP_ZOTAC_Z68, .name = "zotac-z68"}, {.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {.id = ALC662_FIXUP_DELL_MIC_NO_PRESENCE, .name = "alc662-headset-multi"}, {.id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, + {.id = ALC662_FIXUP_HEADSET_MODE, .name = "alc662-headset"}, + {.id = ALC668_FIXUP_HEADSET_MODE, .name = "alc668-headset"}, + {.id = ALC662_FIXUP_BASS_16, .name = "bass16"}, + {.id = ALC662_FIXUP_BASS_1A, .name = "bass1a"}, + {.id = ALC668_FIXUP_AUTO_MUTE, .name = "automute"}, + {.id = ALC668_FIXUP_DELL_XPS13, .name = "dell-xps13"}, + {.id = ALC662_FIXUP_ASUS_Nx50, .name = "asus-nx50"}, + {.id = ALC668_FIXUP_ASUS_Nx51, .name = "asus-nx51"}, + {.id = ALC891_FIXUP_HEADSET_MODE, .name = "alc891-headset"}, + {.id = ALC891_FIXUP_DELL_MIC_NO_PRESENCE, .name = "alc891-headset-multi"}, + {.id = ALC662_FIXUP_ACER_VERITON, .name = "acer-veriton"}, + {.id = ALC892_FIXUP_ASROCK_MOBO, .name = "asrock-mobo"}, + {.id = ALC662_FIXUP_USI_HEADSET_MODE, .name = "usi-headset"}, {.id = ALC662_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, {} }; @@ -8214,18 +8265,20 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->gen.no_analog && spec->gen.beep_nid) { switch (codec->core.vendor_id) { case 0x10ec0662: - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + err = set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); break; case 0x10ec0272: case 0x10ec0663: case 0x10ec0665: case 0x10ec0668: - set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); + err = set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); break; case 0x10ec0273: - set_beep_amp(spec, 0x0b, 0x03, HDA_INPUT); + err = set_beep_amp(spec, 0x0b, 0x03, HDA_INPUT); break; } + if (err < 0) + goto error; } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 63d15b545b33..046705b4691a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -332,33 +332,15 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, } /* hook for controlling mic-mute LED GPIO */ -static void stac_capture_led_hook(struct hda_codec *codec, - struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void stac_capture_led_update(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - unsigned int mask; - bool cur_mute, prev_mute; - if (!kcontrol || !ucontrol) - return; - - mask = 1U << snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - prev_mute = !spec->mic_enabled; - if (ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]) - spec->mic_enabled |= mask; + if (spec->gen.micmute_led.led_value) + spec->gpio_data |= spec->mic_mute_led_gpio; else - spec->mic_enabled &= ~mask; - cur_mute = !spec->mic_enabled; - if (cur_mute != prev_mute) { - if (cur_mute) - spec->gpio_data |= spec->mic_mute_led_gpio; - else - spec->gpio_data &= ~spec->mic_mute_led_gpio; - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data); - } + spec->gpio_data &= ~spec->mic_mute_led_gpio; + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); } static int stac_vrefout_set(struct hda_codec *codec, @@ -4656,8 +4638,7 @@ static void stac_setup_gpio(struct hda_codec *codec) spec->gpio_dir |= spec->mic_mute_led_gpio; spec->mic_enabled = 0; spec->gpio_data |= spec->mic_mute_led_gpio; - - spec->gen.cap_sync_hook = stac_capture_led_hook; + snd_hda_gen_add_micmute_led(codec, stac_capture_led_update); } } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index fc30d1e8aa76..6b9617aee0e6 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -90,13 +90,6 @@ enum VIA_HDA_CODEC { struct via_spec { struct hda_gen_spec gen; - /* codec parameterization */ - const struct snd_kcontrol_new *mixers[6]; - unsigned int num_mixers; - - const struct hda_verb *init_verbs[5]; - unsigned int num_iverbs; - /* HP mode source */ unsigned int dmic_enabled; enum VIA_HDA_CODEC codec_type; @@ -107,8 +100,6 @@ struct via_spec { /* work to check hp jack state */ int hp_work_active; int vt1708_jack_detect; - - unsigned int beep_amp; }; static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec); @@ -262,69 +253,51 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol, return 1; } -static const struct snd_kcontrol_new via_pin_power_ctl_enum[] = { - { +static const struct snd_kcontrol_new via_pin_power_ctl_enum = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Dynamic Power-Control", .info = via_pin_power_ctl_info, .get = via_pin_power_ctl_get, .put = via_pin_power_ctl_put, - }, - {} /* terminator */ }; #ifdef CONFIG_SND_HDA_INPUT_BEEP -static inline void set_beep_amp(struct via_spec *spec, hda_nid_t nid, - int idx, int dir) -{ - spec->gen.beep_nid = nid; - spec->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir); -} - /* additional beep mixers; the actual parameters are overwritten at build */ -static const struct snd_kcontrol_new cxt_beep_mixer[] = { +static const struct snd_kcontrol_new via_beep_mixer[] = { HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0, 1, 0, HDA_OUTPUT), HDA_CODEC_MUTE_BEEP_MONO("Beep Playback Switch", 0, 1, 0, HDA_OUTPUT), - { } /* end */ }; -/* create beep controls if needed */ -static int add_beep_ctls(struct hda_codec *codec) +static int set_beep_amp(struct via_spec *spec, hda_nid_t nid, + int idx, int dir) { - struct via_spec *spec = codec->spec; - int err; + struct snd_kcontrol_new *knew; + unsigned int beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir); + int i; - if (spec->beep_amp) { - const struct snd_kcontrol_new *knew; - for (knew = cxt_beep_mixer; knew->name; knew++) { - struct snd_kcontrol *kctl; - kctl = snd_ctl_new1(knew, codec); - if (!kctl) - return -ENOMEM; - kctl->private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, 0, kctl); - if (err < 0) - return err; - } + spec->gen.beep_nid = nid; + for (i = 0; i < ARRAY_SIZE(via_beep_mixer); i++) { + knew = snd_hda_gen_add_kctl(&spec->gen, NULL, + &via_beep_mixer[i]); + if (!knew) + return -ENOMEM; + knew->private_value = beep_amp; } return 0; } -static void auto_parse_beep(struct hda_codec *codec) +static int auto_parse_beep(struct hda_codec *codec) { struct via_spec *spec = codec->spec; hda_nid_t nid; for_each_hda_codec_node(nid, codec) - if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_BEEP) { - set_beep_amp(spec, nid, 0, HDA_OUTPUT); - break; - } + if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_BEEP) + return set_beep_amp(spec, nid, 0, HDA_OUTPUT); + return 0; } #else -#define set_beep_amp(spec, nid, idx, dir) /* NOP */ -#define add_beep_ctls(codec) 0 -#define auto_parse_beep(codec) +#define auto_parse_beep(codec) 0 #endif /* check AA path's mute status */ @@ -403,30 +376,6 @@ static void analog_low_current_mode(struct hda_codec *codec) return __analog_low_current_mode(codec, false); } -static int via_build_controls(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int err, i; - - err = snd_hda_gen_build_controls(codec); - if (err < 0) - return err; - - err = add_beep_ctls(codec); - if (err < 0) - return err; - - spec->mixers[spec->num_mixers++] = via_pin_power_ctl_enum; - - for (i = 0; i < spec->num_mixers; i++) { - err = snd_hda_add_new_ctls(codec, spec->mixers[i]); - if (err < 0) - return err; - } - - return 0; -} - static void via_playback_pcm_hook(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream, @@ -481,7 +430,7 @@ static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid) static int via_init(struct hda_codec *codec); static const struct hda_codec_ops via_patch_ops = { - .build_controls = via_build_controls, + .build_controls = snd_hda_gen_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = via_init, .free = via_free, @@ -545,16 +494,13 @@ static int vt1708_jack_detect_put(struct snd_kcontrol *kcontrol, return 1; } -static const struct snd_kcontrol_new vt1708_jack_detect_ctl[] = { - { +static const struct snd_kcontrol_new vt1708_jack_detect_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Jack Detect", .count = 1, .info = snd_ctl_boolean_mono_info, .get = vt1708_jack_detect_get, .put = vt1708_jack_detect_put, - }, - {} /* terminator */ }; static const struct badness_table via_main_out_badness = { @@ -586,12 +532,17 @@ static int via_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - auto_parse_beep(codec); - err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); if (err < 0) return err; + err = auto_parse_beep(codec); + if (err < 0) + return err; + + if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &via_pin_power_ctl_enum)) + return -ENOMEM; + /* disable widget PM at start for compatibility */ codec->power_save_node = 0; spec->gen.power_down_unused = 0; @@ -600,12 +551,6 @@ static int via_parse_auto_config(struct hda_codec *codec) static int via_init(struct hda_codec *codec) { - struct via_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_iverbs; i++) - snd_hda_sequence_write(codec, spec->init_verbs[i]); - /* init power states */ __analog_low_current_mode(codec, true); @@ -623,7 +568,7 @@ static int vt1708_build_controls(struct hda_codec *codec) int err; int old_interval = codec->jackpoll_interval; codec->jackpoll_interval = msecs_to_jiffies(100); - err = via_build_controls(codec); + err = snd_hda_gen_build_controls(codec); codec->jackpoll_interval = old_interval; return err; } @@ -684,22 +629,29 @@ static int patch_vt1708(struct hda_codec *codec) vt1708_set_pinconfig_connect(codec, VT1708_HP_PIN_NID); vt1708_set_pinconfig_connect(codec, VT1708_CD_PIN_NID); + err = snd_hda_add_verbs(codec, vt1708_init_verbs); + if (err < 0) + goto error; + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } + if (err < 0) + goto error; /* add jack detect on/off control */ - spec->mixers[spec->num_mixers++] = vt1708_jack_detect_ctl; - - spec->init_verbs[spec->num_iverbs++] = vt1708_init_verbs; + if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &vt1708_jack_detect_ctl)) { + err = -ENOMEM; + goto error; + } /* clear jackpoll_interval again; it's set dynamically */ codec->jackpoll_interval = 0; return 0; + + error: + via_free(codec); + return err; } static int patch_vt1709(struct hda_codec *codec) @@ -715,12 +667,14 @@ static int patch_vt1709(struct hda_codec *codec) spec->gen.mixer_nid = 0x18; err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } + if (err < 0) + goto error; return 0; + + error: + via_free(codec); + return err; } static int patch_vt1708S(struct hda_codec *codec); @@ -741,12 +695,14 @@ static int patch_vt1708B(struct hda_codec *codec) /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } + if (err < 0) + goto error; return 0; + + error: + via_free(codec); + return err; } /* Patch for VT1708S */ @@ -791,16 +747,20 @@ static int patch_vt1708S(struct hda_codec *codec) if (codec->core.vendor_id == 0x11064397) snd_hda_codec_set_name(codec, "VT1705"); + err = snd_hda_add_verbs(codec, vt1708S_init_verbs); + if (err < 0) + goto error; + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } - - spec->init_verbs[spec->num_iverbs++] = vt1708S_init_verbs; + if (err < 0) + goto error; return 0; + + error: + via_free(codec); + return err; } /* Patch for VT1702 */ @@ -832,16 +792,20 @@ static int patch_vt1702(struct hda_codec *codec) (0x5 << AC_AMPCAP_STEP_SIZE_SHIFT) | (1 << AC_AMPCAP_MUTE_SHIFT)); + err = snd_hda_add_verbs(codec, vt1702_init_verbs); + if (err < 0) + goto error; + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } - - spec->init_verbs[spec->num_iverbs++] = vt1702_init_verbs; + if (err < 0) + goto error; return 0; + + error: + via_free(codec); + return err; } /* Patch for VT1718S */ @@ -904,16 +868,20 @@ static int patch_vt1718S(struct hda_codec *codec) override_mic_boost(codec, 0x29, 0, 3, 40); add_secret_dac_path(codec); + err = snd_hda_add_verbs(codec, vt1718S_init_verbs); + if (err < 0) + goto error; + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } - - spec->init_verbs[spec->num_iverbs++] = vt1718S_init_verbs; + if (err < 0) + goto error; return 0; + + error: + via_free(codec); + return err; } /* Patch for VT1716S */ @@ -955,9 +923,9 @@ static int vt1716s_dmic_put(struct snd_kcontrol *kcontrol, return 1; } -static const struct snd_kcontrol_new vt1716s_dmic_mixer[] = { - HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x22, 0x0, HDA_INPUT), - { +static const struct snd_kcontrol_new vt1716s_dmic_mixer_vol = + HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x22, 0x0, HDA_INPUT); +static const struct snd_kcontrol_new vt1716s_dmic_mixer_sw = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Digital Mic Capture Switch", .subdevice = HDA_SUBDEV_NID_FLAG | 0x26, @@ -965,16 +933,12 @@ static const struct snd_kcontrol_new vt1716s_dmic_mixer[] = { .info = vt1716s_dmic_info, .get = vt1716s_dmic_get, .put = vt1716s_dmic_put, - }, - {} /* end */ }; /* mono-out mixer elements */ -static const struct snd_kcontrol_new vt1716S_mono_out_mixer[] = { - HDA_CODEC_MUTE("Mono Playback Switch", 0x2a, 0x0, HDA_OUTPUT), - { } /* end */ -}; +static const struct snd_kcontrol_new vt1716S_mono_out_mixer = + HDA_CODEC_MUTE("Mono Playback Switch", 0x2a, 0x0, HDA_OUTPUT); static const struct hda_verb vt1716S_init_verbs[] = { /* Enable Boost Volume backdoor */ @@ -1000,19 +964,27 @@ static int patch_vt1716S(struct hda_codec *codec) override_mic_boost(codec, 0x1a, 0, 3, 40); override_mic_boost(codec, 0x1e, 0, 3, 40); + err = snd_hda_add_verbs(codec, vt1716S_init_verbs); + if (err < 0) + goto error; + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } - - spec->init_verbs[spec->num_iverbs++] = vt1716S_init_verbs; + if (err < 0) + goto error; - spec->mixers[spec->num_mixers++] = vt1716s_dmic_mixer; - spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer; + if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &vt1716s_dmic_mixer_vol) || + !snd_hda_gen_add_kctl(&spec->gen, NULL, &vt1716s_dmic_mixer_sw) || + !snd_hda_gen_add_kctl(&spec->gen, NULL, &vt1716S_mono_out_mixer)) { + err = -ENOMEM; + goto error; + } return 0; + + error: + via_free(codec); + return err; } /* for vt2002P */ @@ -1107,19 +1079,23 @@ static int patch_vt2002P(struct hda_codec *codec) snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - /* automatic parse from the BIOS config */ - err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } - if (spec->codec_type == VT1802) - spec->init_verbs[spec->num_iverbs++] = vt1802_init_verbs; + err = snd_hda_add_verbs(codec, vt1802_init_verbs); else - spec->init_verbs[spec->num_iverbs++] = vt2002P_init_verbs; + err = snd_hda_add_verbs(codec, vt2002P_init_verbs); + if (err < 0) + goto error; + + /* automatic parse from the BIOS config */ + err = via_parse_auto_config(codec); + if (err < 0) + goto error; return 0; + + error: + via_free(codec); + return err; } /* for vt1812 */ @@ -1148,16 +1124,20 @@ static int patch_vt1812(struct hda_codec *codec) override_mic_boost(codec, 0x29, 0, 3, 40); add_secret_dac_path(codec); + err = snd_hda_add_verbs(codec, vt1812_init_verbs); + if (err < 0) + goto error; + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } - - spec->init_verbs[spec->num_iverbs++] = vt1812_init_verbs; + if (err < 0) + goto error; return 0; + + error: + via_free(codec); + return err; } /* patch for vt3476 */ @@ -1185,16 +1165,20 @@ static int patch_vt3476(struct hda_codec *codec) spec->gen.mixer_nid = 0x3f; add_secret_dac_path(codec); + err = snd_hda_add_verbs(codec, vt3476_init_verbs); + if (err < 0) + goto error; + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); - if (err < 0) { - via_free(codec); - return err; - } - - spec->init_verbs[spec->num_iverbs++] = vt3476_init_verbs; + if (err < 0) + goto error; return 0; + + error: + via_free(codec); + return err; } /* diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 65bb3ac6af4c..97f49b751e6e 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -27,17 +27,11 @@ static void update_tpacpi_mute_led(void *private_data, int enabled) led_set_func(TPACPI_LED_MUTE, !enabled); } -static void update_tpacpi_micmute_led(struct hda_codec *codec, - struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void update_tpacpi_micmute(struct hda_codec *codec) { - if (!ucontrol || !led_set_func) - return; - if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { - /* TODO: How do I verify if it's a mono or stereo here? */ - bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]; - led_set_func(TPACPI_LED_MICMUTE, !val); - } + struct hda_gen_spec *spec = codec->spec; + + led_set_func(TPACPI_LED_MICMUTE, spec->micmute_led.led_value); } static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, @@ -63,15 +57,10 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, spec->vmaster_mute.hook = update_tpacpi_mute_led; removefunc = false; } - if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { - if (spec->num_adc_nids > 1 && !spec->dyn_adc_switch) - codec_dbg(codec, - "Skipping micmute LED control due to several ADCs"); - else { - spec->cap_sync_hook = update_tpacpi_micmute_led; - removefunc = false; - } - } + if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0 && + snd_hda_gen_add_micmute_led(codec, + update_tpacpi_micmute) > 0) + removefunc = false; } if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { diff --git a/sound/pci/ice1712/ak4xxx.c b/sound/pci/ice1712/ak4xxx.c index 179ef7a5f0d1..a553897a4c4f 100644 --- a/sound/pci/ice1712/ak4xxx.c +++ b/sound/pci/ice1712/ak4xxx.c @@ -179,18 +179,6 @@ int snd_ice1712_akm4xxx_build_controls(struct snd_ice1712 *ice) return 0; } -static int __init alsa_ice1712_akm4xxx_module_init(void) -{ - return 0; -} - -static void __exit alsa_ice1712_akm4xxx_module_exit(void) -{ -} - -module_init(alsa_ice1712_akm4xxx_module_init) -module_exit(alsa_ice1712_akm4xxx_module_exit) - EXPORT_SYMBOL(snd_ice1712_akm4xxx_init); EXPORT_SYMBOL(snd_ice1712_akm4xxx_free); EXPORT_SYMBOL(snd_ice1712_akm4xxx_build_controls); diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index d7366ade5a25..c97b5528e4b8 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -314,26 +314,7 @@ static struct snd_kcontrol_new prodigy_hd2_controls[] = { /* --------------- */ -/* - * Logarithmic volume values for WM87*6 - * Computed as 20 * Log10(255 / x) - */ -static const unsigned char wm_vol[256] = { - 127, 48, 42, 39, 36, 34, 33, 31, 30, 29, 28, 27, 27, 26, 25, 25, 24, 24, 23, - 23, 22, 22, 21, 21, 21, 20, 20, 20, 19, 19, 19, 18, 18, 18, 18, 17, 17, 17, - 17, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 14, 14, 14, 13, 13, 13, - 13, 13, 13, 13, 12, 12, 12, 12, 12, 12, 12, 11, 11, 11, 11, 11, 11, 11, 11, - 11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 8, 8, - 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 6, 6, 6, - 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, - 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3, - 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, - 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, - 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0 -}; - -#define WM_VOL_MAX (sizeof(wm_vol) - 1) +#define WM_VOL_MAX 255 #define WM_VOL_MUTE 0x8000 diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 4c24346340f4..5ee468d1aefe 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -351,7 +351,7 @@ enum { struct ichdev { unsigned int ichd; /* ich device number */ unsigned long reg_offset; /* offset to bmaddr */ - u32 *bdbar; /* CPU address (32bit) */ + __le32 *bdbar; /* CPU address (32bit) */ unsigned int bdbar_addr; /* PCI bus address (32bit) */ struct snd_pcm_substream *substream; unsigned int physbuf; /* physical address (32bit) */ @@ -677,7 +677,7 @@ static void snd_intel8x0_ali_codec_write(struct snd_ac97 *ac97, unsigned short r static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ichdev) { int idx; - u32 *bdbar = ichdev->bdbar; + __le32 *bdbar = ichdev->bdbar; unsigned long port = ichdev->reg_offset; iputdword(chip, port + ICH_REG_OFF_BDBAR, ichdev->bdbar_addr); @@ -3143,7 +3143,7 @@ static int snd_intel8x0_create(struct snd_card *card, int_sta_masks = 0; for (i = 0; i < chip->bdbars_count; i++) { ichdev = &chip->ichd[i]; - ichdev->bdbar = ((u32 *)chip->bdbars.area) + + ichdev->bdbar = ((__le32 *)chip->bdbars.area) + (i * ICH_MAX_FRAGS * 2); ichdev->bdbar_addr = chip->bdbars.addr + (i * sizeof(u32) * ICH_MAX_FRAGS * 2); diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 3a4769a97d29..943a726b1c1b 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -168,7 +168,7 @@ enum { ALID_MDMIN, ALID_MDMOUT, ALID_MDMLAST = ALID_MDMOUT }; struct ichdev { unsigned int ichd; /* ich device number */ unsigned long reg_offset; /* offset to bmaddr */ - u32 *bdbar; /* CPU address (32bit) */ + __le32 *bdbar; /* CPU address (32bit) */ unsigned int bdbar_addr; /* PCI bus address (32bit) */ struct snd_pcm_substream *substream; unsigned int physbuf; /* physical address (32bit) */ @@ -395,7 +395,7 @@ static unsigned short snd_intel8x0m_codec_read(struct snd_ac97 *ac97, static void snd_intel8x0m_setup_periods(struct intel8x0m *chip, struct ichdev *ichdev) { int idx; - u32 *bdbar = ichdev->bdbar; + __le32 *bdbar = ichdev->bdbar; unsigned long port = ichdev->reg_offset; iputdword(chip, port + ICH_REG_OFF_BDBAR, ichdev->bdbar_addr); @@ -1217,7 +1217,7 @@ static int snd_intel8x0m_create(struct snd_card *card, int_sta_masks = 0; for (i = 0; i < chip->bdbars_count; i++) { ichdev = &chip->ichd[i]; - ichdev->bdbar = ((u32 *)chip->bdbars.area) + (i * ICH_MAX_FRAGS * 2); + ichdev->bdbar = ((__le32 *)chip->bdbars.area) + (i * ICH_MAX_FRAGS * 2); ichdev->bdbar_addr = chip->bdbars.addr + (i * sizeof(u32) * ICH_MAX_FRAGS * 2); int_sta_masks |= ichdev->int_sta_mask; } diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 4206ba44d8bb..4e189a93f475 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -1326,7 +1326,7 @@ static int snd_korg1212_copy_to(struct snd_pcm_substream *substream, } #endif if (in_kernel) - memcpy((void *)dst, src, size); + memcpy((__force void *)dst, src, size); else if (copy_to_user(dst, src, size)) return -EFAULT; src++; @@ -1365,7 +1365,7 @@ static int snd_korg1212_copy_from(struct snd_pcm_substream *substream, } #endif if (in_kernel) - memcpy((void *)dst, src, size); + memcpy(dst, (__force void *)src, size); else if (copy_from_user(dst, src, size)) return -EFAULT; dst++; diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 9ff600084973..254f24366892 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -369,9 +369,9 @@ static int setup_corb_rirb(struct lola *chip) return err; chip->corb.addr = chip->rb.addr; - chip->corb.buf = (u32 *)chip->rb.area; + chip->corb.buf = (__le32 *)chip->rb.area; chip->rirb.addr = chip->rb.addr + 2048; - chip->rirb.buf = (u32 *)(chip->rb.area + 2048); + chip->rirb.buf = (__le32 *)(chip->rb.area + 2048); /* disable ringbuffer DMAs */ lola_writeb(chip, BAR0, RIRBCTL, 0); diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h index f0b100059efd..bd852fed8bb6 100644 --- a/sound/pci/lola/lola.h +++ b/sound/pci/lola/lola.h @@ -220,7 +220,7 @@ struct lola_bar { /* CORB/RIRB */ struct lola_rb { - u32 *buf; /* CORB/RIRB buffer, 8 byte per each entry */ + __le32 *buf; /* CORB/RIRB buffer, 8 byte per each entry */ dma_addr_t addr; /* physical address of CORB/RIRB buffer */ unsigned short rp, wp; /* read/write pointers */ int cmds; /* number of pending requests */ @@ -275,7 +275,7 @@ struct lola_mixer_array { struct lola_mixer_widget { unsigned int nid; unsigned int caps; - struct lola_mixer_array __user *array; + struct lola_mixer_array __iomem *array; struct lola_mixer_array *array_saved; unsigned int src_stream_outs; unsigned int src_phys_ins; diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index 310b26a756c9..e70276c3ea20 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -316,10 +316,10 @@ static int lola_pcm_hw_free(struct snd_pcm_substream *substream) * set up a BDL entry */ static int setup_bdle(struct snd_pcm_substream *substream, - struct lola_stream *str, u32 **bdlp, + struct lola_stream *str, __le32 **bdlp, int ofs, int size) { - u32 *bdl = *bdlp; + __le32 *bdl = *bdlp; while (size > 0) { dma_addr_t addr; @@ -355,14 +355,14 @@ static int lola_setup_periods(struct lola *chip, struct lola_pcm *pcm, struct snd_pcm_substream *substream, struct lola_stream *str) { - u32 *bdl; + __le32 *bdl; int i, ofs, periods, period_bytes; period_bytes = str->period_bytes; periods = str->bufsize / period_bytes; /* program the initial BDL entries */ - bdl = (u32 *)(pcm->bdl.area + LOLA_BDL_ENTRY_SIZE * str->index); + bdl = (__le32 *)(pcm->bdl.area + LOLA_BDL_ENTRY_SIZE * str->index); ofs = 0; str->frags = 0; for (i = 0; i < periods; i++) { diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 224e942f556d..62962178a9d7 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2103,7 +2103,7 @@ static const u16 minisrc_lpf[MINISRC_LPF_LEN] = { static void snd_m3_assp_init(struct snd_m3 *chip) { unsigned int i; - const u16 *data; + const __le16 *data; /* zero kernel data */ for (i = 0; i < (REV_B_DATA_MEMORY_UNIT_LENGTH * NUM_UNITS_KERNEL_DATA) / 2; i++) @@ -2121,7 +2121,7 @@ static void snd_m3_assp_init(struct snd_m3 *chip) KDATA_DMA_XFER0); /* write kernel into code memory.. */ - data = (const u16 *)chip->assp_kernel_image->data; + data = (const __le16 *)chip->assp_kernel_image->data; for (i = 0 ; i * 2 < chip->assp_kernel_image->size; i++) { snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, REV_B_CODE_MEMORY_BEGIN + i, @@ -2134,7 +2134,7 @@ static void snd_m3_assp_init(struct snd_m3 *chip) * drop it there. It seems that the minisrc doesn't * need vectors, so we won't bother with them.. */ - data = (const u16 *)chip->assp_minisrc_image->data; + data = (const __le16 *)chip->assp_minisrc_image->data; for (i = 0; i * 2 < chip->assp_minisrc_image->size; i++) { snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, 0x400 + i, le16_to_cpu(data[i])); diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index a74f1ad7e7b8..9cd297a42f24 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -182,6 +182,7 @@ static int mixart_set_clock(struct mixart_mgr *mgr, case PIPE_RUNNING: if(rate != 0) break; + /* fall through */ default: if(rate == 0) return 0; /* nothing to do */ diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 8bf2ce32d4a8..71776bfe0485 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -107,7 +107,7 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp, #ifndef __BIG_ENDIAN size /= 4; /* u32 size */ for(i=0; i < size; i++) { - ((u32*)resp->data)[i] = be32_to_cpu(((u32*)resp->data)[i]); + ((u32*)resp->data)[i] = be32_to_cpu(((__be32*)resp->data)[i]); } #endif @@ -519,7 +519,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) /* Traces are text: the swapped msg_data has to be swapped back ! */ int i; for(i=0; i<(resp.size/4); i++) { - (mixart_msg_data)[i] = cpu_to_be32((mixart_msg_data)[i]); + ((__be32*)mixart_msg_data)[i] = cpu_to_be32((mixart_msg_data)[i]); } #endif ((char*)mixart_msg_data)[resp.size - 1] = 0; @@ -540,7 +540,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) dev_err(&mgr->pci->dev, "canceled notification %x !\n", msg); } - /* no break, continue ! */ + /* fall through */ case MSG_TYPE_ANSWER: /* answer or notification to a message we are waiting for*/ mutex_lock(&mgr->msg_lock); diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 5bfd3ac80db5..bc92758de82c 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -73,30 +73,30 @@ static int mixart_wait_nice_for_register_value(struct mixart_mgr *mgr, */ struct snd_mixart_elf32_ehdr { u8 e_ident[16]; - u16 e_type; - u16 e_machine; - u32 e_version; - u32 e_entry; - u32 e_phoff; - u32 e_shoff; - u32 e_flags; - u16 e_ehsize; - u16 e_phentsize; - u16 e_phnum; - u16 e_shentsize; - u16 e_shnum; - u16 e_shstrndx; + __be16 e_type; + __be16 e_machine; + __be32 e_version; + __be32 e_entry; + __be32 e_phoff; + __be32 e_shoff; + __be32 e_flags; + __be16 e_ehsize; + __be16 e_phentsize; + __be16 e_phnum; + __be16 e_shentsize; + __be16 e_shnum; + __be16 e_shstrndx; }; struct snd_mixart_elf32_phdr { - u32 p_type; - u32 p_offset; - u32 p_vaddr; - u32 p_paddr; - u32 p_filesz; - u32 p_memsz; - u32 p_flags; - u32 p_align; + __be32 p_type; + __be32 p_offset; + __be32 p_vaddr; + __be32 p_paddr; + __be32 p_filesz; + __be32 p_memsz; + __be32 p_flags; + __be32 p_align; }; static int mixart_load_elf(struct mixart_mgr *mgr, const struct firmware *dsp ) diff --git a/sound/pci/mixart/mixart_hwdep.h b/sound/pci/mixart/mixart_hwdep.h index 812e288ef2e7..2794cd385b8e 100644 --- a/sound/pci/mixart/mixart_hwdep.h +++ b/sound/pci/mixart/mixart_hwdep.h @@ -26,19 +26,19 @@ #include <sound/hwdep.h> #ifndef readl_be -#define readl_be(x) be32_to_cpu(__raw_readl(x)) +#define readl_be(x) be32_to_cpu((__force __be32)__raw_readl(x)) #endif #ifndef writel_be -#define writel_be(data,addr) __raw_writel(cpu_to_be32(data),addr) +#define writel_be(data,addr) __raw_writel((__force u32)cpu_to_be32(data),addr) #endif #ifndef readl_le -#define readl_le(x) le32_to_cpu(__raw_readl(x)) +#define readl_le(x) le32_to_cpu((__force __le32)__raw_readl(x)) #endif #ifndef writel_le -#define writel_le(data,addr) __raw_writel(cpu_to_le32(data),addr) +#define writel_le(data,addr) __raw_writel((__force u32)cpu_to_le32(data),addr) #endif #define MIXART_MEM(mgr,x) ((mgr)->mem[0].virt + (x)) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 44f3b48d47b1..23017e3bc76c 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -470,10 +470,10 @@ struct snd_riptide { }; struct sgd { /* scatter gather desriptor */ - u32 dwNextLink; - u32 dwSegPtrPhys; - u32 dwSegLen; - u32 dwStat_Ctl; + __le32 dwNextLink; + __le32 dwSegPtrPhys; + __le32 dwSegLen; + __le32 dwStat_Ctl; }; struct pcmhw { /* pcm descriptor */ @@ -1017,7 +1017,7 @@ getsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int *rate) static int setsampleformat(struct cmdif *cif, unsigned char mixer, unsigned char id, - unsigned char channels, unsigned char format) + unsigned char channels, snd_pcm_format_t format) { unsigned char w, ch, sig, order; diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 7fbdb703bfcd..7218f38b59db 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -1433,14 +1433,12 @@ static int snd_sonicvibes_midi(struct sonicvibes *sonic, { struct snd_mpu401 * mpu = rmidi->private_data; struct snd_card *card = sonic->card; - struct snd_rawmidi_str *dir; unsigned int idx; int err; mpu->private_data = sonic; mpu->open_input = snd_sonicvibes_midi_input_open; mpu->close_input = snd_sonicvibes_midi_input_close; - dir = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT]; for (idx = 0; idx < ARRAY_SIZE(snd_sonicvibes_midi_controls); idx++) if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_sonicvibes_midi_controls[idx], sonic))) < 0) return err; diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index cedf13b64803..2f18b1cdc2cd 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -123,7 +123,7 @@ static int snd_trident_probe(struct pci_dev *pci, } else { strcpy(card->shortname, "Trident "); } - strcat(card->shortname, card->driver); + strcat(card->shortname, str); sprintf(card->longname, "%s PCI Audio at 0x%lx, irq %d", card->shortname, trident->port, trident->irq); diff --git a/sound/pci/trident/trident.h b/sound/pci/trident/trident.h index 9624e5937719..2d62c1921255 100644 --- a/sound/pci/trident/trident.h +++ b/sound/pci/trident/trident.h @@ -264,7 +264,7 @@ struct snd_trident_memblk_arg { }; struct snd_trident_tlb { - unsigned int * entries; /* 16k-aligned TLB table */ + __le32 *entries; /* 16k-aligned TLB table */ dma_addr_t entries_dmaaddr; /* 16k-aligned PCI address to TLB table */ unsigned long * shadow_entries; /* shadow entries with virtual addresses */ struct snd_dma_buffer buffer; diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 49c64fae3466..5523e193d556 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3359,7 +3359,7 @@ static int snd_trident_tlb_alloc(struct snd_trident *trident) dev_err(trident->card->dev, "unable to allocate TLB buffer\n"); return -ENOMEM; } - trident->tlb.entries = (unsigned int*)ALIGN((unsigned long)trident->tlb.buffer.area, SNDRV_TRIDENT_MAX_PAGES * 4); + trident->tlb.entries = (__le32 *)ALIGN((unsigned long)trident->tlb.buffer.area, SNDRV_TRIDENT_MAX_PAGES * 4); trident->tlb.entries_dmaaddr = ALIGN(trident->tlb.buffer.addr, SNDRV_TRIDENT_MAX_PAGES * 4); /* allocate shadow TLB page table (virtual addresses) */ trident->tlb.shadow_entries = diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index d4298af6d3ee..c0d0bf44f365 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -275,7 +275,7 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, length >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; length > 0; length--) { - outl(cpu_to_le32(*addr), port); + outl(*addr, port); addr++; } addr = (u32 *)runtime->dma_area; @@ -285,7 +285,7 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, count >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; count > 0; count--) { - outl(cpu_to_le32(*addr), port); + outl(*addr, port); addr++; } @@ -313,7 +313,7 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, length >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; length > 0; length--) - *addr++ = le32_to_cpu(inl(port)); + *addr++ = inl(port); addr = (u32 *)runtime->dma_area; pipe->hw_ptr = 0; } @@ -321,7 +321,7 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, count >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; count > 0; count--) - *addr++ = le32_to_cpu(inl(port)); + *addr++ = inl(port); vx2_release_pseudo_dma(chip); } diff --git a/sound/pci/ymfpci/ymfpci.h b/sound/pci/ymfpci/ymfpci.h index aa9bb065f385..e2fa7e360d79 100644 --- a/sound/pci/ymfpci/ymfpci.h +++ b/sound/pci/ymfpci/ymfpci.h @@ -185,50 +185,50 @@ */ struct snd_ymfpci_playback_bank { - u32 format; - u32 loop_default; - u32 base; /* 32-bit address */ - u32 loop_start; /* 32-bit offset */ - u32 loop_end; /* 32-bit offset */ - u32 loop_frac; /* 8-bit fraction - loop_start */ - u32 delta_end; /* pitch delta end */ - u32 lpfK_end; - u32 eg_gain_end; - u32 left_gain_end; - u32 right_gain_end; - u32 eff1_gain_end; - u32 eff2_gain_end; - u32 eff3_gain_end; - u32 lpfQ; - u32 status; - u32 num_of_frames; - u32 loop_count; - u32 start; - u32 start_frac; - u32 delta; - u32 lpfK; - u32 eg_gain; - u32 left_gain; - u32 right_gain; - u32 eff1_gain; - u32 eff2_gain; - u32 eff3_gain; - u32 lpfD1; - u32 lpfD2; + __le32 format; + __le32 loop_default; + __le32 base; /* 32-bit address */ + __le32 loop_start; /* 32-bit offset */ + __le32 loop_end; /* 32-bit offset */ + __le32 loop_frac; /* 8-bit fraction - loop_start */ + __le32 delta_end; /* pitch delta end */ + __le32 lpfK_end; + __le32 eg_gain_end; + __le32 left_gain_end; + __le32 right_gain_end; + __le32 eff1_gain_end; + __le32 eff2_gain_end; + __le32 eff3_gain_end; + __le32 lpfQ; + __le32 status; + __le32 num_of_frames; + __le32 loop_count; + __le32 start; + __le32 start_frac; + __le32 delta; + __le32 lpfK; + __le32 eg_gain; + __le32 left_gain; + __le32 right_gain; + __le32 eff1_gain; + __le32 eff2_gain; + __le32 eff3_gain; + __le32 lpfD1; + __le32 lpfD2; }; struct snd_ymfpci_capture_bank { - u32 base; /* 32-bit address */ - u32 loop_end; /* 32-bit offset */ - u32 start; /* 32-bit offset */ - u32 num_of_loops; /* counter */ + __le32 base; /* 32-bit address */ + __le32 loop_end; /* 32-bit offset */ + __le32 start; /* 32-bit offset */ + __le32 num_of_loops; /* counter */ }; struct snd_ymfpci_effect_bank { - u32 base; /* 32-bit address */ - u32 loop_end; /* 32-bit offset */ - u32 start; /* 32-bit offset */ - u32 temp; + __le32 base; /* 32-bit address */ + __le32 loop_end; /* 32-bit offset */ + __le32 start; /* 32-bit offset */ + __le32 temp; }; struct snd_ymfpci_pcm; @@ -316,7 +316,7 @@ struct snd_ymfpci { dma_addr_t work_base_addr; struct snd_dma_buffer ac3_tmp_base; - u32 *ctrl_playback; + __le32 *ctrl_playback; struct snd_ymfpci_playback_bank *bank_playback[YDSXG_PLAYBACK_VOICES][2]; struct snd_ymfpci_capture_bank *bank_capture[YDSXG_CAPTURE_VOICES][2]; struct snd_ymfpci_effect_bank *bank_effect[YDSXG_EFFECT_VOICES][2]; diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 6f81396aadc9..a4926fb03991 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -336,7 +336,7 @@ static void snd_ymfpci_pcm_interrupt(struct snd_ymfpci *chip, struct snd_ymfpci_ unsigned int subs = ypcm->substream->number; unsigned int next_bank = 1 - chip->active_bank; struct snd_ymfpci_playback_bank *bank; - u32 volume; + __le32 volume; bank = &voice->bank[next_bank]; volume = cpu_to_le32(chip->pcm_mixer[subs].left << 15); @@ -505,7 +505,7 @@ static void snd_ymfpci_pcm_init_voice(struct snd_ymfpci_pcm *ypcm, unsigned int u32 lpfK = snd_ymfpci_calc_lpfK(runtime->rate); struct snd_ymfpci_playback_bank *bank; unsigned int nbank; - u32 vol_left, vol_right; + __le32 vol_left, vol_right; u8 use_left, use_right; unsigned long flags; @@ -2135,7 +2135,7 @@ static int snd_ymfpci_memalloc(struct snd_ymfpci *chip) chip->bank_base_playback = ptr; chip->bank_base_playback_addr = ptr_addr; - chip->ctrl_playback = (u32 *)ptr; + chip->ctrl_playback = (__le32 *)ptr; chip->ctrl_playback[0] = cpu_to_le32(YDSXG_PLAYBACK_VOICES); ptr += ALIGN(playback_ctrl_size, 0x100); ptr_addr += ALIGN(playback_ctrl_size, 0x100); diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 4a2354b5ae00..98a6863e933c 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -276,7 +276,6 @@ static const struct snd_pcm_ops pdacf_pcm_capture_ops = { .trigger = pdacf_pcm_trigger, .pointer = pdacf_pcm_capture_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c index 8cde40226355..4c4ef1fec69f 100644 --- a/sound/pcmcia/vx/vxp_ops.c +++ b/sound/pcmcia/vx/vxp_ops.c @@ -375,7 +375,7 @@ static void vxp_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, length >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ for (; length > 0; length--) { - outw(cpu_to_le16(*addr), port); + outw(*addr, port); addr++; } addr = (unsigned short *)runtime->dma_area; @@ -385,7 +385,7 @@ static void vxp_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, count >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ for (; count > 0; count--) { - outw(cpu_to_le16(*addr), port); + outw(*addr, port); addr++; } vx_release_pseudo_dma(chip); @@ -417,7 +417,7 @@ static void vxp_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, length >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ for (; length > 0; length--) - *addr++ = le16_to_cpu(inw(port)); + *addr++ = inw(port); addr = (unsigned short *)runtime->dma_area; pipe->hw_ptr = 0; } @@ -425,12 +425,12 @@ static void vxp_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, count >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ for (; count > 1; count--) - *addr++ = le16_to_cpu(inw(port)); + *addr++ = inw(port); /* Disable DMA */ pchip->regDIALOG &= ~VXP_DLG_DMAREAD_SEL_MASK; vx_outb(chip, DIALOG, pchip->regDIALOG); /* Read the last word (16 bits) */ - *addr = le16_to_cpu(inw(port)); + *addr = inw(port); /* Disable 16-bit accesses */ pchip->regDIALOG &= ~VXP_DLG_DMA16_SEL_MASK; vx_outb(chip, DIALOG, pchip->regDIALOG); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 41af6b9cc350..1cf11cf51e1d 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -57,6 +57,7 @@ source "sound/soc/kirkwood/Kconfig" source "sound/soc/img/Kconfig" source "sound/soc/intel/Kconfig" source "sound/soc/mediatek/Kconfig" +source "sound/soc/meson/Kconfig" source "sound/soc/mxs/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/qcom/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 06389a5385d7..62a5f87c3cfc 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -38,6 +38,7 @@ obj-$(CONFIG_SND_SOC) += jz4740/ obj-$(CONFIG_SND_SOC) += img/ obj-$(CONFIG_SND_SOC) += intel/ obj-$(CONFIG_SND_SOC) += mediatek/ +obj-$(CONFIG_SND_SOC) += meson/ obj-$(CONFIG_SND_SOC) += mxs/ obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 6cbf9cf4d1a4..58c1dcb4d255 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -8,6 +8,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_ADAU7002 + select REGULATOR depends on SND_SOC_AMD_ACP && I2C help This option enables machine driver for DA7219 and MAX9835. diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index ccddc6650b9c..8e3275a96a82 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -32,6 +32,8 @@ #include <linux/clk.h> #include <linux/gpio.h> #include <linux/module.h> +#include <linux/regulator/machine.h> +#include <linux/regulator/driver.h> #include <linux/i2c.h> #include <linux/input.h> #include <linux/acpi.h> @@ -148,7 +150,8 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); } @@ -163,7 +166,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -172,13 +175,24 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } -static int cz_dmic_startup(struct snd_pcm_substream *substream) +static int cz_dmic0_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->i2s_instance = I2S_BT_INSTANCE; + return da7219_clk_enable(substream); +} + +static int cz_dmic1_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); machine->i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL0; return da7219_clk_enable(substream); } @@ -197,23 +211,39 @@ static const struct snd_soc_ops cz_max_play_ops = { .shutdown = cz_max_shutdown, }; -static const struct snd_soc_ops cz_dmic_cap_ops = { - .startup = cz_dmic_startup, +static const struct snd_soc_ops cz_dmic0_cap_ops = { + .startup = cz_dmic0_startup, + .shutdown = cz_dmic_shutdown, +}; + +static const struct snd_soc_ops cz_dmic1_cap_ops = { + .startup = cz_dmic1_startup, .shutdown = cz_dmic_shutdown, }; static struct snd_soc_dai_link cz_dai_7219_98357[] = { { - .name = "amd-da7219-play-cap", - .stream_name = "Playback and Capture", + .name = "amd-da7219-play", + .stream_name = "Playback", .platform_name = "acp_audio_dma.0.auto", - .cpu_dai_name = "designware-i2s.3.auto", + .cpu_dai_name = "designware-i2s.1.auto", .codec_dai_name = "da7219-hifi", .codec_name = "i2c-DLGS7219:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, + .ops = &cz_da7219_cap_ops, + }, + { + .name = "amd-da7219-cap", + .stream_name = "Capture", + .platform_name = "acp_audio_dma.0.auto", + .cpu_dai_name = "designware-i2s.2.auto", + .codec_dai_name = "da7219-hifi", + .codec_name = "i2c-DLGS7219:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, .ops = &cz_da7219_cap_ops, }, @@ -221,7 +251,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-max98357-play", .stream_name = "HiFi Playback", .platform_name = "acp_audio_dma.0.auto", - .cpu_dai_name = "designware-i2s.1.auto", + .cpu_dai_name = "designware-i2s.3.auto", .codec_dai_name = "HiFi", .codec_name = "MX98357A:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF @@ -230,8 +260,22 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .ops = &cz_max_play_ops, }, { - .name = "dmic", - .stream_name = "DMIC Capture", + /* C panel DMIC */ + .name = "dmic0", + .stream_name = "DMIC0 Capture", + .platform_name = "acp_audio_dma.0.auto", + .cpu_dai_name = "designware-i2s.3.auto", + .codec_dai_name = "adau7002-hifi", + .codec_name = "ADAU7002:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_dmic0_cap_ops, + }, + { + /* A/B panel DMIC */ + .name = "dmic1", + .stream_name = "DMIC1 Capture", .platform_name = "acp_audio_dma.0.auto", .cpu_dai_name = "designware-i2s.2.auto", .codec_dai_name = "adau7002-hifi", @@ -239,7 +283,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, - .ops = &cz_dmic_cap_ops, + .ops = &cz_dmic1_cap_ops, }, }; @@ -278,11 +322,52 @@ static struct snd_soc_card cz_card = { .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct regulator_consumer_supply acp_da7219_supplies[] = { + REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDIO", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("IOVDD", "ADAU7002:00"), +}; + +static struct regulator_init_data acp_da7219_data = { + .constraints = { + .always_on = 1, + }, + .num_consumer_supplies = ARRAY_SIZE(acp_da7219_supplies), + .consumer_supplies = acp_da7219_supplies, +}; + +static struct regulator_config acp_da7219_cfg = { + .init_data = &acp_da7219_data, +}; + +static struct regulator_ops acp_da7219_ops = { +}; + +static struct regulator_desc acp_da7219_desc = { + .name = "reg-fixed-1.8V", + .type = REGULATOR_VOLTAGE, + .owner = THIS_MODULE, + .ops = &acp_da7219_ops, + .fixed_uV = 1800000, /* 1.8V */ + .n_voltages = 1, +}; + static int cz_probe(struct platform_device *pdev) { int ret; struct snd_soc_card *card; struct acp_platform_info *machine; + struct regulator_dev *rdev; + + acp_da7219_cfg.dev = &pdev->dev; + rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, + &acp_da7219_cfg); + if (IS_ERR(rdev)) { + dev_err(&pdev->dev, "Failed to register regulator: %d\n", + (int)PTR_ERR(rdev)); + return -EINVAL; + } machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 77203841c535..e359938e3d7e 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -224,13 +224,11 @@ static void set_acp_sysmem_dma_descriptors(void __iomem *acp_mmio, switch (asic_type) { case CHIP_STONEY: dmadscr[i].xfer_val |= - BIT(22) | (ACP_DMA_ATTR_SHARED_MEM_TO_DAGB_GARLIC << 16) | (size / 2); break; default: dmadscr[i].xfer_val |= - BIT(22) | (ACP_DMA_ATTR_SHAREDMEM_TO_DAGB_ONION << 16) | (size / 2); } @@ -322,22 +320,87 @@ static void config_acp_dma(void __iomem *acp_mmio, struct audio_substream_data *rtd, u32 asic_type) { + u16 ch_acp_sysmem, ch_acp_i2s; + acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, rtd->pte_offset); + + if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { + ch_acp_sysmem = rtd->ch1; + ch_acp_i2s = rtd->ch2; + } else { + ch_acp_i2s = rtd->ch1; + ch_acp_sysmem = rtd->ch2; + } /* Configure System memory <-> ACP SRAM DMA descriptors */ set_acp_sysmem_dma_descriptors(acp_mmio, rtd->size, rtd->direction, rtd->pte_offset, - rtd->ch1, rtd->sram_bank, + ch_acp_sysmem, rtd->sram_bank, rtd->dma_dscr_idx_1, asic_type); /* Configure ACP SRAM <-> I2S DMA descriptors */ set_acp_to_i2s_dma_descriptors(acp_mmio, rtd->size, rtd->direction, rtd->sram_bank, - rtd->destination, rtd->ch2, + rtd->destination, ch_acp_i2s, rtd->dma_dscr_idx_2, asic_type); } +static void acp_dma_cap_channel_enable(void __iomem *acp_mmio, + u16 cap_channel) +{ + u32 val, ch_reg, imr_reg, res_reg; + + switch (cap_channel) { + case CAP_CHANNEL1: + ch_reg = mmACP_I2SMICSP_RER1; + res_reg = mmACP_I2SMICSP_RCR1; + imr_reg = mmACP_I2SMICSP_IMR1; + break; + case CAP_CHANNEL0: + default: + ch_reg = mmACP_I2SMICSP_RER0; + res_reg = mmACP_I2SMICSP_RCR0; + imr_reg = mmACP_I2SMICSP_IMR0; + break; + } + val = acp_reg_read(acp_mmio, + mmACP_I2S_16BIT_RESOLUTION_EN); + if (val & ACP_I2S_MIC_16BIT_RESOLUTION_EN) { + acp_reg_write(0x0, acp_mmio, ch_reg); + /* Set 16bit resolution on capture */ + acp_reg_write(0x2, acp_mmio, res_reg); + } + val = acp_reg_read(acp_mmio, imr_reg); + val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK; + val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK; + acp_reg_write(val, acp_mmio, imr_reg); + acp_reg_write(0x1, acp_mmio, ch_reg); +} + +static void acp_dma_cap_channel_disable(void __iomem *acp_mmio, + u16 cap_channel) +{ + u32 val, ch_reg, imr_reg; + + switch (cap_channel) { + case CAP_CHANNEL1: + imr_reg = mmACP_I2SMICSP_IMR1; + ch_reg = mmACP_I2SMICSP_RER1; + break; + case CAP_CHANNEL0: + default: + imr_reg = mmACP_I2SMICSP_IMR0; + ch_reg = mmACP_I2SMICSP_RER0; + break; + } + val = acp_reg_read(acp_mmio, imr_reg); + val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK; + val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK; + acp_reg_write(val, acp_mmio, imr_reg); + acp_reg_write(0x0, acp_mmio, ch_reg); +} + /* Start a given DMA channel transfer */ -static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) +static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num, bool is_circular) { u32 dma_ctrl; @@ -356,10 +419,8 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) switch (ch_num) { case ACP_TO_I2S_DMA_CH_NUM: - case ACP_TO_SYSRAM_CH_NUM: case I2S_TO_ACP_DMA_CH_NUM: case ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM: - case ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM: case I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM: dma_ctrl |= ACP_DMA_CNTL_0__DMAChIOCEn_MASK; break; @@ -368,8 +429,11 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) break; } - /* circular for both DMA channel */ - dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + /* enable for ACP to SRAM DMA channel */ + if (is_circular == true) + dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + else + dma_ctrl &= ~ACP_DMA_CNTL_0__Circular_DMA_En_MASK; acp_reg_write(dma_ctrl, acp_mmio, mmACP_DMA_CNTL_0 + ch_num); } @@ -613,6 +677,7 @@ static int acp_deinit(void __iomem *acp_mmio) /* ACP DMA irq handler routine for playback, capture usecases */ static irqreturn_t dma_irq_handler(int irq, void *arg) { + u16 dscr_idx; u32 intr_flag, ext_intr_status; struct audio_drv_data *irq_data; void __iomem *acp_mmio; @@ -644,32 +709,39 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_14) == + CAPTURE_START_DMA_DESCR_CH15) + dscr_idx = CAPTURE_END_DMA_DESCR_CH14; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH14; + config_acp_dma_channel(acp_mmio, ACP_TO_SYSRAM_CH_NUM, dscr_idx, + 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_CH_NUM, false); + snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) != 0) { - valid_irq = true; - acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) << 16, - acp_mmio, mmACP_EXTERNAL_INTR_STAT); - } - if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_10) == + CAPTURE_START_DMA_DESCR_CH11) + dscr_idx = CAPTURE_END_DMA_DESCR_CH10; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH10; + config_acp_dma_channel(acp_mmio, + ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + dscr_idx, 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + false); + snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) { - valid_irq = true; - acp_reg_write((intr_flag & - BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) << 16, - acp_mmio, mmACP_EXTERNAL_INTR_STAT); - } - if (valid_irq) return IRQ_HANDLED; else @@ -773,7 +845,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, if (WARN_ON(!rtd)) return -EINVAL; - rtd->i2s_instance = pinfo->i2s_instance; + if (pinfo) { + rtd->i2s_instance = pinfo->i2s_instance; + rtd->capture_channel = pinfo->capture_channel; + } if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); @@ -841,8 +916,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: rtd->pte_offset = ACP_ST_BT_CAPTURE_PTE_OFFSET; - rtd->ch1 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM; - rtd->ch2 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM; + rtd->ch1 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM; + rtd->ch2 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM; rtd->sram_bank = ACP_SRAM_BANK_4_ADDRESS; rtd->destination = FROM_BLUETOOTH; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10; @@ -851,13 +926,14 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH; rtd->byte_cnt_low_reg_offset = mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_11; adata->capture_i2sbt_stream = substream; break; case I2S_SP_INSTANCE: default: rtd->pte_offset = ACP_CAPTURE_PTE_OFFSET; - rtd->ch1 = ACP_TO_SYSRAM_CH_NUM; - rtd->ch2 = I2S_TO_ACP_DMA_CH_NUM; + rtd->ch1 = I2S_TO_ACP_DMA_CH_NUM; + rtd->ch2 = ACP_TO_SYSRAM_CH_NUM; switch (adata->asic_type) { case CHIP_STONEY: rtd->pte_offset = ACP_ST_CAPTURE_PTE_OFFSET; @@ -874,6 +950,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, mmACP_I2S_RECEIVED_BYTE_CNT_HIGH; rtd->byte_cnt_low_reg_offset = mmACP_I2S_RECEIVED_BYTE_CNT_LOW; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_15; adata->capture_i2ssp_stream = substream; } } @@ -927,6 +1004,8 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) u32 buffersize; u32 pos = 0; u64 bytescount = 0; + u16 dscr; + u32 period_bytes, delay; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; @@ -934,12 +1013,25 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) if (!rtd) return -EINVAL; - buffersize = frames_to_bytes(runtime, runtime->buffer_size); - bytescount = acp_get_byte_count(rtd); - - if (bytescount > rtd->bytescount) - bytescount -= rtd->bytescount; - pos = do_div(bytescount, buffersize); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + period_bytes = frames_to_bytes(runtime, runtime->period_size); + dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr); + if (dscr == rtd->dma_dscr_idx_1) + pos = period_bytes; + else + pos = 0; + bytescount = acp_get_byte_count(rtd); + if (bytescount > rtd->bytescount) + bytescount -= rtd->bytescount; + delay = do_div(bytescount, period_bytes); + runtime->delay = bytes_to_frames(runtime, delay); + } else { + buffersize = frames_to_bytes(runtime, runtime->buffer_size); + bytescount = acp_get_byte_count(rtd); + if (bytescount > rtd->bytescount) + bytescount -= rtd->bytescount; + pos = do_div(bytescount, buffersize); + } return bytes_to_frames(runtime, pos); } @@ -953,16 +1045,24 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + u16 ch_acp_sysmem, ch_acp_i2s; if (!rtd) return -EINVAL; + if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { + ch_acp_sysmem = rtd->ch1; + ch_acp_i2s = rtd->ch2; + } else { + ch_acp_i2s = rtd->ch1; + ch_acp_sysmem = rtd->ch2; + } config_acp_dma_channel(rtd->acp_mmio, - rtd->ch1, + ch_acp_sysmem, rtd->dma_dscr_idx_1, NUM_DSCRS_PER_CHANNEL, 0); config_acp_dma_channel(rtd->acp_mmio, - rtd->ch2, + ch_acp_i2s, rtd->dma_dscr_idx_2, NUM_DSCRS_PER_CHANNEL, 0); return 0; @@ -971,7 +1071,6 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) { int ret; - u64 bytescount = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; @@ -982,37 +1081,32 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - bytescount = acp_get_byte_count(rtd); - if (rtd->bytescount == 0) - rtd->bytescount = bytescount; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - acp_dma_start(rtd->acp_mmio, rtd->ch1); - acp_dma_start(rtd->acp_mmio, rtd->ch2); + rtd->bytescount = acp_get_byte_count(rtd); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (rtd->capture_channel == CAP_CHANNEL0) { + acp_dma_cap_channel_disable(rtd->acp_mmio, + CAP_CHANNEL1); + acp_dma_cap_channel_enable(rtd->acp_mmio, + CAP_CHANNEL0); + } + if (rtd->capture_channel == CAP_CHANNEL1) { + acp_dma_cap_channel_disable(rtd->acp_mmio, + CAP_CHANNEL0); + acp_dma_cap_channel_enable(rtd->acp_mmio, + CAP_CHANNEL1); + } + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); } else { - acp_dma_start(rtd->acp_mmio, rtd->ch2); - acp_dma_start(rtd->acp_mmio, rtd->ch1); + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); + acp_dma_start(rtd->acp_mmio, rtd->ch2, true); } ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - /* For playback, non circular dma should be stopped first - * i.e Sysram to acp dma transfer channel(rtd->ch1) should be - * stopped before stopping cirular dma which is acp sram to i2s - * fifo dma transfer channel(rtd->ch2). Where as in Capture - * scenario, i2s fifo to acp sram dma channel(rtd->ch2) stopped - * first before stopping acp sram to sysram which is circular - * dma(rtd->ch1). - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - acp_dma_stop(rtd->acp_mmio, rtd->ch1); - ret = acp_dma_stop(rtd->acp_mmio, rtd->ch2); - } else { - acp_dma_stop(rtd->acp_mmio, rtd->ch2); - ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); - } - rtd->bytescount = 0; + acp_dma_stop(rtd->acp_mmio, rtd->ch2); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); break; default: ret = -EINVAL; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 9cd3e96c84d4..be3963e8f4fa 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -55,6 +55,8 @@ #define I2S_SP_INSTANCE 0x01 #define I2S_BT_INSTANCE 0x02 +#define CAP_CHANNEL0 0x00 +#define CAP_CHANNEL1 0x01 #define ACP_TILE_ON_MASK 0x03 #define ACP_TILE_OFF_MASK 0x02 @@ -72,16 +74,16 @@ #define ACP_TO_I2S_DMA_CH_NUM 13 /* Capture DMA channels */ -#define ACP_TO_SYSRAM_CH_NUM 14 -#define I2S_TO_ACP_DMA_CH_NUM 15 +#define I2S_TO_ACP_DMA_CH_NUM 14 +#define ACP_TO_SYSRAM_CH_NUM 15 /* Playback DMA Channels for I2S BT instance */ #define SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM 8 #define ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM 9 /* Capture DMA Channels for I2S BT Instance */ -#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 10 -#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 11 +#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 10 +#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 11 #define NUM_DSCRS_PER_CHANNEL 2 @@ -125,6 +127,7 @@ struct audio_substream_data { unsigned int order; u16 num_of_pages; u16 i2s_instance; + u16 capture_channel; u16 direction; u16 ch1; u16 ch2; @@ -135,6 +138,7 @@ struct audio_substream_data { u32 sram_bank; u32 byte_cnt_high_reg_offset; u32 byte_cnt_low_reg_offset; + u32 dma_curr_dscr; uint64_t size; u64 bytescount; void __iomem *acp_mmio; @@ -155,6 +159,7 @@ struct audio_drv_data { */ struct acp_platform_info { u16 i2s_instance; + u16 capture_channel; }; union acp_dma_count { diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 5d3b5af9fd92..d88c1d995036 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -206,7 +206,6 @@ struct atmel_i2s_dev { struct regmap *regmap; struct clk *pclk; struct clk *gclk; - struct clk *aclk; struct snd_dmaengine_dai_dma_data playback; struct snd_dmaengine_dai_dma_data capture; unsigned int fmt; @@ -303,7 +302,7 @@ static int atmel_i2s_get_gck_param(struct atmel_i2s_dev *dev, int fs) { int i, best; - if (!dev->gclk || !dev->aclk) { + if (!dev->gclk) { dev_err(dev->dev, "cannot generate the I2S Master Clock\n"); return -EINVAL; } @@ -421,7 +420,7 @@ static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev, bool enabled) { unsigned int mr, mr_mask; - unsigned long aclk_rate; + unsigned long gclk_rate; int ret; mr = 0; @@ -445,35 +444,18 @@ static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev, /* Disable/unprepare the PMC generated clock. */ clk_disable_unprepare(dev->gclk); - /* Disable/unprepare the PLL audio clock. */ - clk_disable_unprepare(dev->aclk); return 0; } if (!dev->gck_param) return -EINVAL; - aclk_rate = dev->gck_param->mck * (dev->gck_param->imckdiv + 1); + gclk_rate = dev->gck_param->mck * (dev->gck_param->imckdiv + 1); - /* Fist change the PLL audio clock frequency ... */ - ret = clk_set_rate(dev->aclk, aclk_rate); + ret = clk_set_rate(dev->gclk, gclk_rate); if (ret) return ret; - /* - * ... then set the PMC generated clock rate to the very same frequency - * to set the gclk parent to aclk. - */ - ret = clk_set_rate(dev->gclk, aclk_rate); - if (ret) - return ret; - - /* Prepare and enable the PLL audio clock first ... */ - ret = clk_prepare_enable(dev->aclk); - if (ret) - return ret; - - /* ... then prepare and enable the PMC generated clock. */ ret = clk_prepare_enable(dev->gclk); if (ret) return ret; @@ -668,28 +650,14 @@ static int atmel_i2s_probe(struct platform_device *pdev) return err; } - /* Get audio clocks to generate the I2S Master Clock (I2S_MCK) */ - dev->aclk = devm_clk_get(&pdev->dev, "aclk"); + /* Get audio clock to generate the I2S Master Clock (I2S_MCK) */ dev->gclk = devm_clk_get(&pdev->dev, "gclk"); - if (IS_ERR(dev->aclk) && IS_ERR(dev->gclk)) { - if (PTR_ERR(dev->aclk) == -EPROBE_DEFER || - PTR_ERR(dev->gclk) == -EPROBE_DEFER) + if (IS_ERR(dev->gclk)) { + if (PTR_ERR(dev->gclk) == -EPROBE_DEFER) return -EPROBE_DEFER; /* Master Mode not supported */ - dev->aclk = NULL; dev->gclk = NULL; - } else if (IS_ERR(dev->gclk)) { - err = PTR_ERR(dev->gclk); - dev_err(&pdev->dev, - "failed to get the PMC generated clock: %d\n", err); - return err; - } else if (IS_ERR(dev->aclk)) { - err = PTR_ERR(dev->aclk); - dev_err(&pdev->dev, - "failed to get the PLL audio clock: %d\n", err); - return err; } - dev->dev = &pdev->dev; dev->regmap = regmap; platform_set_drvdata(pdev, dev); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 63cf62e9c9aa..efb095dbcd71 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -74,12 +74,12 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7219 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C - select SND_SOC_DIO2125 select SND_SOC_DMIC if GPIOLIB select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C select SND_SOC_ES7134 + select SND_SOC_ES7241 select SND_SOC_GTM601 select SND_SOC_HDAC_HDMI select SND_SOC_ICS43432 @@ -141,8 +141,10 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5668 if I2C select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER + select SND_SOC_RT5682 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE + select SND_SOC_SIMPLE_AMPLIFIER select SND_SOC_SIRF_AUDIO_CODEC select SND_SOC_SPDIF select SND_SOC_SSM2305 @@ -572,10 +574,6 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_DIO2125 - tristate "Dioo DIO2125 Amplifier" - select GPIOLIB - config SND_SOC_DMIC tristate @@ -588,6 +586,9 @@ config SND_SOC_HDMI_CODEC config SND_SOC_ES7134 tristate "Everest Semi ES7134 CODEC" +config SND_SOC_ES7241 + tristate "Everest Semi ES7241 CODEC" + config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C @@ -778,6 +779,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5668=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y + default y if SND_SOC_RT5682=y default y if SND_SOC_RT1305=y default m if SND_SOC_RT5514=m default m if SND_SOC_RT5616=m @@ -791,6 +793,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5668=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m + default m if SND_SOC_RT5682=m default m if SND_SOC_RT1305=m config SND_SOC_RL6347A @@ -871,6 +874,9 @@ config SND_SOC_RT5677_SPI tristate default SND_SOC_RT5677 && SPI +config SND_SOC_RT5682 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate "Freescale SGTL5000 CODEC" @@ -891,6 +897,10 @@ config SND_SOC_SIGMADSP_REGMAP tristate select SND_SOC_SIGMADSP +config SND_SOC_SIMPLE_AMPLIFIER + tristate "Simple Audio Amplifier" + select GPIOLIB + config SND_SOC_SIRF_AUDIO_CODEC tristate "SiRF SoC internal audio codec" select REGMAP_MMIO @@ -953,8 +963,11 @@ config SND_SOC_TAS5086 depends on I2C config SND_SOC_TAS571X - tristate "Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 power amplifiers" + tristate "Texas Instruments TAS571x power amplifiers" depends on I2C + help + Enable support for Texas Instruments TAS5707, TAS5711, TAS5717, + TAS5719 and TAS5721 power amplifiers config SND_SOC_TAS5720 tristate "Texas Instruments TAS5720 Mono Audio amplifier" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e023fdf85221..7ae7c85e8219 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -71,6 +71,7 @@ snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o +snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o @@ -146,6 +147,7 @@ snd-soc-rt5668-objs := rt5668.o snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o +snd-soc-rt5682-objs := rt5682.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -249,9 +251,9 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o snd-soc-zx-aud96p22-objs := zx_aud96p22.o # Amp -snd-soc-dio2125-objs := dio2125.o snd-soc-max9877-objs := max9877.o snd-soc-max98504-objs := max98504.o +snd-soc-simple-amplifier-objs := simple-amplifier.o snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-tas2552-objs := tas2552.o @@ -329,6 +331,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o +obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o @@ -405,6 +408,7 @@ obj-$(CONFIG_SND_SOC_RT5668) += snd-soc-rt5668.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o +obj-$(CONFIG_SND_SOC_RT5682) += snd-soc-rt5682.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o @@ -507,7 +511,7 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o obj-$(CONFIG_SND_SOC_ZX_AUD96P22) += snd-soc-zx-aud96p22.o # Amp -obj-$(CONFIG_SND_SOC_DIO2125) += snd-soc-dio2125.o obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_MAX98504) += snd-soc-max98504.o +obj-$(CONFIG_SND_SOC_SIMPLE_AMPLIFIER) += snd-soc-simple-amplifier.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index ae41edd1c406..57169b8ff14e 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -299,6 +299,7 @@ static const struct snd_soc_dapm_route adau17x1_dsp_dapm_routes[] = { { "DSP", NULL, "Left Decimator" }, { "DSP", NULL, "Right Decimator" }, + { "DSP", NULL, "Playback" }, }; static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index db21ecbe0762..8b9ca7e7a682 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -648,6 +648,7 @@ static int adav80x_set_pll(struct snd_soc_component *component, int pll_id, pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV; break; } + /* fall through */ default: return -EINVAL; } diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 31ec0ba2e639..299ada4dfaa0 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -558,7 +558,7 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev) } #endif /* CONFIG_PM */ -struct snd_soc_component_driver soc_codec_dev_ak4458 = { +static const struct snd_soc_component_driver soc_codec_dev_ak4458 = { .probe = ak4458_probe, .remove = ak4458_remove, .controls = ak4458_snd_controls, diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index b7ee13406d93..2fa83a1a84cf 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -1,13 +1,8 @@ -/* - * ak4554.c - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// ak4554.c +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> #include <linux/module.h> #include <sound/soc.h> @@ -97,6 +92,6 @@ static struct platform_driver ak4554_driver = { }; module_platform_driver(ak4554_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SoC AK4554 driver"); MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>"); diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 8523ff9351cf..c1181a20714d 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -1,18 +1,14 @@ -/* - * ak4613.c -- Asahi Kasei ALSA Soc Audio driver - * - * Copyright (C) 2015 Renesas Electronics Corporation - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * Based on ak4642.c by Kuninori Morimoto - * Based on wm8731.c by Richard Purdie - * Based on ak4535.c by Richard Purdie - * Based on wm8753.c by Liam Girdwood - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ak4613.c -- Asahi Kasei ALSA Soc Audio driver +// +// Copyright (C) 2015 Renesas Electronics Corporation +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> +// +// Based on ak4642.c by Kuninori Morimoto +// Based on wm8731.c by Richard Purdie +// Based on ak4535.c by Richard Purdie +// Based on wm8753.c by Liam Girdwood #include <linux/clk.h> #include <linux/delay.h> diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 605055964529..353237025514 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -1,17 +1,13 @@ -/* - * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Kuninori Morimoto <morimoto.kuninori@renesas.com> - * - * Based on wm8731.c by Richard Purdie - * Based on ak4535.c by Richard Purdie - * Based on wm8753.c by Liam Girdwood - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Kuninori Morimoto <morimoto.kuninori@renesas.com> +// +// Based on wm8731.c by Richard Purdie +// Based on ak4535.c by Richard Purdie +// Based on wm8753.c by Liam Girdwood /* ** CAUTION ** * @@ -709,4 +705,4 @@ module_i2c_driver(ak4642_i2c_driver); MODULE_DESCRIPTION("Soc AK4642 driver"); MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index f4ed5cc40661..448bb90c9c8e 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -322,13 +322,13 @@ static int __maybe_unused ak5558_runtime_resume(struct device *dev) return regcache_sync(ak5558->regmap); } -const struct dev_pm_ops ak5558_pm = { +static const struct dev_pm_ops ak5558_pm = { SET_RUNTIME_PM_OPS(ak5558_runtime_suspend, ak5558_runtime_resume, NULL) SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) }; -struct snd_soc_component_driver soc_codec_dev_ak5558 = { +static const struct snd_soc_component_driver soc_codec_dev_ak5558 = { .probe = ak5558_probe, .remove = ak5558_remove, .controls = ak5558_snd_controls, diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 2a7a4168c072..3c266eeb89bf 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -219,7 +219,7 @@ static bool cs4270_reg_is_volatile(struct device *dev, unsigned int reg) { /* Unreadable registers are considered volatile */ if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) - return 1; + return true; return reg == CS4270_CHIPID; } diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 196e9c343aeb..45e50fe3bf25 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -235,6 +235,9 @@ ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP2", 1), +WM_ADSP_FW_CONTROL("DSP3", 2), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); @@ -1283,6 +1286,12 @@ static int cs47l24_probe(struct platform_device *pdev) return ret; } + ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1); + if (ret != 0) + dev_warn(&pdev->dev, + "Failed to set compressed IRQ as a wake source: %d\n", + ret); + arizona_init_common(arizona); ret = arizona_init_vol_limit(arizona); @@ -1306,6 +1315,7 @@ static int cs47l24_probe(struct platform_device *pdev) err_spk_irqs: arizona_free_spk_irqs(arizona); err_dsp_irq: + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, cs47l24); return ret; @@ -1323,6 +1333,7 @@ static int cs47l24_remove(struct platform_device *pdev) arizona_free_spk_irqs(arizona); + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, cs47l24); return 0; diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 07dd33b09596..ab174b5114dc 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -362,8 +362,27 @@ static int cx20442_component_probe(struct snd_soc_component *component) return -ENOMEM; cx20442->por = regulator_get(component->dev, "POR"); - if (IS_ERR(cx20442->por)) - dev_warn(component->dev, "failed to get the regulator"); + if (IS_ERR(cx20442->por)) { + int err = PTR_ERR(cx20442->por); + + dev_warn(component->dev, "failed to get POR supply (%d)", err); + /* + * When running on a non-dt platform and requested regulator + * is not available, regulator_get() never returns + * -EPROBE_DEFER as it is not able to justify if the regulator + * may still appear later. On the other hand, the board can + * still set full constraints flag at late_initcall in order + * to instruct regulator_get() to return a dummy one if + * sufficient. Hence, if we get -ENODEV here, let's convert + * it to -EPROBE_DEFER and wait for the board to decide or + * let Deferred Probe infrastructure handle this error. + */ + if (err == -ENODEV) + err = -EPROBE_DEFER; + kfree(cx20442); + return err; + } + cx20442->tty = NULL; snd_soc_component_set_drvdata(component, cx20442); diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a664111b7184..e172913d04a4 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1,19 +1,14 @@ -/* - * DA7210 ALSA Soc codec driver - * - * Copyright (c) 2009 Dialog Semiconductor - * Written by David Chen <Dajun.chen@diasemi.com> - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Cleanups by Kuninori Morimoto <morimoto.kuninori@renesas.com> - * - * Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// DA7210 ALSA Soc codec driver +// +// Copyright (c) 2009 Dialog Semiconductor +// Written by David Chen <Dajun.chen@diasemi.com> +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Cleanups by Kuninori Morimoto <morimoto.kuninori@renesas.com> +// +// Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S #include <linux/delay.h> #include <linux/i2c.h> diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 54cb5f24969f..92d006a5283e 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1140,9 +1140,9 @@ static bool da7213_volatile_register(struct device *dev, unsigned int reg) case DA7213_ALC_OFFSET_AUTO_M_R: case DA7213_ALC_OFFSET_AUTO_U_R: case DA7213_ALC_CIC_OP_LVL_DATA: - return 1; + return true; default: - return 0; + return false; } } diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index a49ab751a036..2c7d5088e6f2 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -59,6 +59,7 @@ static void da7219_aad_btn_det_work(struct work_struct *work) container_of(work, struct da7219_aad_priv, btn_det_work); struct snd_soc_component *component = da7219_aad->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 statusa, micbias_ctrl; bool micbias_up = false; int retries = 0; @@ -86,6 +87,8 @@ static void da7219_aad_btn_det_work(struct work_struct *work) if (retries >= DA7219_AAD_MICBIAS_CHK_RETRIES) dev_warn(component->dev, "Mic bias status check timed out"); + da7219->micbias_on_event = true; + /* * Mic bias pulse required to enable mic, must be done before enabling * button detection to prevent erroneous button readings. @@ -439,6 +442,8 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) snd_soc_component_update_bits(component, DA7219_ACCDET_CONFIG_1, DA7219_BUTTON_CONFIG_MASK, 0); + da7219->micbias_on_event = false; + /* Disable mic bias */ snd_soc_dapm_disable_pin(dapm, "Mic Bias"); snd_soc_dapm_sync(dapm); diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 980a6a8bf56d..e46e9f4bc994 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -768,6 +768,30 @@ static const struct snd_kcontrol_new da7219_st_out_filtr_mix_controls[] = { * DAPM Events */ +static int da7219_mic_pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (da7219->micbias_on_event) { + /* + * Delay only for first capture after bias enabled to + * avoid possible DC offset related noise. + */ + da7219->micbias_on_event = false; + msleep(da7219->mic_pga_delay); + } + break; + default: + break; + } + + return 0; +} + static int da7219_dai_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -937,12 +961,12 @@ static const struct snd_soc_dapm_widget da7219_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC"), /* Input PGAs */ - SND_SOC_DAPM_PGA("Mic PGA", DA7219_MIC_1_CTRL, - DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT, - NULL, 0), - SND_SOC_DAPM_PGA("Mixin PGA", DA7219_MIXIN_L_CTRL, - DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT, - NULL, 0), + SND_SOC_DAPM_PGA_E("Mic PGA", DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0, da7219_mic_pga_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("Mixin PGA", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0, da7219_settling_event, SND_SOC_DAPM_POST_PMU), /* Input Filters */ SND_SOC_DAPM_ADC("ADC", NULL, DA7219_ADC_L_CTRL, DA7219_ADC_L_EN_SHIFT, @@ -1847,6 +1871,14 @@ static void da7219_handle_pdata(struct snd_soc_component *component) snd_soc_component_write(component, DA7219_MICBIAS_CTRL, micbias_lvl); + /* + * Calculate delay required to compensate for DC offset in + * Mic PGA, based on Mic Bias voltage. + */ + da7219->mic_pga_delay = DA7219_MIC_PGA_BASE_DELAY + + (pdata->micbias_lvl * + DA7219_MIC_PGA_OFFSET_DELAY); + /* Mic */ switch (pdata->mic_amp_in_sel) { case DA7219_MIC_AMP_IN_SEL_DIFF: @@ -2143,9 +2175,9 @@ static bool da7219_volatile_register(struct device *dev, unsigned int reg) case DA7219_ACCDET_IRQ_EVENT_B: case DA7219_ACCDET_CONFIG_8: case DA7219_SYSTEM_STATUS: - return 1; + return true; default: - return 0; + return false; } } diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 1b00023e33cd..3a006862f0e7 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -781,8 +781,10 @@ #define DA7219_SYS_STAT_CHECK_DELAY 50 /* Power up/down Delays */ -#define DA7219_SETTLING_DELAY 40 -#define DA7219_MIN_GAIN_DELAY 30 +#define DA7219_SETTLING_DELAY 40 +#define DA7219_MIN_GAIN_DELAY 30 +#define DA7219_MIC_PGA_BASE_DELAY 100 +#define DA7219_MIC_PGA_OFFSET_DELAY 40 enum da7219_clk_src { DA7219_CLKSRC_MCLK = 0, @@ -828,6 +830,8 @@ struct da7219_priv { bool master; bool alc_en; + bool micbias_on_event; + unsigned int mic_pga_delay; u8 gain_ramp_ctrl; }; diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index afdf90c78884..f6a7bf9560e7 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1041,9 +1041,9 @@ static bool da9055_volatile_register(struct device *dev, case DA9055_HP_R_GAIN_STATUS: case DA9055_LINE_GAIN_STATUS: case DA9055_ALC_CIC_OP_LVL_DATA: - return 1; + return true; default: - return 0; + return false; } } diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index 58515bb1a303..6d7bca7b78ca 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -17,6 +17,7 @@ * in the file called COPYING. */ +#include <linux/of_platform.h> #include <linux/module.h> #include <sound/soc.h> @@ -24,6 +25,82 @@ * The everest 7134 is a very simple DA converter with no register */ +struct es7134_clock_mode { + unsigned int rate_min; + unsigned int rate_max; + unsigned int *mclk_fs; + unsigned int mclk_fs_num; +}; + +struct es7134_chip { + struct snd_soc_dai_driver *dai_drv; + const struct es7134_clock_mode *modes; + unsigned int mode_num; + const struct snd_soc_dapm_widget *extra_widgets; + unsigned int extra_widget_num; + const struct snd_soc_dapm_route *extra_routes; + unsigned int extra_route_num; +}; + +struct es7134_data { + unsigned int mclk; + const struct es7134_chip *chip; +}; + +static int es7134_check_mclk(struct snd_soc_dai *dai, + struct es7134_data *priv, + unsigned int rate) +{ + unsigned int mfs = priv->mclk / rate; + int i, j; + + for (i = 0; i < priv->chip->mode_num; i++) { + const struct es7134_clock_mode *mode = &priv->chip->modes[i]; + + if (rate < mode->rate_min || rate > mode->rate_max) + continue; + + for (j = 0; j < mode->mclk_fs_num; j++) { + if (mode->mclk_fs[j] == mfs) + return 0; + } + + dev_err(dai->dev, "unsupported mclk_fs %u for rate %u\n", + mfs, rate); + return -EINVAL; + } + + /* should not happen */ + dev_err(dai->dev, "unsupported rate: %u\n", rate); + return -EINVAL; +} + +static int es7134_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct es7134_data *priv = snd_soc_dai_get_drvdata(dai); + + /* mclk has not been provided, assume it is OK */ + if (!priv->mclk) + return 0; + + return es7134_check_mclk(dai, priv, params_rate(params)); +} + +static int es7134_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct es7134_data *priv = snd_soc_dai_get_drvdata(dai); + + if (dir == SND_SOC_CLOCK_IN && clk_id == 0) { + priv->mclk = freq; + return 0; + } + + return -ENOTSUPP; +} + static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { fmt &= (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK | @@ -38,8 +115,38 @@ static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } +static int es7134_component_probe(struct snd_soc_component *c) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(c); + struct es7134_data *priv = snd_soc_component_get_drvdata(c); + const struct es7134_chip *chip = priv->chip; + int ret; + + if (chip->extra_widget_num) { + ret = snd_soc_dapm_new_controls(dapm, chip->extra_widgets, + chip->extra_widget_num); + if (ret) { + dev_err(c->dev, "failed to add extra widgets\n"); + return ret; + } + } + + if (chip->extra_route_num) { + ret = snd_soc_dapm_add_routes(dapm, chip->extra_routes, + chip->extra_route_num); + if (ret) { + dev_err(c->dev, "failed to add extra routes\n"); + return ret; + } + } + + return 0; +} + static const struct snd_soc_dai_ops es7134_dai_ops = { .set_fmt = es7134_set_fmt, + .hw_params = es7134_hw_params, + .set_sysclk = es7134_set_sysclk, }; static struct snd_soc_dai_driver es7134_dai = { @@ -48,7 +155,11 @@ static struct snd_soc_dai_driver es7134_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = (SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE | @@ -58,18 +169,56 @@ static struct snd_soc_dai_driver es7134_dai = { .ops = &es7134_dai_ops, }; +static const struct es7134_clock_mode es7134_modes[] = { + { + /* Single speed mode */ + .rate_min = 8000, + .rate_max = 50000, + .mclk_fs = (unsigned int[]) { 256, 384, 512, 768, 1024 }, + .mclk_fs_num = 5, + }, { + /* Double speed mode */ + .rate_min = 84000, + .rate_max = 100000, + .mclk_fs = (unsigned int[]) { 128, 192, 256, 384, 512 }, + .mclk_fs_num = 5, + }, { + /* Quad speed mode */ + .rate_min = 167000, + .rate_max = 192000, + .mclk_fs = (unsigned int[]) { 128, 192, 256 }, + .mclk_fs_num = 3, + }, +}; + +/* Digital I/O are also supplied by VDD on the es7134 */ +static const struct snd_soc_dapm_route es7134_extra_routes[] = { + { "Playback", NULL, "VDD", } +}; + +static const struct es7134_chip es7134_chip = { + .dai_drv = &es7134_dai, + .modes = es7134_modes, + .mode_num = ARRAY_SIZE(es7134_modes), + .extra_routes = es7134_extra_routes, + .extra_route_num = ARRAY_SIZE(es7134_extra_routes), +}; + static const struct snd_soc_dapm_widget es7134_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("AOUTL"), SND_SOC_DAPM_OUTPUT("AOUTR"), SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDD", 0, 0), }; static const struct snd_soc_dapm_route es7134_dapm_routes[] = { { "AOUTL", NULL, "DAC" }, { "AOUTR", NULL, "DAC" }, + { "DAC", NULL, "VDD" }, }; static const struct snd_soc_component_driver es7134_component_driver = { + .probe = es7134_component_probe, .dapm_widgets = es7134_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(es7134_dapm_widgets), .dapm_routes = es7134_dapm_routes, @@ -80,17 +229,87 @@ static const struct snd_soc_component_driver es7134_component_driver = { .non_legacy_dai_naming = 1, }; +static struct snd_soc_dai_driver es7154_dai = { + .name = "es7154-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S18_3LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &es7134_dai_ops, +}; + +static const struct es7134_clock_mode es7154_modes[] = { + { + /* Single speed mode */ + .rate_min = 8000, + .rate_max = 50000, + .mclk_fs = (unsigned int[]) { 32, 64, 128, 192, 256, + 384, 512, 768, 1024 }, + .mclk_fs_num = 9, + }, { + /* Double speed mode */ + .rate_min = 84000, + .rate_max = 100000, + .mclk_fs = (unsigned int[]) { 128, 192, 256, 384, 512, + 768, 1024}, + .mclk_fs_num = 7, + } +}; + +/* Es7154 has a separate supply for digital I/O */ +static const struct snd_soc_dapm_widget es7154_extra_widgets[] = { + SND_SOC_DAPM_REGULATOR_SUPPLY("PVDD", 0, 0), +}; + +static const struct snd_soc_dapm_route es7154_extra_routes[] = { + { "Playback", NULL, "PVDD", } +}; + +static const struct es7134_chip es7154_chip = { + .dai_drv = &es7154_dai, + .modes = es7154_modes, + .mode_num = ARRAY_SIZE(es7154_modes), + .extra_routes = es7154_extra_routes, + .extra_route_num = ARRAY_SIZE(es7154_extra_routes), + .extra_widgets = es7154_extra_widgets, + .extra_widget_num = ARRAY_SIZE(es7154_extra_widgets), +}; + static int es7134_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; + struct es7134_data *priv; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + priv->chip = of_device_get_match_data(dev); + if (!priv->chip) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + return devm_snd_soc_register_component(&pdev->dev, &es7134_component_driver, - &es7134_dai, 1); + priv->chip->dai_drv, 1); } #ifdef CONFIG_OF static const struct of_device_id es7134_ids[] = { - { .compatible = "everest,es7134", }, - { .compatible = "everest,es7144", }, + { .compatible = "everest,es7134", .data = &es7134_chip }, + { .compatible = "everest,es7144", .data = &es7134_chip }, + { .compatible = "everest,es7154", .data = &es7154_chip }, { } }; MODULE_DEVICE_TABLE(of, es7134_ids); diff --git a/sound/soc/codecs/es7241.c b/sound/soc/codecs/es7241.c new file mode 100644 index 000000000000..87991bd4acef --- /dev/null +++ b/sound/soc/codecs/es7241.c @@ -0,0 +1,322 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/gpio/consumer.h> +#include <linux/of_platform.h> +#include <linux/module.h> +#include <sound/soc.h> + +struct es7241_clock_mode { + unsigned int rate_min; + unsigned int rate_max; + unsigned int *slv_mfs; + unsigned int slv_mfs_num; + unsigned int mst_mfs; + unsigned int mst_m0:1; + unsigned int mst_m1:1; +}; + +struct es7241_chip { + const struct es7241_clock_mode *modes; + unsigned int mode_num; +}; + +struct es7241_data { + struct gpio_desc *reset; + struct gpio_desc *m0; + struct gpio_desc *m1; + unsigned int fmt; + unsigned int mclk; + bool is_slave; + const struct es7241_chip *chip; +}; + +static void es7241_set_mode(struct es7241_data *priv, int m0, int m1) +{ + /* put the device in reset */ + gpiod_set_value_cansleep(priv->reset, 0); + + /* set the mode */ + gpiod_set_value_cansleep(priv->m0, m0); + gpiod_set_value_cansleep(priv->m1, m1); + + /* take the device out of reset - datasheet does not specify a delay */ + gpiod_set_value_cansleep(priv->reset, 1); +} + +static int es7241_set_slave_mode(struct es7241_data *priv, + const struct es7241_clock_mode *mode, + unsigned int mfs) +{ + int j; + + if (!mfs) + goto out_ok; + + for (j = 0; j < mode->slv_mfs_num; j++) { + if (mode->slv_mfs[j] == mfs) + goto out_ok; + } + + return -EINVAL; + +out_ok: + es7241_set_mode(priv, 1, 1); + return 0; +} + +static int es7241_set_master_mode(struct es7241_data *priv, + const struct es7241_clock_mode *mode, + unsigned int mfs) +{ + /* + * We can't really set clock ratio, if the mclk/lrclk is different + * from what we provide, then error out + */ + if (mfs && mfs != mode->mst_mfs) + return -EINVAL; + + es7241_set_mode(priv, mode->mst_m0, mode->mst_m1); + + return 0; +} + +static int es7241_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct es7241_data *priv = snd_soc_dai_get_drvdata(dai); + unsigned int rate = params_rate(params); + unsigned int mfs = priv->mclk / rate; + int i; + + for (i = 0; i < priv->chip->mode_num; i++) { + const struct es7241_clock_mode *mode = &priv->chip->modes[i]; + + if (rate < mode->rate_min || rate >= mode->rate_max) + continue; + + if (priv->is_slave) + return es7241_set_slave_mode(priv, mode, mfs); + else + return es7241_set_master_mode(priv, mode, mfs); + } + + /* should not happen */ + dev_err(dai->dev, "unsupported rate: %u\n", rate); + return -EINVAL; +} + +static int es7241_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct es7241_data *priv = snd_soc_dai_get_drvdata(dai); + + if (dir == SND_SOC_CLOCK_IN && clk_id == 0) { + priv->mclk = freq; + return 0; + } + + return -ENOTSUPP; +} + +static int es7241_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct es7241_data *priv = snd_soc_dai_get_drvdata(dai); + + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) { + dev_err(dai->dev, "Unsupported dai clock inversion\n"); + return -EINVAL; + } + + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != priv->fmt) { + dev_err(dai->dev, "Invalid dai format\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + priv->is_slave = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + priv->is_slave = false; + break; + + default: + dev_err(dai->dev, "Unsupported clock configuration\n"); + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops es7241_dai_ops = { + .set_fmt = es7241_set_fmt, + .hw_params = es7241_hw_params, + .set_sysclk = es7241_set_sysclk, +}; + +static struct snd_soc_dai_driver es7241_dai = { + .name = "es7241-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &es7241_dai_ops, +}; + +static const struct es7241_clock_mode es7241_modes[] = { + { + /* Single speed mode */ + .rate_min = 8000, + .rate_max = 50000, + .slv_mfs = (unsigned int[]) { 256, 384, 512, 768, 1024 }, + .slv_mfs_num = 5, + .mst_mfs = 256, + .mst_m0 = 0, + .mst_m1 = 0, + }, { + /* Double speed mode */ + .rate_min = 50000, + .rate_max = 100000, + .slv_mfs = (unsigned int[]) { 128, 192 }, + .slv_mfs_num = 2, + .mst_mfs = 128, + .mst_m0 = 1, + .mst_m1 = 0, + }, { + /* Quad speed mode */ + .rate_min = 100000, + .rate_max = 200000, + .slv_mfs = (unsigned int[]) { 64 }, + .slv_mfs_num = 1, + .mst_mfs = 64, + .mst_m0 = 0, + .mst_m1 = 1, + }, +}; + +static const struct es7241_chip es7241_chip = { + .modes = es7241_modes, + .mode_num = ARRAY_SIZE(es7241_modes), +}; + +static const struct snd_soc_dapm_widget es7241_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("AINL"), + SND_SOC_DAPM_INPUT("AINR"), + SND_SOC_DAPM_DAC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDDP", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDDD", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDDA", 0, 0), +}; + +static const struct snd_soc_dapm_route es7241_dapm_routes[] = { + { "ADC", NULL, "AINL", }, + { "ADC", NULL, "AINR", }, + { "ADC", NULL, "VDDA", }, + { "Capture", NULL, "VDDP", }, + { "Capture", NULL, "VDDD", }, +}; + +static const struct snd_soc_component_driver es7241_component_driver = { + .dapm_widgets = es7241_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es7241_dapm_widgets), + .dapm_routes = es7241_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es7241_dapm_routes), + .idle_bias_on = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static void es7241_parse_fmt(struct device *dev, struct es7241_data *priv) +{ + bool is_leftj; + + /* + * The format is given by a pull resistor on the SDOUT pin: + * pull-up for i2s, pull-down for left justified. + */ + is_leftj = of_property_read_bool(dev->of_node, + "everest,sdout-pull-down"); + if (is_leftj) + priv->fmt = SND_SOC_DAIFMT_LEFT_J; + else + priv->fmt = SND_SOC_DAIFMT_I2S; +} + +static int es7241_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct es7241_data *priv; + int err; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + priv->chip = of_device_get_match_data(dev); + if (!priv->chip) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + es7241_parse_fmt(dev, priv); + + priv->reset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW); + if (IS_ERR(priv->reset)) { + err = PTR_ERR(priv->reset); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'reset' gpio: %d", err); + return err; + } + + priv->m0 = devm_gpiod_get_optional(dev, "m0", GPIOD_OUT_LOW); + if (IS_ERR(priv->m0)) { + err = PTR_ERR(priv->m0); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'm0' gpio: %d", err); + return err; + } + + priv->m1 = devm_gpiod_get_optional(dev, "m1", GPIOD_OUT_LOW); + if (IS_ERR(priv->m1)) { + err = PTR_ERR(priv->m1); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'm1' gpio: %d", err); + return err; + } + + return devm_snd_soc_register_component(&pdev->dev, + &es7241_component_driver, + &es7241_dai, 1); +} + +#ifdef CONFIG_OF +static const struct of_device_id es7241_ids[] = { + { .compatible = "everest,es7241", .data = &es7241_chip }, + { } +}; +MODULE_DEVICE_TABLE(of, es7241_ids); +#endif + +static struct platform_driver es7241_driver = { + .driver = { + .name = "es7241", + .of_match_table = of_match_ptr(es7241_ids), + }, + .probe = es7241_probe, +}; + +module_platform_driver(es7241_driver); + +MODULE_DESCRIPTION("ASoC ES7241 audio codec driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 84f7a7a36e4b..7b8533abf637 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -85,7 +85,7 @@ struct hdac_hdmi_pin { bool mst_capable; struct hdac_hdmi_port *ports; int num_ports; - struct hdac_ext_device *edev; + struct hdac_device *hdev; }; struct hdac_hdmi_port { @@ -126,6 +126,9 @@ struct hdac_hdmi_drv_data { }; struct hdac_hdmi_priv { + struct hdac_device *hdev; + struct snd_soc_component *component; + struct snd_card *card; struct hdac_hdmi_dai_port_map dai_map[HDA_MAX_CVTS]; struct list_head pin_list; struct list_head cvt_list; @@ -139,7 +142,7 @@ struct hdac_hdmi_priv { struct snd_soc_dai_driver *dai_drv; }; -#define hdev_to_hdmi_priv(_hdev) ((to_ehdac_device(_hdev))->private_data) +#define hdev_to_hdmi_priv(_hdev) dev_get_drvdata(&(_hdev)->dev) static struct hdac_hdmi_pcm * hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, @@ -158,7 +161,7 @@ hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, struct hdac_hdmi_port *port, bool is_connect) { - struct hdac_ext_device *edev = port->pin->edev; + struct hdac_device *hdev = port->pin->hdev; if (is_connect) snd_soc_dapm_enable_pin(port->dapm, port->jack_pin); @@ -172,7 +175,7 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, * ports. */ if (pcm->jack_event == 0) { - dev_dbg(&edev->hdev.dev, + dev_dbg(&hdev->dev, "jack report for pcm=%d\n", pcm->pcm_id); snd_soc_jack_report(pcm->jack, SND_JACK_AVOUT, @@ -198,19 +201,18 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, /* * Get the no devices that can be connected to a port on the Pin widget. */ -static int hdac_hdmi_get_port_len(struct hdac_ext_device *edev, hda_nid_t nid) +static int hdac_hdmi_get_port_len(struct hdac_device *hdev, hda_nid_t nid) { unsigned int caps; unsigned int type, param; - caps = get_wcaps(&edev->hdev, nid); + caps = get_wcaps(hdev, nid); type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL) || (type != AC_WID_PIN)) return 0; - param = snd_hdac_read_parm_uncached(&edev->hdev, nid, - AC_PAR_DEVLIST_LEN); + param = snd_hdac_read_parm_uncached(hdev, nid, AC_PAR_DEVLIST_LEN); if (param == -1) return param; @@ -222,10 +224,10 @@ static int hdac_hdmi_get_port_len(struct hdac_ext_device *edev, hda_nid_t nid) * id selected on the pin. Return 0 means the first port entry * is selected or MST is not supported. */ -static int hdac_hdmi_port_select_get(struct hdac_ext_device *edev, +static int hdac_hdmi_port_select_get(struct hdac_device *hdev, struct hdac_hdmi_port *port) { - return snd_hdac_codec_read(&edev->hdev, port->pin->nid, + return snd_hdac_codec_read(hdev, port->pin->nid, 0, AC_VERB_GET_DEVICE_SEL, 0); } @@ -233,7 +235,7 @@ static int hdac_hdmi_port_select_get(struct hdac_ext_device *edev, * Sets the selected port entry for the configuring Pin widget verb. * returns error if port set is not equal to port get otherwise success */ -static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev, +static int hdac_hdmi_port_select_set(struct hdac_device *hdev, struct hdac_hdmi_port *port) { int num_ports; @@ -242,8 +244,7 @@ static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev, return 0; /* AC_PAR_DEVLIST_LEN is 0 based. */ - num_ports = hdac_hdmi_get_port_len(edev, port->pin->nid); - + num_ports = hdac_hdmi_get_port_len(hdev, port->pin->nid); if (num_ports < 0) return -EIO; /* @@ -253,13 +254,13 @@ static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev, if (num_ports + 1 < port->id) return 0; - snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0, + snd_hdac_codec_write(hdev, port->pin->nid, 0, AC_VERB_SET_DEVICE_SEL, port->id); - if (port->id != hdac_hdmi_port_select_get(edev, port)) + if (port->id != hdac_hdmi_port_select_get(hdev, port)) return -EIO; - dev_dbg(&edev->hdev.dev, "Selected the port=%d\n", port->id); + dev_dbg(&hdev->dev, "Selected the port=%d\n", port->id); return 0; } @@ -277,13 +278,6 @@ static struct hdac_hdmi_pcm *get_hdmi_pcm_from_id(struct hdac_hdmi_priv *hdmi, return NULL; } -static inline struct hdac_ext_device *to_hda_ext_device(struct device *dev) -{ - struct hdac_device *hdev = dev_to_hdac_dev(dev); - - return to_ehdac_device(hdev); -} - static unsigned int sad_format(const u8 *sad) { return ((sad[0] >> 0x3) & 0x1f); @@ -324,15 +318,13 @@ format_constraint: } static void -hdac_hdmi_set_dip_index(struct hdac_ext_device *edev, hda_nid_t pin_nid, +hdac_hdmi_set_dip_index(struct hdac_device *hdev, hda_nid_t pin_nid, int packet_index, int byte_index) { int val; val = (packet_index << 5) | (byte_index & 0x1f); - - snd_hdac_codec_write(&edev->hdev, pin_nid, 0, - AC_VERB_SET_HDMI_DIP_INDEX, val); + snd_hdac_codec_write(hdev, pin_nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val); } struct dp_audio_infoframe { @@ -347,14 +339,14 @@ struct dp_audio_infoframe { u8 LFEPBL01_LSV36_DM_INH7; }; -static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *edev, +static int hdac_hdmi_setup_audio_infoframe(struct hdac_device *hdev, struct hdac_hdmi_pcm *pcm, struct hdac_hdmi_port *port) { uint8_t buffer[HDMI_INFOFRAME_HEADER_SIZE + HDMI_AUDIO_INFOFRAME_SIZE]; struct hdmi_audio_infoframe frame; struct hdac_hdmi_pin *pin = port->pin; struct dp_audio_infoframe dp_ai; - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_cvt *cvt = pcm->cvt; u8 *dip; int ret; @@ -363,11 +355,11 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *edev, u8 conn_type; int channels, ca; - ca = snd_hdac_channel_allocation(&edev->hdev, port->eld.info.spk_alloc, + ca = snd_hdac_channel_allocation(hdev, port->eld.info.spk_alloc, pcm->channels, pcm->chmap_set, true, pcm->chmap); channels = snd_hdac_get_active_channels(ca); - hdmi->chmap.ops.set_channel_count(&edev->hdev, cvt->nid, channels); + hdmi->chmap.ops.set_channel_count(hdev, cvt->nid, channels); snd_hdac_setup_channel_mapping(&hdmi->chmap, pin->nid, false, ca, pcm->channels, pcm->chmap, pcm->chmap_set); @@ -400,32 +392,31 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *edev, break; default: - dev_err(&edev->hdev.dev, "Invalid connection type: %d\n", - conn_type); + dev_err(&hdev->dev, "Invalid connection type: %d\n", conn_type); return -EIO; } /* stop infoframe transmission */ - hdac_hdmi_set_dip_index(edev, pin->nid, 0x0, 0x0); - snd_hdac_codec_write(&edev->hdev, pin->nid, 0, + hdac_hdmi_set_dip_index(hdev, pin->nid, 0x0, 0x0); + snd_hdac_codec_write(hdev, pin->nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_DISABLE); /* Fill infoframe. Index auto-incremented */ - hdac_hdmi_set_dip_index(edev, pin->nid, 0x0, 0x0); + hdac_hdmi_set_dip_index(hdev, pin->nid, 0x0, 0x0); if (conn_type == DRM_ELD_CONN_TYPE_HDMI) { for (i = 0; i < sizeof(buffer); i++) - snd_hdac_codec_write(&edev->hdev, pin->nid, 0, + snd_hdac_codec_write(hdev, pin->nid, 0, AC_VERB_SET_HDMI_DIP_DATA, buffer[i]); } else { for (i = 0; i < sizeof(dp_ai); i++) - snd_hdac_codec_write(&edev->hdev, pin->nid, 0, + snd_hdac_codec_write(hdev, pin->nid, 0, AC_VERB_SET_HDMI_DIP_DATA, dip[i]); } /* Start infoframe */ - hdac_hdmi_set_dip_index(edev, pin->nid, 0x0, 0x0); - snd_hdac_codec_write(&edev->hdev, pin->nid, 0, + hdac_hdmi_set_dip_index(hdev, pin->nid, 0x0, 0x0); + snd_hdac_codec_write(hdev, pin->nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_BEST); return 0; @@ -435,12 +426,12 @@ static int hdac_hdmi_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { - struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); + struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_pcm *pcm; - dev_dbg(&edev->hdev.dev, "%s: strm_tag: %d\n", __func__, tx_mask); + dev_dbg(&hdev->dev, "%s: strm_tag: %d\n", __func__, tx_mask); dai_map = &hdmi->dai_map[dai->id]; @@ -455,8 +446,8 @@ static int hdac_hdmi_set_tdm_slot(struct snd_soc_dai *dai, static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hparams, struct snd_soc_dai *dai) { - struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); + struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_port *port; struct hdac_hdmi_pcm *pcm; @@ -469,7 +460,7 @@ static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream, return -ENODEV; if ((!port->eld.monitor_present) || (!port->eld.eld_valid)) { - dev_err(&edev->hdev.dev, + dev_err(&hdev->dev, "device is not configured for this pin:port%d:%d\n", port->pin->nid, port->id); return -ENODEV; @@ -489,28 +480,28 @@ static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream, return 0; } -static int hdac_hdmi_query_port_connlist(struct hdac_ext_device *edev, +static int hdac_hdmi_query_port_connlist(struct hdac_device *hdev, struct hdac_hdmi_pin *pin, struct hdac_hdmi_port *port) { - if (!(get_wcaps(&edev->hdev, pin->nid) & AC_WCAP_CONN_LIST)) { - dev_warn(&edev->hdev.dev, + if (!(get_wcaps(hdev, pin->nid) & AC_WCAP_CONN_LIST)) { + dev_warn(&hdev->dev, "HDMI: pin %d wcaps %#x does not support connection list\n", - pin->nid, get_wcaps(&edev->hdev, pin->nid)); + pin->nid, get_wcaps(hdev, pin->nid)); return -EINVAL; } - if (hdac_hdmi_port_select_set(edev, port) < 0) + if (hdac_hdmi_port_select_set(hdev, port) < 0) return -EIO; - port->num_mux_nids = snd_hdac_get_connections(&edev->hdev, pin->nid, + port->num_mux_nids = snd_hdac_get_connections(hdev, pin->nid, port->mux_nids, HDA_MAX_CONNECTIONS); if (port->num_mux_nids == 0) - dev_warn(&edev->hdev.dev, + dev_warn(&hdev->dev, "No connections found for pin:port %d:%d\n", pin->nid, port->id); - dev_dbg(&edev->hdev.dev, "num_mux_nids %d for pin:port %d:%d\n", + dev_dbg(&hdev->dev, "num_mux_nids %d for pin:port %d:%d\n", port->num_mux_nids, pin->nid, port->id); return port->num_mux_nids; @@ -526,7 +517,7 @@ static int hdac_hdmi_query_port_connlist(struct hdac_ext_device *edev, * connected. */ static struct hdac_hdmi_port *hdac_hdmi_get_port_from_cvt( - struct hdac_ext_device *edev, + struct hdac_device *hdev, struct hdac_hdmi_priv *hdmi, struct hdac_hdmi_cvt *cvt) { @@ -541,7 +532,7 @@ static struct hdac_hdmi_port *hdac_hdmi_get_port_from_cvt( list_for_each_entry(port, &pcm->port_list, head) { mutex_lock(&pcm->lock); - ret = hdac_hdmi_query_port_connlist(edev, + ret = hdac_hdmi_query_port_connlist(hdev, port->pin, port); mutex_unlock(&pcm->lock); if (ret < 0) @@ -568,8 +559,8 @@ static struct hdac_hdmi_port *hdac_hdmi_get_port_from_cvt( static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); + struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_cvt *cvt; struct hdac_hdmi_port *port; @@ -578,7 +569,7 @@ static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream, dai_map = &hdmi->dai_map[dai->id]; cvt = dai_map->cvt; - port = hdac_hdmi_get_port_from_cvt(edev, hdmi, cvt); + port = hdac_hdmi_get_port_from_cvt(hdev, hdmi, cvt); /* * To make PA and other userland happy. @@ -589,7 +580,7 @@ static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream, if ((!port->eld.monitor_present) || (!port->eld.eld_valid)) { - dev_warn(&edev->hdev.dev, + dev_warn(&hdev->dev, "Failed: present?:%d ELD valid?:%d pin:port: %d:%d\n", port->eld.monitor_present, port->eld.eld_valid, port->pin->nid, port->id); @@ -611,8 +602,7 @@ static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream, static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_device *edev = snd_soc_dai_get_drvdata(dai); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_pcm *pcm; @@ -695,10 +685,10 @@ static void hdac_hdmi_fill_route(struct snd_soc_dapm_route *route, route->connected = handler; } -static struct hdac_hdmi_pcm *hdac_hdmi_get_pcm(struct hdac_ext_device *edev, +static struct hdac_hdmi_pcm *hdac_hdmi_get_pcm(struct hdac_device *hdev, struct hdac_hdmi_port *port) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm = NULL; struct hdac_hdmi_port *p; @@ -715,33 +705,32 @@ static struct hdac_hdmi_pcm *hdac_hdmi_get_pcm(struct hdac_ext_device *edev, return NULL; } -static void hdac_hdmi_set_power_state(struct hdac_ext_device *edev, +static void hdac_hdmi_set_power_state(struct hdac_device *hdev, hda_nid_t nid, unsigned int pwr_state) { int count; unsigned int state; - if (get_wcaps(&edev->hdev, nid) & AC_WCAP_POWER) { - if (!snd_hdac_check_power_state(&edev->hdev, nid, pwr_state)) { + if (get_wcaps(hdev, nid) & AC_WCAP_POWER) { + if (!snd_hdac_check_power_state(hdev, nid, pwr_state)) { for (count = 0; count < 10; count++) { - snd_hdac_codec_read(&edev->hdev, nid, 0, + snd_hdac_codec_read(hdev, nid, 0, AC_VERB_SET_POWER_STATE, pwr_state); - state = snd_hdac_sync_power_state(&edev->hdev, + state = snd_hdac_sync_power_state(hdev, nid, pwr_state); if (!(state & AC_PWRST_ERROR)) break; } } - } } -static void hdac_hdmi_set_amp(struct hdac_ext_device *edev, +static void hdac_hdmi_set_amp(struct hdac_device *hdev, hda_nid_t nid, int val) { - if (get_wcaps(&edev->hdev, nid) & AC_WCAP_OUT_AMP) - snd_hdac_codec_write(&edev->hdev, nid, 0, + if (get_wcaps(hdev, nid) & AC_WCAP_OUT_AMP) + snd_hdac_codec_write(hdev, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, val); } @@ -750,40 +739,40 @@ static int hdac_hdmi_pin_output_widget_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kc, int event) { struct hdac_hdmi_port *port = w->priv; - struct hdac_ext_device *edev = to_hda_ext_device(w->dapm->dev); + struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev); struct hdac_hdmi_pcm *pcm; - dev_dbg(&edev->hdev.dev, "%s: widget: %s event: %x\n", + dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n", __func__, w->name, event); - pcm = hdac_hdmi_get_pcm(edev, port); + pcm = hdac_hdmi_get_pcm(hdev, port); if (!pcm) return -EIO; /* set the device if pin is mst_capable */ - if (hdac_hdmi_port_select_set(edev, port) < 0) + if (hdac_hdmi_port_select_set(hdev, port) < 0) return -EIO; switch (event) { case SND_SOC_DAPM_PRE_PMU: - hdac_hdmi_set_power_state(edev, port->pin->nid, AC_PWRST_D0); + hdac_hdmi_set_power_state(hdev, port->pin->nid, AC_PWRST_D0); /* Enable out path for this pin widget */ - snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0, + snd_hdac_codec_write(hdev, port->pin->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - hdac_hdmi_set_amp(edev, port->pin->nid, AMP_OUT_UNMUTE); + hdac_hdmi_set_amp(hdev, port->pin->nid, AMP_OUT_UNMUTE); - return hdac_hdmi_setup_audio_infoframe(edev, pcm, port); + return hdac_hdmi_setup_audio_infoframe(hdev, pcm, port); case SND_SOC_DAPM_POST_PMD: - hdac_hdmi_set_amp(edev, port->pin->nid, AMP_OUT_MUTE); + hdac_hdmi_set_amp(hdev, port->pin->nid, AMP_OUT_MUTE); /* Disable out path for this pin widget */ - snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0, + snd_hdac_codec_write(hdev, port->pin->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - hdac_hdmi_set_power_state(edev, port->pin->nid, AC_PWRST_D3); + hdac_hdmi_set_power_state(hdev, port->pin->nid, AC_PWRST_D3); break; } @@ -795,11 +784,11 @@ static int hdac_hdmi_cvt_output_widget_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kc, int event) { struct hdac_hdmi_cvt *cvt = w->priv; - struct hdac_ext_device *edev = to_hda_ext_device(w->dapm->dev); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm; - dev_dbg(&edev->hdev.dev, "%s: widget: %s event: %x\n", + dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n", __func__, w->name, event); pcm = hdac_hdmi_get_pcm_from_cvt(hdmi, cvt); @@ -808,29 +797,29 @@ static int hdac_hdmi_cvt_output_widget_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - hdac_hdmi_set_power_state(edev, cvt->nid, AC_PWRST_D0); + hdac_hdmi_set_power_state(hdev, cvt->nid, AC_PWRST_D0); /* Enable transmission */ - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_DIGI_CONVERT_1, 1); /* Category Code (CC) to zero */ - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_DIGI_CONVERT_2, 0); - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_CHANNEL_STREAMID, pcm->stream_tag); - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_STREAM_FORMAT, pcm->format); break; case SND_SOC_DAPM_POST_PMD: - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); - snd_hdac_codec_write(&edev->hdev, cvt->nid, 0, + snd_hdac_codec_write(hdev, cvt->nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); - hdac_hdmi_set_power_state(edev, cvt->nid, AC_PWRST_D3); + hdac_hdmi_set_power_state(hdev, cvt->nid, AC_PWRST_D3); break; } @@ -842,10 +831,10 @@ static int hdac_hdmi_pin_mux_widget_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kc, int event) { struct hdac_hdmi_port *port = w->priv; - struct hdac_ext_device *edev = to_hda_ext_device(w->dapm->dev); + struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev); int mux_idx; - dev_dbg(&edev->hdev.dev, "%s: widget: %s event: %x\n", + dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n", __func__, w->name, event); if (!kc) @@ -854,11 +843,11 @@ static int hdac_hdmi_pin_mux_widget_event(struct snd_soc_dapm_widget *w, mux_idx = dapm_kcontrol_get_value(kc); /* set the device if pin is mst_capable */ - if (hdac_hdmi_port_select_set(edev, port) < 0) + if (hdac_hdmi_port_select_set(hdev, port) < 0) return -EIO; if (mux_idx > 0) { - snd_hdac_codec_write(&edev->hdev, port->pin->nid, 0, + snd_hdac_codec_write(hdev, port->pin->nid, 0, AC_VERB_SET_CONNECT_SEL, (mux_idx - 1)); } @@ -877,8 +866,8 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget *w = snd_soc_dapm_kcontrol_widget(kcontrol); struct snd_soc_dapm_context *dapm = w->dapm; struct hdac_hdmi_port *port = w->priv; - struct hdac_ext_device *edev = to_hda_ext_device(dapm->dev); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_device *hdev = dev_to_hdac_dev(dapm->dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm = NULL; const char *cvt_name = e->texts[ucontrol->value.enumerated.item[0]]; @@ -931,12 +920,12 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, * care of selecting the right one and leaving all other inputs selected to * "NONE" */ -static int hdac_hdmi_create_pin_port_muxs(struct hdac_ext_device *edev, +static int hdac_hdmi_create_pin_port_muxs(struct hdac_device *hdev, struct hdac_hdmi_port *port, struct snd_soc_dapm_widget *widget, const char *widget_name) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pin *pin = port->pin; struct snd_kcontrol_new *kc; struct hdac_hdmi_cvt *cvt; @@ -948,17 +937,17 @@ static int hdac_hdmi_create_pin_port_muxs(struct hdac_ext_device *edev, int i = 0; int num_items = hdmi->num_cvt + 1; - kc = devm_kzalloc(&edev->hdev.dev, sizeof(*kc), GFP_KERNEL); + kc = devm_kzalloc(&hdev->dev, sizeof(*kc), GFP_KERNEL); if (!kc) return -ENOMEM; - se = devm_kzalloc(&edev->hdev.dev, sizeof(*se), GFP_KERNEL); + se = devm_kzalloc(&hdev->dev, sizeof(*se), GFP_KERNEL); if (!se) return -ENOMEM; snprintf(kc_name, NAME_SIZE, "Pin %d port %d Input", pin->nid, port->id); - kc->name = devm_kstrdup(&edev->hdev.dev, kc_name, GFP_KERNEL); + kc->name = devm_kstrdup(&hdev->dev, kc_name, GFP_KERNEL); if (!kc->name) return -ENOMEM; @@ -976,35 +965,35 @@ static int hdac_hdmi_create_pin_port_muxs(struct hdac_ext_device *edev, se->mask = roundup_pow_of_two(se->items) - 1; sprintf(mux_items, "NONE"); - items[i] = devm_kstrdup(&edev->hdev.dev, mux_items, GFP_KERNEL); + items[i] = devm_kstrdup(&hdev->dev, mux_items, GFP_KERNEL); if (!items[i]) return -ENOMEM; list_for_each_entry(cvt, &hdmi->cvt_list, head) { i++; sprintf(mux_items, "cvt %d", cvt->nid); - items[i] = devm_kstrdup(&edev->hdev.dev, mux_items, GFP_KERNEL); + items[i] = devm_kstrdup(&hdev->dev, mux_items, GFP_KERNEL); if (!items[i]) return -ENOMEM; } - se->texts = devm_kmemdup(&edev->hdev.dev, items, + se->texts = devm_kmemdup(&hdev->dev, items, (num_items * sizeof(char *)), GFP_KERNEL); if (!se->texts) return -ENOMEM; - return hdac_hdmi_fill_widget_info(&edev->hdev.dev, widget, + return hdac_hdmi_fill_widget_info(&hdev->dev, widget, snd_soc_dapm_mux, port, widget_name, NULL, kc, 1, hdac_hdmi_pin_mux_widget_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_REG); } /* Add cvt <- input <- mux route map */ -static void hdac_hdmi_add_pinmux_cvt_route(struct hdac_ext_device *edev, +static void hdac_hdmi_add_pinmux_cvt_route(struct hdac_device *hdev, struct snd_soc_dapm_widget *widgets, struct snd_soc_dapm_route *route, int rindex) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); const struct snd_kcontrol_new *kc; struct soc_enum *se; int mux_index = hdmi->num_cvt + hdmi->num_ports; @@ -1046,8 +1035,8 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *widgets; struct snd_soc_dapm_route *route; - struct hdac_ext_device *edev = to_hda_ext_device(dapm->dev); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_device *hdev = dev_to_hdac_dev(dapm->dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct snd_soc_dai_driver *dai_drv = hdmi->dai_drv; char widget_name[NAME_SIZE]; struct hdac_hdmi_cvt *cvt; @@ -1099,7 +1088,7 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm) for (j = 0; j < pin->num_ports; j++) { sprintf(widget_name, "Pin%d-Port%d Mux", pin->nid, pin->ports[j].id); - ret = hdac_hdmi_create_pin_port_muxs(edev, + ret = hdac_hdmi_create_pin_port_muxs(hdev, &pin->ports[j], &widgets[i], widget_name); if (ret < 0) @@ -1134,7 +1123,7 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm) } } - hdac_hdmi_add_pinmux_cvt_route(edev, widgets, route, i); + hdac_hdmi_add_pinmux_cvt_route(hdev, widgets, route, i); snd_soc_dapm_new_controls(dapm, widgets, ((2 * hdmi->num_ports) + hdmi->num_cvt)); @@ -1146,9 +1135,9 @@ static int create_fill_widget_route_map(struct snd_soc_dapm_context *dapm) } -static int hdac_hdmi_init_dai_map(struct hdac_ext_device *edev) +static int hdac_hdmi_init_dai_map(struct hdac_device *hdev) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_cvt *cvt; int dai_id = 0; @@ -1164,7 +1153,7 @@ static int hdac_hdmi_init_dai_map(struct hdac_ext_device *edev) dai_id++; if (dai_id == HDA_MAX_CVTS) { - dev_warn(&edev->hdev.dev, + dev_warn(&hdev->dev, "Max dais supported: %d\n", dai_id); break; } @@ -1173,9 +1162,9 @@ static int hdac_hdmi_init_dai_map(struct hdac_ext_device *edev) return 0; } -static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid) +static int hdac_hdmi_add_cvt(struct hdac_device *hdev, hda_nid_t nid) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_cvt *cvt; char name[NAME_SIZE]; @@ -1190,10 +1179,10 @@ static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid) list_add_tail(&cvt->head, &hdmi->cvt_list); hdmi->num_cvt++; - return hdac_hdmi_query_cvt_params(&edev->hdev, cvt); + return hdac_hdmi_query_cvt_params(hdev, cvt); } -static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev, +static int hdac_hdmi_parse_eld(struct hdac_device *hdev, struct hdac_hdmi_port *port) { unsigned int ver, mnl; @@ -1202,7 +1191,7 @@ static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev, >> DRM_ELD_VER_SHIFT; if (ver != ELD_VER_CEA_861D && ver != ELD_VER_PARTIAL) { - dev_err(&edev->hdev.dev, "HDMI: Unknown ELD version %d\n", ver); + dev_err(&hdev->dev, "HDMI: Unknown ELD version %d\n", ver); return -EINVAL; } @@ -1210,7 +1199,7 @@ static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev, DRM_ELD_MNL_MASK) >> DRM_ELD_MNL_SHIFT; if (mnl > ELD_MAX_MNL) { - dev_err(&edev->hdev.dev, "HDMI: MNL Invalid %d\n", mnl); + dev_err(&hdev->dev, "HDMI: MNL Invalid %d\n", mnl); return -EINVAL; } @@ -1222,8 +1211,8 @@ static int hdac_hdmi_parse_eld(struct hdac_ext_device *edev, static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, struct hdac_hdmi_port *port) { - struct hdac_ext_device *edev = pin->edev; - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_device *hdev = pin->hdev; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm; int size = 0; int port_id = -1; @@ -1241,14 +1230,14 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, if (pin->mst_capable) port_id = port->id; - size = snd_hdac_acomp_get_eld(&edev->hdev, pin->nid, port_id, + size = snd_hdac_acomp_get_eld(hdev, pin->nid, port_id, &port->eld.monitor_present, port->eld.eld_buffer, ELD_MAX_SIZE); if (size > 0) { size = min(size, ELD_MAX_SIZE); - if (hdac_hdmi_parse_eld(edev, port) < 0) + if (hdac_hdmi_parse_eld(hdev, port) < 0) size = -EINVAL; } @@ -1260,11 +1249,11 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, port->eld.eld_size = 0; } - pcm = hdac_hdmi_get_pcm(edev, port); + pcm = hdac_hdmi_get_pcm(hdev, port); if (!port->eld.monitor_present || !port->eld.eld_valid) { - dev_err(&edev->hdev.dev, "%s: disconnect for pin:port %d:%d\n", + dev_err(&hdev->dev, "%s: disconnect for pin:port %d:%d\n", __func__, pin->nid, port->id); /* @@ -1316,9 +1305,9 @@ static int hdac_hdmi_add_ports(struct hdac_hdmi_priv *hdmi, return 0; } -static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid) +static int hdac_hdmi_add_pin(struct hdac_device *hdev, hda_nid_t nid) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pin *pin; int ret; @@ -1328,7 +1317,7 @@ static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid) pin->nid = nid; pin->mst_capable = false; - pin->edev = edev; + pin->hdev = hdev; ret = hdac_hdmi_add_ports(hdmi, pin); if (ret < 0) return ret; @@ -1459,15 +1448,14 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdev, * Parse all nodes and store the cvt/pin nids in array * Add one time initialization for pin and cvt widgets */ -static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev, +static int hdac_hdmi_parse_and_map_nid(struct hdac_device *hdev, struct snd_soc_dai_driver **dais, int *num_dais) { hda_nid_t nid; int i, num_nodes; struct hdac_hdmi_cvt *temp_cvt, *cvt_next; struct hdac_hdmi_pin *temp_pin, *pin_next; - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); - struct hdac_device *hdev = &edev->hdev; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); int ret; hdac_hdmi_skl_enable_all_pins(hdev); @@ -1492,13 +1480,13 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev, switch (type) { case AC_WID_AUD_OUT: - ret = hdac_hdmi_add_cvt(edev, nid); + ret = hdac_hdmi_add_cvt(hdev, nid); if (ret < 0) goto free_widgets; break; case AC_WID_PIN: - ret = hdac_hdmi_add_pin(edev, nid); + ret = hdac_hdmi_add_pin(hdev, nid); if (ret < 0) goto free_widgets; break; @@ -1518,7 +1506,7 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev, } *num_dais = hdmi->num_cvt; - ret = hdac_hdmi_init_dai_map(edev); + ret = hdac_hdmi_init_dai_map(hdev); if (ret < 0) goto free_widgets; @@ -1542,19 +1530,24 @@ free_widgets: return ret; } +static int hdac_hdmi_pin2port(void *aptr, int pin) +{ + return pin - 4; /* map NID 0x05 -> port #1 */ +} + static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) { - struct hdac_ext_device *edev = aptr; - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_device *hdev = aptr; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pin *pin = NULL; struct hdac_hdmi_port *hport = NULL; - struct snd_soc_component *component = edev->scodec; + struct snd_soc_component *component = hdmi->component; int i; /* Don't know how this mapping is derived */ hda_nid_t pin_nid = port + 0x04; - dev_dbg(&edev->hdev.dev, "%s: for pin:%d port=%d\n", __func__, + dev_dbg(&hdev->dev, "%s: for pin:%d port=%d\n", __func__, pin_nid, pipe); /* @@ -1567,7 +1560,7 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) SNDRV_CTL_POWER_D0) return; - if (atomic_read(&edev->hdev.in_pm)) + if (atomic_read(&hdev->in_pm)) return; list_for_each_entry(pin, &hdmi->pin_list, head) { @@ -1595,7 +1588,8 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) } -static struct i915_audio_component_audio_ops aops = { +static struct drm_audio_component_audio_ops aops = { + .pin2port = hdac_hdmi_pin2port, .pin_eld_notify = hdac_hdmi_eld_notify_cb, }; @@ -1614,15 +1608,15 @@ static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card, /* create jack pin kcontrols */ static int create_fill_jack_kcontrols(struct snd_soc_card *card, - struct hdac_ext_device *edev) + struct hdac_device *hdev) { struct hdac_hdmi_pin *pin; struct snd_kcontrol_new *kc; char kc_name[NAME_SIZE], xname[NAME_SIZE]; char *name; int i = 0, j; - struct snd_soc_component *component = edev->scodec; - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); + struct snd_soc_component *component = hdmi->component; kc = devm_kcalloc(component->dev, hdmi->num_ports, sizeof(*kc), GFP_KERNEL); @@ -1659,8 +1653,8 @@ static int create_fill_jack_kcontrols(struct snd_soc_card *card, int hdac_hdmi_jack_port_init(struct snd_soc_component *component, struct snd_soc_dapm_context *dapm) { - struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_pin *pin; struct snd_soc_dapm_widget *widgets; struct snd_soc_dapm_route *route; @@ -1715,7 +1709,7 @@ int hdac_hdmi_jack_port_init(struct snd_soc_component *component, return ret; /* Add Jack Pin switch Kcontrol */ - ret = create_fill_jack_kcontrols(dapm->card, edev); + ret = create_fill_jack_kcontrols(dapm->card, hdev); if (ret < 0) return ret; @@ -1735,8 +1729,8 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, struct snd_soc_jack *jack) { struct snd_soc_component *component = dai->component; - struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_pcm *pcm; struct snd_pcm *snd_pcm; int err; @@ -1758,7 +1752,7 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, if (snd_pcm) { err = snd_hdac_add_chmap_ctls(snd_pcm, device, &hdmi->chmap); if (err < 0) { - dev_err(&edev->hdev.dev, + dev_err(&hdev->dev, "chmap control add failed with err: %d for pcm: %d\n", err, device); kfree(pcm); @@ -1772,7 +1766,7 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, } EXPORT_SYMBOL_GPL(hdac_hdmi_jack_init); -static void hdac_hdmi_present_sense_all_pins(struct hdac_ext_device *edev, +static void hdac_hdmi_present_sense_all_pins(struct hdac_device *hdev, struct hdac_hdmi_priv *hdmi, bool detect_pin_caps) { int i; @@ -1781,7 +1775,7 @@ static void hdac_hdmi_present_sense_all_pins(struct hdac_ext_device *edev, list_for_each_entry(pin, &hdmi->pin_list, head) { if (detect_pin_caps) { - if (hdac_hdmi_get_port_len(edev, pin->nid) == 0) + if (hdac_hdmi_get_port_len(hdev, pin->nid) == 0) pin->mst_capable = false; else pin->mst_capable = true; @@ -1798,68 +1792,67 @@ static void hdac_hdmi_present_sense_all_pins(struct hdac_ext_device *edev, static int hdmi_codec_probe(struct snd_soc_component *component) { - struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct hdac_ext_link *hlink = NULL; int ret; - edev->scodec = component; + hdmi->component = component; /* * hold the ref while we probe, also no need to drop the ref on * exit, we call pm_runtime_suspend() so that will do for us */ - hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdev.dev)); + hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&edev->hdev.dev, "hdac link not found\n"); + dev_err(&hdev->dev, "hdac link not found\n"); return -EIO; } - snd_hdac_ext_bus_link_get(edev->ebus, hlink); + snd_hdac_ext_bus_link_get(hdev->bus, hlink); ret = create_fill_widget_route_map(dapm); if (ret < 0) return ret; - aops.audio_ptr = edev; - ret = snd_hdac_i915_register_notifier(&aops); + aops.audio_ptr = hdev; + ret = snd_hdac_acomp_register_notifier(hdev->bus, &aops); if (ret < 0) { - dev_err(&edev->hdev.dev, "notifier register failed: err: %d\n", - ret); + dev_err(&hdev->dev, "notifier register failed: err: %d\n", ret); return ret; } - hdac_hdmi_present_sense_all_pins(edev, hdmi, true); + hdac_hdmi_present_sense_all_pins(hdev, hdmi, true); /* Imp: Store the card pointer in hda_codec */ - edev->card = dapm->card->snd_card; + hdmi->card = dapm->card->snd_card; /* * hdac_device core already sets the state to active and calls * get_noresume. So enable runtime and set the device to suspend. */ - pm_runtime_enable(&edev->hdev.dev); - pm_runtime_put(&edev->hdev.dev); - pm_runtime_suspend(&edev->hdev.dev); + pm_runtime_enable(&hdev->dev); + pm_runtime_put(&hdev->dev); + pm_runtime_suspend(&hdev->dev); return 0; } static void hdmi_codec_remove(struct snd_soc_component *component) { - struct hdac_ext_device *edev = snd_soc_component_get_drvdata(component); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; - pm_runtime_disable(&edev->hdev.dev); + pm_runtime_disable(&hdev->dev); } #ifdef CONFIG_PM static int hdmi_codec_prepare(struct device *dev) { - struct hdac_ext_device *edev = to_hda_ext_device(dev); - struct hdac_device *hdev = &edev->hdev; + struct hdac_device *hdev = dev_to_hdac_dev(dev); - pm_runtime_get_sync(&edev->hdev.dev); + pm_runtime_get_sync(&hdev->dev); /* * Power down afg. @@ -1876,16 +1869,15 @@ static int hdmi_codec_prepare(struct device *dev) static void hdmi_codec_complete(struct device *dev) { - struct hdac_ext_device *edev = to_hda_ext_device(dev); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); - struct hdac_device *hdev = &edev->hdev; + struct hdac_device *hdev = dev_to_hdac_dev(dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); /* Power up afg */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - hdac_hdmi_skl_enable_all_pins(&edev->hdev); - hdac_hdmi_skl_enable_dp12(&edev->hdev); + hdac_hdmi_skl_enable_all_pins(hdev); + hdac_hdmi_skl_enable_dp12(hdev); /* * As the ELD notify callback request is not entertained while the @@ -1893,9 +1885,9 @@ static void hdmi_codec_complete(struct device *dev) * all pins here. pin capablity change is not support, so use the * already set pin caps. */ - hdac_hdmi_present_sense_all_pins(edev, hdmi, false); + hdac_hdmi_present_sense_all_pins(hdev, hdmi, false); - pm_runtime_put_sync(&edev->hdev.dev); + pm_runtime_put_sync(&hdev->dev); } #else #define hdmi_codec_prepare NULL @@ -1922,7 +1914,6 @@ static void hdac_hdmi_get_chmap(struct hdac_device *hdev, int pcm_idx, static void hdac_hdmi_set_chmap(struct hdac_device *hdev, int pcm_idx, unsigned char *chmap, int prepared) { - struct hdac_ext_device *edev = to_ehdac_device(hdev); struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); struct hdac_hdmi_port *port; @@ -1938,7 +1929,7 @@ static void hdac_hdmi_set_chmap(struct hdac_device *hdev, int pcm_idx, memcpy(pcm->chmap, chmap, ARRAY_SIZE(pcm->chmap)); list_for_each_entry(port, &pcm->port_list, head) if (prepared) - hdac_hdmi_setup_audio_infoframe(edev, pcm, port); + hdac_hdmi_setup_audio_infoframe(hdev, pcm, port); mutex_unlock(&pcm->lock); } @@ -1987,10 +1978,9 @@ static struct hdac_hdmi_drv_data intel_drv_data = { .vendor_nid = INTEL_VENDOR_NID, }; -static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) +static int hdac_hdmi_dev_probe(struct hdac_device *hdev) { - struct hdac_device *hdev = &edev->hdev; - struct hdac_hdmi_priv *hdmi_priv; + struct hdac_hdmi_priv *hdmi_priv = NULL; struct snd_soc_dai_driver *hdmi_dais = NULL; struct hdac_ext_link *hlink = NULL; int num_dais = 0; @@ -1999,24 +1989,24 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) const struct hda_device_id *hdac_id = hdac_get_device_id(hdev, hdrv); /* hold the ref while we probe */ - hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdev.dev)); + hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&edev->hdev.dev, "hdac link not found\n"); + dev_err(&hdev->dev, "hdac link not found\n"); return -EIO; } - snd_hdac_ext_bus_link_get(edev->ebus, hlink); + snd_hdac_ext_bus_link_get(hdev->bus, hlink); hdmi_priv = devm_kzalloc(&hdev->dev, sizeof(*hdmi_priv), GFP_KERNEL); if (hdmi_priv == NULL) return -ENOMEM; - edev->private_data = hdmi_priv; snd_hdac_register_chmap_ops(hdev, &hdmi_priv->chmap); hdmi_priv->chmap.ops.get_chmap = hdac_hdmi_get_chmap; hdmi_priv->chmap.ops.set_chmap = hdac_hdmi_set_chmap; hdmi_priv->chmap.ops.is_pcm_attached = is_hdac_hdmi_pcm_attached; hdmi_priv->chmap.ops.get_spk_alloc = hdac_hdmi_get_spk_alloc; + hdmi_priv->hdev = hdev; if (!hdac_id) return -ENODEV; @@ -2027,7 +2017,7 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) else hdmi_priv->drv_data = &intel_drv_data; - dev_set_drvdata(&hdev->dev, edev); + dev_set_drvdata(&hdev->dev, hdmi_priv); INIT_LIST_HEAD(&hdmi_priv->pin_list); INIT_LIST_HEAD(&hdmi_priv->cvt_list); @@ -2038,15 +2028,15 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) * Turned off in the runtime_suspend during the first explicit * pm_runtime_suspend call. */ - ret = snd_hdac_display_power(edev->hdev.bus, true); + ret = snd_hdac_display_power(hdev->bus, true); if (ret < 0) { - dev_err(&edev->hdev.dev, + dev_err(&hdev->dev, "Cannot turn on display power on i915 err: %d\n", ret); return ret; } - ret = hdac_hdmi_parse_and_map_nid(edev, &hdmi_dais, &num_dais); + ret = hdac_hdmi_parse_and_map_nid(hdev, &hdmi_dais, &num_dais); if (ret < 0) { dev_err(&hdev->dev, "Failed in parse and map nid with err: %d\n", ret); @@ -2058,14 +2048,14 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) ret = devm_snd_soc_register_component(&hdev->dev, &hdmi_hda_codec, hdmi_dais, num_dais); - snd_hdac_ext_bus_link_put(edev->ebus, hlink); + snd_hdac_ext_bus_link_put(hdev->bus, hlink); return ret; } -static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) +static int hdac_hdmi_dev_remove(struct hdac_device *hdev) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(&edev->hdev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pin *pin, *pin_next; struct hdac_hdmi_cvt *cvt, *cvt_next; struct hdac_hdmi_pcm *pcm, *pcm_next; @@ -2103,12 +2093,79 @@ static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) } #ifdef CONFIG_PM +/* + * Power management sequences + * ========================== + * + * The following explains the PM handling of HDAC HDMI with its parent + * device SKL and display power usage + * + * Probe + * ----- + * In SKL probe, + * 1. skl_probe_work() powers up the display (refcount++ -> 1) + * 2. enumerates the codecs on the link + * 3. powers down the display (refcount-- -> 0) + * + * In HDAC HDMI probe, + * 1. hdac_hdmi_dev_probe() powers up the display (refcount++ -> 1) + * 2. probe the codec + * 3. put the HDAC HDMI device to runtime suspend + * 4. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * + * Once children are runtime suspended, SKL device also goes to runtime + * suspend + * + * HDMI Playback + * ------------- + * Open HDMI device, + * 1. skl_runtime_resume() invoked + * 2. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) + * + * Close HDMI device, + * 1. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * 2. skl_runtime_suspend() invoked + * + * S0/S3 Cycle with playback in progress + * ------------------------------------- + * When the device is opened for playback, the device is runtime active + * already and the display refcount is 1 as explained above. + * + * Entering to S3, + * 1. hdmi_codec_prepare() invoke the runtime resume of codec which just + * increments the PM runtime usage count of the codec since the device + * is in use already + * 2. skl_suspend() powers down the display (refcount-- -> 0) + * + * Wakeup from S3, + * 1. skl_resume() powers up the display (refcount++ -> 1) + * 2. hdmi_codec_complete() invokes the runtime suspend of codec which just + * decrements the PM runtime usage count of the codec since the device + * is in use already + * + * Once playback is stopped, the display refcount is set to 0 as explained + * above in the HDMI playback sequence. The PM handlings are designed in + * such way that to balance the refcount of display power when the codec + * device put to S3 while playback is going on. + * + * S0/S3 Cycle without playback in progress + * ---------------------------------------- + * Entering to S3, + * 1. hdmi_codec_prepare() invoke the runtime resume of codec + * 2. skl_runtime_resume() invoked + * 3. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) + * 4. skl_suspend() powers down the display (refcount-- -> 0) + * + * Wakeup from S3, + * 1. skl_resume() powers up the display (refcount++ -> 1) + * 2. hdmi_codec_complete() invokes the runtime suspend of codec + * 3. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * 4. skl_runtime_suspend() invoked + */ static int hdac_hdmi_runtime_suspend(struct device *dev) { - struct hdac_ext_device *edev = to_hda_ext_device(dev); - struct hdac_device *hdev = &edev->hdev; + struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_bus *bus = hdev->bus; - struct hdac_ext_bus *ebus = hbus_to_ebus(bus); struct hdac_ext_link *hlink = NULL; int err; @@ -2129,27 +2186,25 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) AC_PWRST_D3); err = snd_hdac_display_power(bus, false); if (err < 0) { - dev_err(bus->dev, "Cannot turn on display power on i915\n"); + dev_err(dev, "Cannot turn on display power on i915\n"); return err; } - hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev)); if (!hlink) { dev_err(dev, "hdac link not found\n"); return -EIO; } - snd_hdac_ext_bus_link_put(ebus, hlink); + snd_hdac_ext_bus_link_put(bus, hlink); return 0; } static int hdac_hdmi_runtime_resume(struct device *dev) { - struct hdac_ext_device *edev = to_hda_ext_device(dev); - struct hdac_device *hdev = &edev->hdev; + struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_bus *bus = hdev->bus; - struct hdac_ext_bus *ebus = hbus_to_ebus(bus); struct hdac_ext_link *hlink = NULL; int err; @@ -2159,22 +2214,22 @@ static int hdac_hdmi_runtime_resume(struct device *dev) if (!bus) return 0; - hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev)); if (!hlink) { dev_err(dev, "hdac link not found\n"); return -EIO; } - snd_hdac_ext_bus_link_get(ebus, hlink); + snd_hdac_ext_bus_link_get(bus, hlink); err = snd_hdac_display_power(bus, true); if (err < 0) { - dev_err(bus->dev, "Cannot turn on display power on i915\n"); + dev_err(dev, "Cannot turn on display power on i915\n"); return err; } - hdac_hdmi_skl_enable_all_pins(&edev->hdev); - hdac_hdmi_skl_enable_dp12(&edev->hdev); + hdac_hdmi_skl_enable_all_pins(hdev); + hdac_hdmi_skl_enable_dp12(hdev); /* Power up afg */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, @@ -2206,14 +2261,12 @@ static const struct hda_device_id hdmi_list[] = { MODULE_DEVICE_TABLE(hdaudio, hdmi_list); -static struct hdac_ext_driver hdmi_driver = { - . hdac = { - .driver = { - .name = "HDMI HDA Codec", - .pm = &hdac_hdmi_pm, - }, - .id_table = hdmi_list, +static struct hdac_driver hdmi_driver = { + .driver = { + .name = "HDMI HDA Codec", + .pm = &hdac_hdmi_pm, }, + .id_table = hdmi_list, .probe = hdac_hdmi_dev_probe, .remove = hdac_hdmi_dev_remove, }; diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 38e4a8515709..d00734d31e04 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -291,10 +291,6 @@ static const struct snd_soc_dapm_widget hdmi_widgets[] = { SND_SOC_DAPM_OUTPUT("TX"), }; -static const struct snd_soc_dapm_route hdmi_routes[] = { - { "TX", NULL, "Playback" }, -}; - enum { DAI_ID_I2S = 0, DAI_ID_SPDIF, @@ -689,9 +685,23 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, return snd_ctl_add(rtd->card->snd_card, kctl); } +static int hdmi_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_dapm_context *dapm; + struct snd_soc_dapm_route route = { + .sink = "TX", + .source = dai->driver->playback.stream_name, + }; + + dapm = snd_soc_component_get_dapm(dai->component); + + return snd_soc_dapm_add_routes(dapm, &route, 1); +} + static const struct snd_soc_dai_driver hdmi_i2s_dai = { .name = "i2s-hifi", .id = DAI_ID_I2S, + .probe = hdmi_dai_probe, .playback = { .stream_name = "I2S Playback", .channels_min = 2, @@ -707,6 +717,7 @@ static const struct snd_soc_dai_driver hdmi_i2s_dai = { static const struct snd_soc_dai_driver hdmi_spdif_dai = { .name = "spdif-hifi", .id = DAI_ID_SPDIF, + .probe = hdmi_dai_probe, .playback = { .stream_name = "SPDIF Playback", .channels_min = 2, @@ -733,8 +744,6 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, static const struct snd_soc_component_driver hdmi_driver = { .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), - .dapm_routes = hdmi_routes, - .num_dapm_routes = ARRAY_SIZE(hdmi_routes), .of_xlate_dai_id = hdmi_of_xlate_dai_id, .idle_bias_on = 1, .use_pmdown_time = 1, diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index a92586106932..92b7125ea169 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -488,6 +488,7 @@ static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv, static bool max98373_readable_register(struct device *dev, unsigned int reg) { switch (reg) { + case MAX98373_R2000_SW_RESET: case MAX98373_R2001_INT_RAW1 ... MAX98373_R200C_INT_EN3: case MAX98373_R2010_IRQ_CTRL: case MAX98373_R2014_THERM_WARN_THRESH diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 74d7f52c7e73..6e6134589588 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -52,9 +52,9 @@ static bool max9850_volatile_register(struct device *dev, unsigned int reg) switch (reg) { case MAX9850_STATUSA: case MAX9850_STATUSB: - return 1; + return true; default: - return 0; + return false; } } diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index 17104f8dc1a9..e3c8cd17daf2 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -362,11 +362,8 @@ static const struct snd_soc_dapm_route nau8540_dapm_routes[] = { static int nau8540_clock_check(struct nau8540 *nau8540, int rate, int osr) { - int osrate; - if (osr >= ARRAY_SIZE(osr_adc_sel)) return -EINVAL; - osrate = osr_adc_sel[osr].osr; if (rate * osr > CLK_ADC_MAX) { dev_err(nau8540->dev, "exceed the maximum frequency of CLK_ADC\n"); diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 6bd14453f06e..468d5143e2c4 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1274,7 +1274,7 @@ static int nau8824_calc_fll_param(unsigned int fll_in, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index dc6ea4987b7d..b9fed99d8b5e 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2016,7 +2016,7 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; diff --git a/sound/soc/codecs/pcm1789.c b/sound/soc/codecs/pcm1789.c index 21f15219b3ad..8df6447c76a6 100644 --- a/sound/soc/codecs/pcm1789.c +++ b/sound/soc/codecs/pcm1789.c @@ -262,8 +262,7 @@ int pcm1789_common_exit(struct device *dev) { struct pcm1789_private *priv = dev_get_drvdata(dev); - if (&priv->work) - flush_work(&priv->work); + flush_work(&priv->work); return 0; } diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index 88fde70b1e9e..690c26e7389e 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -265,7 +265,7 @@ static int pcm186x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct pcm186x_priv *priv = snd_soc_component_get_drvdata(component); unsigned int rate = params_rate(params); - unsigned int format = params_format(params); + snd_pcm_format_t format = params_format(params); unsigned int width = params_width(params); unsigned int channels = params_channels(params); unsigned int div_lrck; diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index f4c8c45f4010..c4452efc7970 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -1066,7 +1066,7 @@ static void rt1305_calibrate(struct rt1305_priv *rt1305) pr_debug("Left_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl); pr_info("Left channel %d.%dohm\n", (r0ohm/10), (r0ohm%10)); - r0l = 562949953421312; + r0l = 562949953421312ULL; if (rhl != 0) do_div(r0l, rhl); pr_debug("Left_r0 = 0x%llx\n", r0l); @@ -1083,7 +1083,7 @@ static void rt1305_calibrate(struct rt1305_priv *rt1305) pr_debug("Right_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl); pr_info("Right channel %d.%dohm\n", (r0ohm/10), (r0ohm%10)); - r0r = 562949953421312; + r0r = 562949953421312ULL; if (rhl != 0) do_div(r0r, rhl); pr_debug("Right_r0 = 0x%llx\n", r0r); @@ -1150,17 +1150,11 @@ static int rt1305_i2c_probe(struct i2c_client *i2c, rt1305_reset(rt1305->regmap); rt1305_calibrate(rt1305); - return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt1305, + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_rt1305, rt1305_dai, ARRAY_SIZE(rt1305_dai)); } -static int rt1305_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_component(&i2c->dev); - - return 0; -} - static void rt1305_i2c_shutdown(struct i2c_client *client) { struct rt1305_priv *rt1305 = i2c_get_clientdata(client); @@ -1180,7 +1174,6 @@ static struct i2c_driver rt1305_i2c_driver = { #endif }, .probe = rt1305_i2c_probe, - .remove = rt1305_i2c_remove, .shutdown = rt1305_i2c_shutdown, .id_table = rt1305_i2c_id, }; diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 18686ffb0cd5..6478d10c4f4a 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -268,7 +268,6 @@ static const struct snd_pcm_ops rt5514_spi_pcm_ops = { .hw_params = rt5514_spi_hw_params, .hw_free = rt5514_spi_hw_free, .pointer = rt5514_spi_pcm_pointer, - .mmap = snd_pcm_lib_mmap_vmalloc, .page = snd_pcm_lib_get_vmalloc_page, }; diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 1570b91bf018..dca82dd6e3bf 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, {RT5514_ASRC_IN_CTRL1, 0x00000003}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, }; static const struct reg_default rt5514_reg[] = { @@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = { {RT5514_ASRC_IN_CTRL1, 0x00000003}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, {RT5514_DOWNFILTER1_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER1_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_LDO18_16, 0x02000345}, {RT5514_ANA_CTRL_ADC12, 0x0000a2a8}, diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index cf6dce69eb2a..865f49ac38dd 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -105,9 +105,9 @@ static bool rt5631_volatile_register(struct device *dev, unsigned int reg) case RT5631_INDEX_ADD: case RT5631_INDEX_DATA: case RT5631_EQ_CTRL: - return 1; + return true; default: - return 0; + return false; } } @@ -164,9 +164,9 @@ static bool rt5631_readable_register(struct device *dev, unsigned int reg) case RT5631_VENDOR_ID: case RT5631_VENDOR_ID1: case RT5631_VENDOR_ID2: - return 1; + return true; default: - return 0; + return false; } } @@ -229,10 +229,10 @@ static SOC_ENUM_SINGLE_DECL(rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG, static const struct snd_kcontrol_new rt5631_snd_controls[] = { /* MIC */ SOC_ENUM("MIC1 Mode Control", rt5631_mic1_mode_enum), - SOC_SINGLE_TLV("MIC1 Boost", RT5631_MIC_CTRL_2, + SOC_SINGLE_TLV("MIC1 Boost Volume", RT5631_MIC_CTRL_2, RT5631_MIC1_BOOST_SHIFT, 8, 0, mic_bst_tlv), SOC_ENUM("MIC2 Mode Control", rt5631_mic2_mode_enum), - SOC_SINGLE_TLV("MIC2 Boost", RT5631_MIC_CTRL_2, + SOC_SINGLE_TLV("MIC2 Boost Volume", RT5631_MIC_CTRL_2, RT5631_MIC2_BOOST_SHIFT, 8, 0, mic_bst_tlv), /* MONO IN */ SOC_ENUM("MONOIN Mode Control", rt5631_monoin_mode_enum), diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 8bf8d360c25f..27770143ae8f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1665,6 +1665,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id) break; case RT5640_IF_113: ret |= RT5640_U_IF1; + /* fall through */ case RT5640_IF_312: case RT5640_IF_213: ret |= RT5640_U_IF2; @@ -1680,6 +1681,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id) break; case RT5640_IF_223: ret |= RT5640_U_IF1; + /* fall through */ case RT5640_IF_123: case RT5640_IF_321: ret |= RT5640_U_IF2; diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 6b5669f3e85d..985852fd9723 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -331,11 +331,13 @@ static const struct snd_kcontrol_new rt5651_snd_controls[] = { SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5651_DAC2_DIG_VOL, RT5651_L_VOL_SFT, RT5651_R_VOL_SFT, 175, 0, dac_vol_tlv), - /* IN1/IN2 Control */ + /* IN1/IN2/IN3 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5651_IN1_IN2, RT5651_BST_SFT1, 8, 0, bst_tlv), SOC_SINGLE_TLV("IN2 Boost", RT5651_IN1_IN2, RT5651_BST_SFT2, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN3 Boost", RT5651_IN3, + RT5651_BST_SFT1, 8, 0, bst_tlv), /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5651_INL1_INR1_VOL, RT5651_INL_VOL_SFT, RT5651_INR_VOL_SFT, @@ -1581,6 +1583,24 @@ static void rt5651_disable_micbias1_for_ovcd(struct snd_soc_component *component snd_soc_dapm_mutex_unlock(dapm); } +static void rt5651_enable_micbias1_ovcd_irq(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2, + RT5651_IRQ_MB1_OC_MASK, RT5651_IRQ_MB1_OC_NOR); + rt5651->ovcd_irq_enabled = true; +} + +static void rt5651_disable_micbias1_ovcd_irq(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2, + RT5651_IRQ_MB1_OC_MASK, RT5651_IRQ_MB1_OC_BP); + rt5651->ovcd_irq_enabled = false; +} + static void rt5651_clear_micbias1_ovcd(struct snd_soc_component *component) { snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2, @@ -1622,10 +1642,80 @@ static bool rt5651_jack_inserted(struct snd_soc_component *component) return val == 0; } -/* Jack detect timings */ +/* Jack detect and button-press timings */ #define JACK_SETTLE_TIME 100 /* milli seconds */ #define JACK_DETECT_COUNT 5 #define JACK_DETECT_MAXCOUNT 20 /* Aprox. 2 seconds worth of tries */ +#define JACK_UNPLUG_TIME 80 /* milli seconds */ +#define BP_POLL_TIME 10 /* milli seconds */ +#define BP_POLL_MAXCOUNT 200 /* assume something is wrong after this */ +#define BP_THRESHOLD 3 + +static void rt5651_start_button_press_work(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + rt5651->poll_count = 0; + rt5651->press_count = 0; + rt5651->release_count = 0; + rt5651->pressed = false; + rt5651->press_reported = false; + rt5651_clear_micbias1_ovcd(component); + schedule_delayed_work(&rt5651->bp_work, msecs_to_jiffies(BP_POLL_TIME)); +} + +static void rt5651_button_press_work(struct work_struct *work) +{ + struct rt5651_priv *rt5651 = + container_of(work, struct rt5651_priv, bp_work.work); + struct snd_soc_component *component = rt5651->component; + + /* Check the jack was not removed underneath us */ + if (!rt5651_jack_inserted(component)) + return; + + if (rt5651_micbias1_ovcd(component)) { + rt5651->release_count = 0; + rt5651->press_count++; + /* Remember till after JACK_UNPLUG_TIME wait */ + if (rt5651->press_count >= BP_THRESHOLD) + rt5651->pressed = true; + rt5651_clear_micbias1_ovcd(component); + } else { + rt5651->press_count = 0; + rt5651->release_count++; + } + + /* + * The pins get temporarily shorted on jack unplug, so we poll for + * at least JACK_UNPLUG_TIME milli-seconds before reporting a press. + */ + rt5651->poll_count++; + if (rt5651->poll_count < (JACK_UNPLUG_TIME / BP_POLL_TIME)) { + schedule_delayed_work(&rt5651->bp_work, + msecs_to_jiffies(BP_POLL_TIME)); + return; + } + + if (rt5651->pressed && !rt5651->press_reported) { + dev_dbg(component->dev, "headset button press\n"); + snd_soc_jack_report(rt5651->hp_jack, SND_JACK_BTN_0, + SND_JACK_BTN_0); + rt5651->press_reported = true; + } + + if (rt5651->release_count >= BP_THRESHOLD) { + if (rt5651->press_reported) { + dev_dbg(component->dev, "headset button release\n"); + snd_soc_jack_report(rt5651->hp_jack, 0, SND_JACK_BTN_0); + } + /* Re-enable OVCD IRQ to detect next press */ + rt5651_enable_micbias1_ovcd_irq(component); + return; /* Stop polling */ + } + + schedule_delayed_work(&rt5651->bp_work, msecs_to_jiffies(BP_POLL_TIME)); +} static int rt5651_detect_headset(struct snd_soc_component *component) { @@ -1676,15 +1766,58 @@ static void rt5651_jack_detect_work(struct work_struct *work) { struct rt5651_priv *rt5651 = container_of(work, struct rt5651_priv, jack_detect_work); + struct snd_soc_component *component = rt5651->component; int report = 0; - if (rt5651_jack_inserted(rt5651->component)) { - rt5651_enable_micbias1_for_ovcd(rt5651->component); - report = rt5651_detect_headset(rt5651->component); - rt5651_disable_micbias1_for_ovcd(rt5651->component); + if (!rt5651_jack_inserted(component)) { + /* Jack removed, or spurious IRQ? */ + if (rt5651->hp_jack->status & SND_JACK_HEADPHONE) { + if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) { + cancel_delayed_work_sync(&rt5651->bp_work); + rt5651_disable_micbias1_ovcd_irq(component); + rt5651_disable_micbias1_for_ovcd(component); + } + snd_soc_jack_report(rt5651->hp_jack, 0, + SND_JACK_HEADSET | SND_JACK_BTN_0); + dev_dbg(component->dev, "jack unplugged\n"); + } + } else if (!(rt5651->hp_jack->status & SND_JACK_HEADPHONE)) { + /* Jack inserted */ + WARN_ON(rt5651->ovcd_irq_enabled); + rt5651_enable_micbias1_for_ovcd(component); + report = rt5651_detect_headset(component); + if (report == SND_JACK_HEADSET) { + /* Enable ovcd IRQ for button press detect. */ + rt5651_enable_micbias1_ovcd_irq(component); + } else { + /* No more need for overcurrent detect. */ + rt5651_disable_micbias1_for_ovcd(component); + } + dev_dbg(component->dev, "detect report %#02x\n", report); + snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET); + } else if (rt5651->ovcd_irq_enabled && rt5651_micbias1_ovcd(component)) { + dev_dbg(component->dev, "OVCD IRQ\n"); + + /* + * The ovcd IRQ keeps firing while the button is pressed, so + * we disable it and start polling the button until released. + * + * The disable will make the IRQ pin 0 again and since we get + * IRQs on both edges (so as to detect both jack plugin and + * unplug) this means we will immediately get another IRQ. + * The ovcd_irq_enabled check above makes the 2ND IRQ a NOP. + */ + rt5651_disable_micbias1_ovcd_irq(component); + rt5651_start_button_press_work(component); + + /* + * If the jack-detect IRQ flag goes high (unplug) after our + * above rt5651_jack_inserted() check and before we have + * disabled the OVCD IRQ, the IRQ pin will stay high and as + * we react to edges, we miss the unplug event -> recheck. + */ + queue_work(system_long_wq, &rt5651->jack_detect_work); } - - snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET); } static irqreturn_t rt5651_irq(int irq, void *data) @@ -1696,14 +1829,18 @@ static irqreturn_t rt5651_irq(int irq, void *data) return IRQ_HANDLED; } -static int rt5651_set_jack(struct snd_soc_component *component, - struct snd_soc_jack *hp_jack, void *data) +static void rt5651_cancel_work(void *data) { - struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); - int ret; + struct rt5651_priv *rt5651 = data; - if (!rt5651->irq) - return -EINVAL; + cancel_work_sync(&rt5651->jack_detect_work); + cancel_delayed_work_sync(&rt5651->bp_work); +} + +static void rt5651_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hp_jack) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); /* IRQ output on GPIO1 */ snd_soc_component_update_bits(component, RT5651_GPIO_CTRL1, @@ -1730,10 +1867,10 @@ static int rt5651_set_jack(struct snd_soc_component *component, RT5651_JD2_IRQ_EN, RT5651_JD2_IRQ_EN); break; case RT5651_JD_NULL: - return 0; + return; default: dev_err(component->dev, "Currently only JD1_1 / JD1_2 / JD2 are supported\n"); - return -EINVAL; + return; } /* Enable jack detect power */ @@ -1767,19 +1904,39 @@ static int rt5651_set_jack(struct snd_soc_component *component, RT5651_MB1_OC_STKY_MASK, RT5651_MB1_OC_STKY_EN); rt5651->hp_jack = hp_jack; - - ret = devm_request_threaded_irq(component->dev, rt5651->irq, NULL, - rt5651_irq, - IRQF_TRIGGER_RISING | - IRQF_TRIGGER_FALLING | - IRQF_ONESHOT, "rt5651", rt5651); - if (ret) { - dev_err(component->dev, "Failed to reguest IRQ: %d\n", ret); - return ret; + if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) { + rt5651_enable_micbias1_for_ovcd(component); + rt5651_enable_micbias1_ovcd_irq(component); } + enable_irq(rt5651->irq); /* sync initial jack state */ queue_work(system_power_efficient_wq, &rt5651->jack_detect_work); +} + +static void rt5651_disable_jack_detect(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + disable_irq(rt5651->irq); + rt5651_cancel_work(rt5651); + + if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) { + rt5651_disable_micbias1_ovcd_irq(component); + rt5651_disable_micbias1_for_ovcd(component); + snd_soc_jack_report(rt5651->hp_jack, 0, SND_JACK_BTN_0); + } + + rt5651->hp_jack = NULL; +} + +static int rt5651_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + rt5651_enable_jack_detect(component, jack); + else + rt5651_disable_jack_detect(component); return 0; } @@ -2034,8 +2191,26 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, rt5651->irq = i2c->irq; rt5651->hp_mute = 1; + INIT_DELAYED_WORK(&rt5651->bp_work, rt5651_button_press_work); INIT_WORK(&rt5651->jack_detect_work, rt5651_jack_detect_work); + /* Make sure work is stopped on probe-error / remove */ + ret = devm_add_action_or_reset(&i2c->dev, rt5651_cancel_work, rt5651); + if (ret) + return ret; + + ret = devm_request_irq(&i2c->dev, rt5651->irq, rt5651_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5651", rt5651); + if (ret == 0) { + /* Gets re-enabled by rt5651_set_jack() */ + disable_irq(rt5651->irq); + } else { + dev_warn(&i2c->dev, "Failed to reguest IRQ %d: %d\n", + rt5651->irq, ret); + rt5651->irq = -ENXIO; + } + ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5651, rt5651_dai, ARRAY_SIZE(rt5651_dai)); @@ -2043,15 +2218,6 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, return ret; } -static int rt5651_i2c_remove(struct i2c_client *i2c) -{ - struct rt5651_priv *rt5651 = i2c_get_clientdata(i2c); - - cancel_work_sync(&rt5651->jack_detect_work); - - return 0; -} - static struct i2c_driver rt5651_i2c_driver = { .driver = { .name = "rt5651", @@ -2059,7 +2225,6 @@ static struct i2c_driver rt5651_i2c_driver = { .of_match_table = of_match_ptr(rt5651_of_match), }, .probe = rt5651_i2c_probe, - .remove = rt5651_i2c_remove, .id_table = rt5651_i2c_id, }; module_i2c_driver(rt5651_i2c_driver); diff --git a/sound/soc/codecs/rt5651.h b/sound/soc/codecs/rt5651.h index 3a0968c53fde..ac6de6fb5414 100644 --- a/sound/soc/codecs/rt5651.h +++ b/sound/soc/codecs/rt5651.h @@ -2071,8 +2071,16 @@ struct rt5651_pll_code { struct rt5651_priv { struct snd_soc_component *component; struct regmap *regmap; + /* Jack and button detect data */ struct snd_soc_jack *hp_jack; struct work_struct jack_detect_work; + struct delayed_work bp_work; + bool ovcd_irq_enabled; + bool pressed; + bool press_reported; + int press_count; + int release_count; + int poll_count; unsigned int jd_src; unsigned int ovcd_th; unsigned int ovcd_sf; diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8a0181a2db08..9b7a1833d331 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4417,6 +4417,7 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, break; case 25: slot_width_25 = 0x8080; + /* fall through */ case 24: val |= (2 << 8); break; @@ -5007,7 +5008,7 @@ static const struct regmap_config rt5677_regmap = { }; static const struct of_device_id rt5677_of_match[] = { - { .compatible = "realtek,rt5677", RT5677 }, + { .compatible = "realtek,rt5677", .data = (const void *)RT5677 }, { } }; MODULE_DEVICE_TABLE(of, rt5677_of_match); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c new file mode 100644 index 000000000000..640d400ca013 --- /dev/null +++ b/sound/soc/codecs/rt5682.c @@ -0,0 +1,2681 @@ +/* + * rt5682.c -- RT5682 ALSA SoC audio component driver + * + * Copyright 2018 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao@realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/mutex.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/rt5682.h> + +#include "rl6231.h" +#include "rt5682.h" + +#define RT5682_NUM_SUPPLIES 3 + +static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = { + "AVDD", + "MICVDD", + "VBAT", +}; + +struct rt5682_priv { + struct snd_soc_component *component; + struct rt5682_platform_data pdata; + struct regmap *regmap; + struct snd_soc_jack *hs_jack; + struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES]; + struct delayed_work jack_detect_work; + struct delayed_work jd_check_work; + struct mutex calibrate_mutex; + + int sysclk; + int sysclk_src; + int lrck[RT5682_AIFS]; + int bclk[RT5682_AIFS]; + int master[RT5682_AIFS]; + + int pll_src; + int pll_in; + int pll_out; + + int jack_type; +}; + +static const struct reg_sequence patch_list[] = { + {0x01c1, 0x1000}, +}; + +static const struct reg_default rt5682_reg[] = { + {0x0002, 0x8080}, + {0x0003, 0x8000}, + {0x0005, 0x0000}, + {0x0006, 0x0000}, + {0x0008, 0x800f}, + {0x000b, 0x0000}, + {0x0010, 0x4040}, + {0x0011, 0x0000}, + {0x0012, 0x1404}, + {0x0013, 0x1000}, + {0x0014, 0xa00a}, + {0x0015, 0x0404}, + {0x0016, 0x0404}, + {0x0019, 0xafaf}, + {0x001c, 0x2f2f}, + {0x001f, 0x0000}, + {0x0022, 0x5757}, + {0x0023, 0x0039}, + {0x0024, 0x000b}, + {0x0026, 0xc0c4}, + {0x0029, 0x8080}, + {0x002a, 0xa0a0}, + {0x002b, 0x0300}, + {0x0030, 0x0000}, + {0x003c, 0x0080}, + {0x0044, 0x0c0c}, + {0x0049, 0x0000}, + {0x0061, 0x0000}, + {0x0062, 0x0000}, + {0x0063, 0x003f}, + {0x0064, 0x0000}, + {0x0065, 0x0000}, + {0x0066, 0x0030}, + {0x0067, 0x0000}, + {0x006b, 0x0000}, + {0x006c, 0x0000}, + {0x006d, 0x2200}, + {0x006e, 0x0a10}, + {0x0070, 0x8000}, + {0x0071, 0x8000}, + {0x0073, 0x0000}, + {0x0074, 0x0000}, + {0x0075, 0x0002}, + {0x0076, 0x0001}, + {0x0079, 0x0000}, + {0x007a, 0x0000}, + {0x007b, 0x0000}, + {0x007c, 0x0100}, + {0x007e, 0x0000}, + {0x0080, 0x0000}, + {0x0081, 0x0000}, + {0x0082, 0x0000}, + {0x0083, 0x0000}, + {0x0084, 0x0000}, + {0x0085, 0x0000}, + {0x0086, 0x0005}, + {0x0087, 0x0000}, + {0x0088, 0x0000}, + {0x008c, 0x0003}, + {0x008d, 0x0000}, + {0x008e, 0x0060}, + {0x008f, 0x1000}, + {0x0091, 0x0c26}, + {0x0092, 0x0073}, + {0x0093, 0x0000}, + {0x0094, 0x0080}, + {0x0098, 0x0000}, + {0x009a, 0x0000}, + {0x009b, 0x0000}, + {0x009c, 0x0000}, + {0x009d, 0x0000}, + {0x009e, 0x100c}, + {0x009f, 0x0000}, + {0x00a0, 0x0000}, + {0x00a3, 0x0002}, + {0x00a4, 0x0001}, + {0x00ae, 0x2040}, + {0x00af, 0x0000}, + {0x00b6, 0x0000}, + {0x00b7, 0x0000}, + {0x00b8, 0x0000}, + {0x00b9, 0x0002}, + {0x00be, 0x0000}, + {0x00c0, 0x0160}, + {0x00c1, 0x82a0}, + {0x00c2, 0x0000}, + {0x00d0, 0x0000}, + {0x00d1, 0x2244}, + {0x00d2, 0x3300}, + {0x00d3, 0x2200}, + {0x00d4, 0x0000}, + {0x00d9, 0x0009}, + {0x00da, 0x0000}, + {0x00db, 0x0000}, + {0x00dc, 0x00c0}, + {0x00dd, 0x2220}, + {0x00de, 0x3131}, + {0x00df, 0x3131}, + {0x00e0, 0x3131}, + {0x00e2, 0x0000}, + {0x00e3, 0x4000}, + {0x00e4, 0x0aa0}, + {0x00e5, 0x3131}, + {0x00e6, 0x3131}, + {0x00e7, 0x3131}, + {0x00e8, 0x3131}, + {0x00ea, 0xb320}, + {0x00eb, 0x0000}, + {0x00f0, 0x0000}, + {0x00f1, 0x00d0}, + {0x00f2, 0x00d0}, + {0x00f6, 0x0000}, + {0x00fa, 0x0000}, + {0x00fb, 0x0000}, + {0x00fc, 0x0000}, + {0x00fd, 0x0000}, + {0x00fe, 0x10ec}, + {0x00ff, 0x6530}, + {0x0100, 0xa0a0}, + {0x010b, 0x0000}, + {0x010c, 0xae00}, + {0x010d, 0xaaa0}, + {0x010e, 0x8aa2}, + {0x010f, 0x02a2}, + {0x0110, 0xc000}, + {0x0111, 0x04a2}, + {0x0112, 0x2800}, + {0x0113, 0x0000}, + {0x0117, 0x0100}, + {0x0125, 0x0410}, + {0x0132, 0x6026}, + {0x0136, 0x5555}, + {0x0138, 0x3700}, + {0x013a, 0x2000}, + {0x013b, 0x2000}, + {0x013c, 0x2005}, + {0x013f, 0x0000}, + {0x0142, 0x0000}, + {0x0145, 0x0002}, + {0x0146, 0x0000}, + {0x0147, 0x0000}, + {0x0148, 0x0000}, + {0x0149, 0x0000}, + {0x0150, 0x79a1}, + {0x0151, 0x0000}, + {0x0160, 0x4ec0}, + {0x0161, 0x0080}, + {0x0162, 0x0200}, + {0x0163, 0x0800}, + {0x0164, 0x0000}, + {0x0165, 0x0000}, + {0x0166, 0x0000}, + {0x0167, 0x000f}, + {0x0168, 0x000f}, + {0x0169, 0x0021}, + {0x0190, 0x413d}, + {0x0194, 0x0000}, + {0x0195, 0x0000}, + {0x0197, 0x0022}, + {0x0198, 0x0000}, + {0x0199, 0x0000}, + {0x01af, 0x0000}, + {0x01b0, 0x0400}, + {0x01b1, 0x0000}, + {0x01b2, 0x0000}, + {0x01b3, 0x0000}, + {0x01b4, 0x0000}, + {0x01b5, 0x0000}, + {0x01b6, 0x01c3}, + {0x01b7, 0x02a0}, + {0x01b8, 0x03e9}, + {0x01b9, 0x1389}, + {0x01ba, 0xc351}, + {0x01bb, 0x0009}, + {0x01bc, 0x0018}, + {0x01bd, 0x002a}, + {0x01be, 0x004c}, + {0x01bf, 0x0097}, + {0x01c0, 0x433d}, + {0x01c2, 0x0000}, + {0x01c3, 0x0000}, + {0x01c4, 0x0000}, + {0x01c5, 0x0000}, + {0x01c6, 0x0000}, + {0x01c7, 0x0000}, + {0x01c8, 0x40af}, + {0x01c9, 0x0702}, + {0x01ca, 0x0000}, + {0x01cb, 0x0000}, + {0x01cc, 0x5757}, + {0x01cd, 0x5757}, + {0x01ce, 0x5757}, + {0x01cf, 0x5757}, + {0x01d0, 0x5757}, + {0x01d1, 0x5757}, + {0x01d2, 0x5757}, + {0x01d3, 0x5757}, + {0x01d4, 0x5757}, + {0x01d5, 0x5757}, + {0x01d6, 0x0000}, + {0x01d7, 0x0008}, + {0x01d8, 0x0029}, + {0x01d9, 0x3333}, + {0x01da, 0x0000}, + {0x01db, 0x0004}, + {0x01dc, 0x0000}, + {0x01de, 0x7c00}, + {0x01df, 0x0320}, + {0x01e0, 0x06a1}, + {0x01e1, 0x0000}, + {0x01e2, 0x0000}, + {0x01e3, 0x0000}, + {0x01e4, 0x0000}, + {0x01e6, 0x0001}, + {0x01e7, 0x0000}, + {0x01e8, 0x0000}, + {0x01ea, 0x0000}, + {0x01eb, 0x0000}, + {0x01ec, 0x0000}, + {0x01ed, 0x0000}, + {0x01ee, 0x0000}, + {0x01ef, 0x0000}, + {0x01f0, 0x0000}, + {0x01f1, 0x0000}, + {0x01f2, 0x0000}, + {0x01f3, 0x0000}, + {0x01f4, 0x0000}, + {0x0210, 0x6297}, + {0x0211, 0xa005}, + {0x0212, 0x824c}, + {0x0213, 0xf7ff}, + {0x0214, 0xf24c}, + {0x0215, 0x0102}, + {0x0216, 0x00a3}, + {0x0217, 0x0048}, + {0x0218, 0xa2c0}, + {0x0219, 0x0400}, + {0x021a, 0x00c8}, + {0x021b, 0x00c0}, + {0x021c, 0x0000}, + {0x0250, 0x4500}, + {0x0251, 0x40b3}, + {0x0252, 0x0000}, + {0x0253, 0x0000}, + {0x0254, 0x0000}, + {0x0255, 0x0000}, + {0x0256, 0x0000}, + {0x0257, 0x0000}, + {0x0258, 0x0000}, + {0x0259, 0x0000}, + {0x025a, 0x0005}, + {0x0270, 0x0000}, + {0x02ff, 0x0110}, + {0x0300, 0x001f}, + {0x0301, 0x032c}, + {0x0302, 0x5f21}, + {0x0303, 0x4000}, + {0x0304, 0x4000}, + {0x0305, 0x06d5}, + {0x0306, 0x8000}, + {0x0307, 0x0700}, + {0x0310, 0x4560}, + {0x0311, 0xa4a8}, + {0x0312, 0x7418}, + {0x0313, 0x0000}, + {0x0314, 0x0006}, + {0x0315, 0xffff}, + {0x0316, 0xc400}, + {0x0317, 0x0000}, + {0x03c0, 0x7e00}, + {0x03c1, 0x8000}, + {0x03c2, 0x8000}, + {0x03c3, 0x8000}, + {0x03c4, 0x8000}, + {0x03c5, 0x8000}, + {0x03c6, 0x8000}, + {0x03c7, 0x8000}, + {0x03c8, 0x8000}, + {0x03c9, 0x8000}, + {0x03ca, 0x8000}, + {0x03cb, 0x8000}, + {0x03cc, 0x8000}, + {0x03d0, 0x0000}, + {0x03d1, 0x0000}, + {0x03d2, 0x0000}, + {0x03d3, 0x0000}, + {0x03d4, 0x2000}, + {0x03d5, 0x2000}, + {0x03d6, 0x0000}, + {0x03d7, 0x0000}, + {0x03d8, 0x2000}, + {0x03d9, 0x2000}, + {0x03da, 0x2000}, + {0x03db, 0x2000}, + {0x03dc, 0x0000}, + {0x03dd, 0x0000}, + {0x03de, 0x0000}, + {0x03df, 0x2000}, + {0x03e0, 0x0000}, + {0x03e1, 0x0000}, + {0x03e2, 0x0000}, + {0x03e3, 0x0000}, + {0x03e4, 0x0000}, + {0x03e5, 0x0000}, + {0x03e6, 0x0000}, + {0x03e7, 0x0000}, + {0x03e8, 0x0000}, + {0x03e9, 0x0000}, + {0x03ea, 0x0000}, + {0x03eb, 0x0000}, + {0x03ec, 0x0000}, + {0x03ed, 0x0000}, + {0x03ee, 0x0000}, + {0x03ef, 0x0000}, + {0x03f0, 0x0800}, + {0x03f1, 0x0800}, + {0x03f2, 0x0800}, + {0x03f3, 0x0800}, +}; + +static bool rt5682_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5682_RESET: + case RT5682_CBJ_CTRL_2: + case RT5682_INT_ST_1: + case RT5682_4BTN_IL_CMD_1: + case RT5682_AJD1_CTRL: + case RT5682_HP_CALIB_CTRL_1: + case RT5682_DEVICE_ID: + case RT5682_I2C_MODE: + case RT5682_HP_CALIB_CTRL_10: + case RT5682_EFUSE_CTRL_2: + case RT5682_JD_TOP_VC_VTRL: + case RT5682_HP_IMP_SENS_CTRL_19: + case RT5682_IL_CMD_1: + case RT5682_SAR_IL_CMD_2: + case RT5682_SAR_IL_CMD_4: + case RT5682_SAR_IL_CMD_10: + case RT5682_SAR_IL_CMD_11: + case RT5682_EFUSE_CTRL_6...RT5682_EFUSE_CTRL_11: + case RT5682_HP_CALIB_STA_1...RT5682_HP_CALIB_STA_11: + return true; + default: + return false; + } +} + +static bool rt5682_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5682_RESET: + case RT5682_VERSION_ID: + case RT5682_VENDOR_ID: + case RT5682_DEVICE_ID: + case RT5682_HP_CTRL_1: + case RT5682_HP_CTRL_2: + case RT5682_HPL_GAIN: + case RT5682_HPR_GAIN: + case RT5682_I2C_CTRL: + case RT5682_CBJ_BST_CTRL: + case RT5682_CBJ_CTRL_1: + case RT5682_CBJ_CTRL_2: + case RT5682_CBJ_CTRL_3: + case RT5682_CBJ_CTRL_4: + case RT5682_CBJ_CTRL_5: + case RT5682_CBJ_CTRL_6: + case RT5682_CBJ_CTRL_7: + case RT5682_DAC1_DIG_VOL: + case RT5682_STO1_ADC_DIG_VOL: + case RT5682_STO1_ADC_BOOST: + case RT5682_HP_IMP_GAIN_1: + case RT5682_HP_IMP_GAIN_2: + case RT5682_SIDETONE_CTRL: + case RT5682_STO1_ADC_MIXER: + case RT5682_AD_DA_MIXER: + case RT5682_STO1_DAC_MIXER: + case RT5682_A_DAC1_MUX: + case RT5682_DIG_INF2_DATA: + case RT5682_REC_MIXER: + case RT5682_CAL_REC: + case RT5682_ALC_BACK_GAIN: + case RT5682_PWR_DIG_1: + case RT5682_PWR_DIG_2: + case RT5682_PWR_ANLG_1: + case RT5682_PWR_ANLG_2: + case RT5682_PWR_ANLG_3: + case RT5682_PWR_MIXER: + case RT5682_PWR_VOL: + case RT5682_CLK_DET: + case RT5682_RESET_LPF_CTRL: + case RT5682_RESET_HPF_CTRL: + case RT5682_DMIC_CTRL_1: + case RT5682_I2S1_SDP: + case RT5682_I2S2_SDP: + case RT5682_ADDA_CLK_1: + case RT5682_ADDA_CLK_2: + case RT5682_I2S1_F_DIV_CTRL_1: + case RT5682_I2S1_F_DIV_CTRL_2: + case RT5682_TDM_CTRL: + case RT5682_TDM_ADDA_CTRL_1: + case RT5682_TDM_ADDA_CTRL_2: + case RT5682_DATA_SEL_CTRL_1: + case RT5682_TDM_TCON_CTRL: + case RT5682_GLB_CLK: + case RT5682_PLL_CTRL_1: + case RT5682_PLL_CTRL_2: + case RT5682_PLL_TRACK_1: + case RT5682_PLL_TRACK_2: + case RT5682_PLL_TRACK_3: + case RT5682_PLL_TRACK_4: + case RT5682_PLL_TRACK_5: + case RT5682_PLL_TRACK_6: + case RT5682_PLL_TRACK_11: + case RT5682_SDW_REF_CLK: + case RT5682_DEPOP_1: + case RT5682_DEPOP_2: + case RT5682_HP_CHARGE_PUMP_1: + case RT5682_HP_CHARGE_PUMP_2: + case RT5682_MICBIAS_1: + case RT5682_MICBIAS_2: + case RT5682_PLL_TRACK_12: + case RT5682_PLL_TRACK_14: + case RT5682_PLL2_CTRL_1: + case RT5682_PLL2_CTRL_2: + case RT5682_PLL2_CTRL_3: + case RT5682_PLL2_CTRL_4: + case RT5682_RC_CLK_CTRL: + case RT5682_I2S_M_CLK_CTRL_1: + case RT5682_I2S2_F_DIV_CTRL_1: + case RT5682_I2S2_F_DIV_CTRL_2: + case RT5682_EQ_CTRL_1: + case RT5682_EQ_CTRL_2: + case RT5682_IRQ_CTRL_1: + case RT5682_IRQ_CTRL_2: + case RT5682_IRQ_CTRL_3: + case RT5682_IRQ_CTRL_4: + case RT5682_INT_ST_1: + case RT5682_GPIO_CTRL_1: + case RT5682_GPIO_CTRL_2: + case RT5682_GPIO_CTRL_3: + case RT5682_HP_AMP_DET_CTRL_1: + case RT5682_HP_AMP_DET_CTRL_2: + case RT5682_MID_HP_AMP_DET: + case RT5682_LOW_HP_AMP_DET: + case RT5682_DELAY_BUF_CTRL: + case RT5682_SV_ZCD_1: + case RT5682_SV_ZCD_2: + case RT5682_IL_CMD_1: + case RT5682_IL_CMD_2: + case RT5682_IL_CMD_3: + case RT5682_IL_CMD_4: + case RT5682_IL_CMD_5: + case RT5682_IL_CMD_6: + case RT5682_4BTN_IL_CMD_1: + case RT5682_4BTN_IL_CMD_2: + case RT5682_4BTN_IL_CMD_3: + case RT5682_4BTN_IL_CMD_4: + case RT5682_4BTN_IL_CMD_5: + case RT5682_4BTN_IL_CMD_6: + case RT5682_4BTN_IL_CMD_7: + case RT5682_ADC_STO1_HP_CTRL_1: + case RT5682_ADC_STO1_HP_CTRL_2: + case RT5682_AJD1_CTRL: + case RT5682_JD1_THD: + case RT5682_JD2_THD: + case RT5682_JD_CTRL_1: + case RT5682_DUMMY_1: + case RT5682_DUMMY_2: + case RT5682_DUMMY_3: + case RT5682_DAC_ADC_DIG_VOL1: + case RT5682_BIAS_CUR_CTRL_2: + case RT5682_BIAS_CUR_CTRL_3: + case RT5682_BIAS_CUR_CTRL_4: + case RT5682_BIAS_CUR_CTRL_5: + case RT5682_BIAS_CUR_CTRL_6: + case RT5682_BIAS_CUR_CTRL_7: + case RT5682_BIAS_CUR_CTRL_8: + case RT5682_BIAS_CUR_CTRL_9: + case RT5682_BIAS_CUR_CTRL_10: + case RT5682_VREF_REC_OP_FB_CAP_CTRL: + case RT5682_CHARGE_PUMP_1: + case RT5682_DIG_IN_CTRL_1: + case RT5682_PAD_DRIVING_CTRL: + case RT5682_SOFT_RAMP_DEPOP: + case RT5682_CHOP_DAC: + case RT5682_CHOP_ADC: + case RT5682_CALIB_ADC_CTRL: + case RT5682_VOL_TEST: + case RT5682_SPKVDD_DET_STA: + case RT5682_TEST_MODE_CTRL_1: + case RT5682_TEST_MODE_CTRL_2: + case RT5682_TEST_MODE_CTRL_3: + case RT5682_TEST_MODE_CTRL_4: + case RT5682_TEST_MODE_CTRL_5: + case RT5682_PLL1_INTERNAL: + case RT5682_PLL2_INTERNAL: + case RT5682_STO_NG2_CTRL_1: + case RT5682_STO_NG2_CTRL_2: + case RT5682_STO_NG2_CTRL_3: + case RT5682_STO_NG2_CTRL_4: + case RT5682_STO_NG2_CTRL_5: + case RT5682_STO_NG2_CTRL_6: + case RT5682_STO_NG2_CTRL_7: + case RT5682_STO_NG2_CTRL_8: + case RT5682_STO_NG2_CTRL_9: + case RT5682_STO_NG2_CTRL_10: + case RT5682_STO1_DAC_SIL_DET: + case RT5682_SIL_PSV_CTRL1: + case RT5682_SIL_PSV_CTRL2: + case RT5682_SIL_PSV_CTRL3: + case RT5682_SIL_PSV_CTRL4: + case RT5682_SIL_PSV_CTRL5: + case RT5682_HP_IMP_SENS_CTRL_01: + case RT5682_HP_IMP_SENS_CTRL_02: + case RT5682_HP_IMP_SENS_CTRL_03: + case RT5682_HP_IMP_SENS_CTRL_04: + case RT5682_HP_IMP_SENS_CTRL_05: + case RT5682_HP_IMP_SENS_CTRL_06: + case RT5682_HP_IMP_SENS_CTRL_07: + case RT5682_HP_IMP_SENS_CTRL_08: + case RT5682_HP_IMP_SENS_CTRL_09: + case RT5682_HP_IMP_SENS_CTRL_10: + case RT5682_HP_IMP_SENS_CTRL_11: + case RT5682_HP_IMP_SENS_CTRL_12: + case RT5682_HP_IMP_SENS_CTRL_13: + case RT5682_HP_IMP_SENS_CTRL_14: + case RT5682_HP_IMP_SENS_CTRL_15: + case RT5682_HP_IMP_SENS_CTRL_16: + case RT5682_HP_IMP_SENS_CTRL_17: + case RT5682_HP_IMP_SENS_CTRL_18: + case RT5682_HP_IMP_SENS_CTRL_19: + case RT5682_HP_IMP_SENS_CTRL_20: + case RT5682_HP_IMP_SENS_CTRL_21: + case RT5682_HP_IMP_SENS_CTRL_22: + case RT5682_HP_IMP_SENS_CTRL_23: + case RT5682_HP_IMP_SENS_CTRL_24: + case RT5682_HP_IMP_SENS_CTRL_25: + case RT5682_HP_IMP_SENS_CTRL_26: + case RT5682_HP_IMP_SENS_CTRL_27: + case RT5682_HP_IMP_SENS_CTRL_28: + case RT5682_HP_IMP_SENS_CTRL_29: + case RT5682_HP_IMP_SENS_CTRL_30: + case RT5682_HP_IMP_SENS_CTRL_31: + case RT5682_HP_IMP_SENS_CTRL_32: + case RT5682_HP_IMP_SENS_CTRL_33: + case RT5682_HP_IMP_SENS_CTRL_34: + case RT5682_HP_IMP_SENS_CTRL_35: + case RT5682_HP_IMP_SENS_CTRL_36: + case RT5682_HP_IMP_SENS_CTRL_37: + case RT5682_HP_IMP_SENS_CTRL_38: + case RT5682_HP_IMP_SENS_CTRL_39: + case RT5682_HP_IMP_SENS_CTRL_40: + case RT5682_HP_IMP_SENS_CTRL_41: + case RT5682_HP_IMP_SENS_CTRL_42: + case RT5682_HP_IMP_SENS_CTRL_43: + case RT5682_HP_LOGIC_CTRL_1: + case RT5682_HP_LOGIC_CTRL_2: + case RT5682_HP_LOGIC_CTRL_3: + case RT5682_HP_CALIB_CTRL_1: + case RT5682_HP_CALIB_CTRL_2: + case RT5682_HP_CALIB_CTRL_3: + case RT5682_HP_CALIB_CTRL_4: + case RT5682_HP_CALIB_CTRL_5: + case RT5682_HP_CALIB_CTRL_6: + case RT5682_HP_CALIB_CTRL_7: + case RT5682_HP_CALIB_CTRL_9: + case RT5682_HP_CALIB_CTRL_10: + case RT5682_HP_CALIB_CTRL_11: + case RT5682_HP_CALIB_STA_1: + case RT5682_HP_CALIB_STA_2: + case RT5682_HP_CALIB_STA_3: + case RT5682_HP_CALIB_STA_4: + case RT5682_HP_CALIB_STA_5: + case RT5682_HP_CALIB_STA_6: + case RT5682_HP_CALIB_STA_7: + case RT5682_HP_CALIB_STA_8: + case RT5682_HP_CALIB_STA_9: + case RT5682_HP_CALIB_STA_10: + case RT5682_HP_CALIB_STA_11: + case RT5682_SAR_IL_CMD_1: + case RT5682_SAR_IL_CMD_2: + case RT5682_SAR_IL_CMD_3: + case RT5682_SAR_IL_CMD_4: + case RT5682_SAR_IL_CMD_5: + case RT5682_SAR_IL_CMD_6: + case RT5682_SAR_IL_CMD_7: + case RT5682_SAR_IL_CMD_8: + case RT5682_SAR_IL_CMD_9: + case RT5682_SAR_IL_CMD_10: + case RT5682_SAR_IL_CMD_11: + case RT5682_SAR_IL_CMD_12: + case RT5682_SAR_IL_CMD_13: + case RT5682_EFUSE_CTRL_1: + case RT5682_EFUSE_CTRL_2: + case RT5682_EFUSE_CTRL_3: + case RT5682_EFUSE_CTRL_4: + case RT5682_EFUSE_CTRL_5: + case RT5682_EFUSE_CTRL_6: + case RT5682_EFUSE_CTRL_7: + case RT5682_EFUSE_CTRL_8: + case RT5682_EFUSE_CTRL_9: + case RT5682_EFUSE_CTRL_10: + case RT5682_EFUSE_CTRL_11: + case RT5682_JD_TOP_VC_VTRL: + case RT5682_DRC1_CTRL_0: + case RT5682_DRC1_CTRL_1: + case RT5682_DRC1_CTRL_2: + case RT5682_DRC1_CTRL_3: + case RT5682_DRC1_CTRL_4: + case RT5682_DRC1_CTRL_5: + case RT5682_DRC1_CTRL_6: + case RT5682_DRC1_HARD_LMT_CTRL_1: + case RT5682_DRC1_HARD_LMT_CTRL_2: + case RT5682_DRC1_PRIV_1: + case RT5682_DRC1_PRIV_2: + case RT5682_DRC1_PRIV_3: + case RT5682_DRC1_PRIV_4: + case RT5682_DRC1_PRIV_5: + case RT5682_DRC1_PRIV_6: + case RT5682_DRC1_PRIV_7: + case RT5682_DRC1_PRIV_8: + case RT5682_EQ_AUTO_RCV_CTRL1: + case RT5682_EQ_AUTO_RCV_CTRL2: + case RT5682_EQ_AUTO_RCV_CTRL3: + case RT5682_EQ_AUTO_RCV_CTRL4: + case RT5682_EQ_AUTO_RCV_CTRL5: + case RT5682_EQ_AUTO_RCV_CTRL6: + case RT5682_EQ_AUTO_RCV_CTRL7: + case RT5682_EQ_AUTO_RCV_CTRL8: + case RT5682_EQ_AUTO_RCV_CTRL9: + case RT5682_EQ_AUTO_RCV_CTRL10: + case RT5682_EQ_AUTO_RCV_CTRL11: + case RT5682_EQ_AUTO_RCV_CTRL12: + case RT5682_EQ_AUTO_RCV_CTRL13: + case RT5682_ADC_L_EQ_LPF1_A1: + case RT5682_R_EQ_LPF1_A1: + case RT5682_L_EQ_LPF1_H0: + case RT5682_R_EQ_LPF1_H0: + case RT5682_L_EQ_BPF1_A1: + case RT5682_R_EQ_BPF1_A1: + case RT5682_L_EQ_BPF1_A2: + case RT5682_R_EQ_BPF1_A2: + case RT5682_L_EQ_BPF1_H0: + case RT5682_R_EQ_BPF1_H0: + case RT5682_L_EQ_BPF2_A1: + case RT5682_R_EQ_BPF2_A1: + case RT5682_L_EQ_BPF2_A2: + case RT5682_R_EQ_BPF2_A2: + case RT5682_L_EQ_BPF2_H0: + case RT5682_R_EQ_BPF2_H0: + case RT5682_L_EQ_BPF3_A1: + case RT5682_R_EQ_BPF3_A1: + case RT5682_L_EQ_BPF3_A2: + case RT5682_R_EQ_BPF3_A2: + case RT5682_L_EQ_BPF3_H0: + case RT5682_R_EQ_BPF3_H0: + case RT5682_L_EQ_BPF4_A1: + case RT5682_R_EQ_BPF4_A1: + case RT5682_L_EQ_BPF4_A2: + case RT5682_R_EQ_BPF4_A2: + case RT5682_L_EQ_BPF4_H0: + case RT5682_R_EQ_BPF4_H0: + case RT5682_L_EQ_HPF1_A1: + case RT5682_R_EQ_HPF1_A1: + case RT5682_L_EQ_HPF1_H0: + case RT5682_R_EQ_HPF1_H0: + case RT5682_L_EQ_PRE_VOL: + case RT5682_R_EQ_PRE_VOL: + case RT5682_L_EQ_POST_VOL: + case RT5682_R_EQ_POST_VOL: + case RT5682_I2C_MODE: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static const DECLARE_TLV_DB_RANGE(bst_tlv, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0) +); + +/* Interface data select */ +static const char * const rt5682_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682_if2_adc_enum, + RT5682_DIG_INF2_DATA, RT5682_IF2_ADC_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_01_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC1_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_23_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC2_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_45_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC3_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_67_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC4_SEL_SFT, rt5682_data_select); + +static const struct snd_kcontrol_new rt5682_if2_adc_swap_mux = + SOC_DAPM_ENUM("IF2 ADC Swap Mux", rt5682_if2_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_01_adc_swap_mux = + SOC_DAPM_ENUM("IF1 01 ADC Swap Mux", rt5682_if1_01_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_23_adc_swap_mux = + SOC_DAPM_ENUM("IF1 23 ADC Swap Mux", rt5682_if1_23_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_45_adc_swap_mux = + SOC_DAPM_ENUM("IF1 45 ADC Swap Mux", rt5682_if1_45_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_67_adc_swap_mux = + SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682_if1_67_adc_enum); + +static void rt5682_reset(struct regmap *regmap) +{ + regmap_write(regmap, RT5682_RESET, 0); + regmap_write(regmap, RT5682_I2C_MODE, 1); +} +/** + * rt5682_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @component: SoC audio component device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5682 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the component driver will turn on + * ASRC for these filters if ASRC is selected as their clock source. + */ +int rt5682_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src) +{ + + switch (clk_src) { + case RT5682_CLK_SEL_SYS: + case RT5682_CLK_SEL_I2S1_ASRC: + case RT5682_CLK_SEL_I2S2_ASRC: + break; + + default: + return -EINVAL; + } + + if (filter_mask & RT5682_DA_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5682_PLL_TRACK_2, + RT5682_FILTER_CLK_SEL_MASK, + clk_src << RT5682_FILTER_CLK_SEL_SFT); + } + + if (filter_mask & RT5682_AD_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5682_PLL_TRACK_3, + RT5682_FILTER_CLK_SEL_MASK, + clk_src << RT5682_FILTER_CLK_SEL_SFT); + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5682_sel_asrc_clk_src); + +static int rt5682_button_detect(struct snd_soc_component *component) +{ + int btn_type, val; + + val = snd_soc_component_read32(component, RT5682_4BTN_IL_CMD_1); + btn_type = val & 0xfff0; + snd_soc_component_write(component, RT5682_4BTN_IL_CMD_1, val); + pr_debug("%s btn_type=%x\n", __func__, btn_type); + snd_soc_component_update_bits(component, + RT5682_SAR_IL_CMD_2, 0x10, 0x10); + + return btn_type; +} + +static void rt5682_enable_push_button_irq(struct snd_soc_component *component, + bool enable) +{ + if (enable) { + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_EN); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_13, + RT5682_SAR_SOUR_MASK, RT5682_SAR_SOUR_BTN); + snd_soc_component_write(component, RT5682_IL_CMD_1, 0x0040); + snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2, + RT5682_4BTN_IL_MASK | RT5682_4BTN_IL_RST_MASK, + RT5682_4BTN_IL_EN | RT5682_4BTN_IL_NOR); + snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3, + RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_EN); + } else { + snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3, + RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_DIS); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_DIS); + snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2, + RT5682_4BTN_IL_MASK, RT5682_4BTN_IL_DIS); + snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2, + RT5682_4BTN_IL_RST_MASK, RT5682_4BTN_IL_RST); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_13, + RT5682_SAR_SOUR_MASK, RT5682_SAR_SOUR_TYPE); + } +} + +/** + * rt5682_headset_detect - Detect headset. + * @component: SoC audio component device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +static int rt5682_headset_detect(struct snd_soc_component *component, + int jack_insert) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + unsigned int val, count; + + if (jack_insert) { + snd_soc_dapm_force_enable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_HIGH); + + count = 0; + val = snd_soc_component_read32(component, RT5682_CBJ_CTRL_2) + & RT5682_JACK_TYPE_MASK; + while (val == 0 && count < 50) { + usleep_range(10000, 15000); + val = snd_soc_component_read32(component, + RT5682_CBJ_CTRL_2) & RT5682_JACK_TYPE_MASK; + count++; + } + + switch (val) { + case 0x1: + case 0x2: + rt5682->jack_type = SND_JACK_HEADSET; + rt5682_enable_push_button_irq(component, true); + break; + default: + rt5682->jack_type = SND_JACK_HEADPHONE; + } + + } else { + rt5682_enable_push_button_irq(component, false); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); + snd_soc_dapm_disable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + + rt5682->jack_type = 0; + } + + dev_dbg(component->dev, "jack_type = %d\n", rt5682->jack_type); + return rt5682->jack_type; +} + +static irqreturn_t rt5682_irq(int irq, void *data) +{ + struct rt5682_priv *rt5682 = data; + + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(250)); + + return IRQ_HANDLED; +} + +static void rt5682_jd_check_handler(struct work_struct *work) +{ + struct rt5682_priv *rt5682 = container_of(work, struct rt5682_priv, + jd_check_work.work); + + if (snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL) + & RT5682_JDH_RS_MASK) { + /* jack out */ + rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0); + + snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + } else { + schedule_delayed_work(&rt5682->jd_check_work, 500); + } +} + +static int rt5682_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + switch (rt5682->pdata.jd_src) { + case RT5682_JD1: + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2, + RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); + snd_soc_component_write(component, RT5682_CBJ_CTRL_1, 0xd042); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_3, + RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_IRQ); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_IRQ | RT5682_POW_JDH | + RT5682_POW_ANA, RT5682_POW_IRQ | + RT5682_POW_JDH | RT5682_POW_ANA); + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2, + RT5682_PWR_JDH | RT5682_PWR_JDL, + RT5682_PWR_JDH | RT5682_PWR_JDL); + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK, + RT5682_JD1_EN | RT5682_JD1_POL_NOR); + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(250)); + break; + + case RT5682_JD_NULL: + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK, RT5682_JD1_DIS); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_JDH | RT5682_POW_JDL, 0); + break; + + default: + dev_warn(component->dev, "Wrong JD source\n"); + break; + } + + rt5682->hs_jack = hs_jack; + + return 0; +} + +static void rt5682_jack_detect_handler(struct work_struct *work) +{ + struct rt5682_priv *rt5682 = + container_of(work, struct rt5682_priv, jack_detect_work.work); + int val, btn_type; + + while (!rt5682->component) + usleep_range(10000, 15000); + + while (!rt5682->component->card->instantiated) + usleep_range(10000, 15000); + + mutex_lock(&rt5682->calibrate_mutex); + + val = snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL) + & RT5682_JDH_RS_MASK; + if (!val) { + /* jack in */ + if (rt5682->jack_type == 0) { + /* jack was out, report jack type */ + rt5682->jack_type = + rt5682_headset_detect(rt5682->component, 1); + } else { + /* jack is already in, report button event */ + rt5682->jack_type = SND_JACK_HEADSET; + btn_type = rt5682_button_detect(rt5682->component); + /** + * rt5682 can report three kinds of button behavior, + * one click, double click and hold. However, + * currently we will report button pressed/released + * event. So all the three button behaviors are + * treated as button pressed. + */ + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + rt5682->jack_type |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + rt5682->jack_type |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + rt5682->jack_type |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + rt5682->jack_type |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + btn_type = 0; + dev_err(rt5682->component->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + } else { + /* jack out */ + rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0); + } + + snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (rt5682->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3)) + schedule_delayed_work(&rt5682->jd_check_work, 0); + else + cancel_delayed_work_sync(&rt5682->jd_check_work); + + mutex_unlock(&rt5682->calibrate_mutex); +} + +static const struct snd_kcontrol_new rt5682_snd_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5682_HPL_GAIN, + RT5682_HPR_GAIN, RT5682_G_HP_SFT, 15, 1, hp_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL, + RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 175, 0, dac_vol_tlv), + + /* IN Boost Volume */ + SOC_SINGLE_TLV("CBJ Boost Volume", RT5682_CBJ_BST_CTRL, + RT5682_BST_CBJ_SFT, 8, 0, bst_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("STO1 ADC Capture Switch", RT5682_STO1_ADC_DIG_VOL, + RT5682_L_MUTE_SFT, RT5682_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("STO1 ADC Capture Volume", RT5682_STO1_ADC_DIG_VOL, + RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5682_STO1_ADC_BOOST, + RT5682_STO1_ADC_L_BST_SFT, RT5682_STO1_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), +}; + + +static int rt5682_div_sel(struct rt5682_priv *rt5682, + int target, const int div[], int size) +{ + int i; + + if (rt5682->sysclk < target) { + pr_err("sysclk rate %d is too low\n", + rt5682->sysclk); + return 0; + } + + for (i = 0; i < size - 1; i++) { + pr_info("div[%d]=%d\n", i, div[i]); + if (target * div[i] == rt5682->sysclk) + return i; + if (target * div[i + 1] > rt5682->sysclk) { + pr_err("can't find div for sysclk %d\n", + rt5682->sysclk); + return i; + } + } + + if (target * div[i] < rt5682->sysclk) + pr_err("sysclk rate %d is too high\n", + rt5682->sysclk); + + return size - 1; + +} + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + int idx = -EINVAL; + static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128}; + + idx = rt5682_div_sel(rt5682, 1500000, div, ARRAY_SIZE(div)); + + snd_soc_component_update_bits(component, RT5682_DMIC_CTRL_1, + RT5682_DMIC_CLK_MASK, idx << RT5682_DMIC_CLK_SFT); + + return 0; +} + +static int set_filter_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + int ref, val, reg, sft, mask, idx = -EINVAL; + static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; + static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48}; + + val = snd_soc_component_read32(component, RT5682_GPIO_CTRL_1) & + RT5682_GP4_PIN_MASK; + if (w->shift == RT5682_PWR_ADC_S1F_BIT && + val == RT5682_GP4_PIN_ADCDAT2) + ref = 256 * rt5682->lrck[RT5682_AIF2]; + else + ref = 256 * rt5682->lrck[RT5682_AIF1]; + + idx = rt5682_div_sel(rt5682, ref, div_f, ARRAY_SIZE(div_f)); + + if (w->shift == RT5682_PWR_ADC_S1F_BIT) { + reg = RT5682_PLL_TRACK_3; + sft = RT5682_ADC_OSR_SFT; + mask = RT5682_ADC_OSR_MASK; + } else { + reg = RT5682_PLL_TRACK_2; + sft = RT5682_DAC_OSR_SFT; + mask = RT5682_DAC_OSR_MASK; + } + + snd_soc_component_update_bits(component, reg, + RT5682_FILTER_CLK_DIV_MASK, idx << RT5682_FILTER_CLK_DIV_SFT); + + /* select over sample rate */ + for (idx = 0; idx < ARRAY_SIZE(div_o); idx++) { + if (rt5682->sysclk <= 12288000 * div_o[idx]) + break; + } + + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_1, + mask, idx << sft); + + return 0; +} + +static int is_sys_clk_from_pll1(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + val = snd_soc_component_read32(component, RT5682_GLB_CLK); + val &= RT5682_SCLK_SRC_MASK; + if (val == RT5682_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int is_using_asrc(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (w->shift) { + case RT5682_ADC_STO1_ASRC_SFT: + reg = RT5682_PLL_TRACK_3; + shift = RT5682_FILTER_CLK_SEL_SFT; + break; + case RT5682_DAC_STO1_ASRC_SFT: + reg = RT5682_PLL_TRACK_2; + shift = RT5682_FILTER_CLK_SEL_SFT; + break; + default: + return 0; + } + + val = (snd_soc_component_read32(component, reg) >> shift) & 0xf; + switch (val) { + case RT5682_CLK_SEL_I2S1_ASRC: + case RT5682_CLK_SEL_I2S2_ASRC: + return 1; + default: + return 0; + } + +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5682_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682_AD_DA_MIXER, + RT5682_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5682_AD_DA_MIXER, + RT5682_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682_AD_DA_MIXER, + RT5682_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5682_AD_DA_MIXER, + RT5682_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_sto1_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_L1_STO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_sto1_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_L1_STO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_R1_STO_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5682_rec1_l_mix[] = { + SOC_DAPM_SINGLE("CBJ Switch", RT5682_REC_MIXER, + RT5682_M_CBJ_RM1_L_SFT, 1, 1), +}; + +/* STO1 ADC1 Source */ +/* MX-26 [13] [5] */ +static const char * const rt5682_sto1_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc1l_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC1L_SRC_SFT, rt5682_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc1l_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682_sto1_adc1l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc1r_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC1R_SRC_SFT, rt5682_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc1r_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682_sto1_adc1r_enum); + +/* STO1 ADC Source */ +/* MX-26 [11:10] [3:2] */ +static const char * const rt5682_sto1_adc_src[] = { + "ADC1 L", "ADC1 R" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adcl_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADCL_SRC_SFT, rt5682_sto1_adc_src); + +static const struct snd_kcontrol_new rt5682_sto1_adcl_mux = + SOC_DAPM_ENUM("Stereo1 ADCL Source", rt5682_sto1_adcl_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adcr_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADCR_SRC_SFT, rt5682_sto1_adc_src); + +static const struct snd_kcontrol_new rt5682_sto1_adcr_mux = + SOC_DAPM_ENUM("Stereo1 ADCR Source", rt5682_sto1_adcr_enum); + +/* STO1 ADC2 Source */ +/* MX-26 [12] [4] */ +static const char * const rt5682_sto1_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc2l_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC2L_SRC_SFT, rt5682_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc2l_mux = + SOC_DAPM_ENUM("Stereo1 ADC2L Source", rt5682_sto1_adc2l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc2r_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC2R_SRC_SFT, rt5682_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc2r_mux = + SOC_DAPM_ENUM("Stereo1 ADC2R Source", rt5682_sto1_adc2r_enum); + +/* MX-79 [6:4] I2S1 ADC data location */ +static const unsigned int rt5682_if1_adc_slot_values[] = { + 0, + 2, + 4, + 6, +}; + +static const char * const rt5682_if1_adc_slot_src[] = { + "Slot 0", "Slot 2", "Slot 4", "Slot 6" +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_if1_adc_slot_enum, + RT5682_TDM_CTRL, RT5682_TDM_ADC_LCA_SFT, RT5682_TDM_ADC_LCA_MASK, + rt5682_if1_adc_slot_src, rt5682_if1_adc_slot_values); + +static const struct snd_kcontrol_new rt5682_if1_adc_slot_mux = + SOC_DAPM_ENUM("IF1 ADC Slot location", rt5682_if1_adc_slot_enum); + +/* Analog DAC L1 Source, Analog DAC R1 Source*/ +/* MX-2B [4], MX-2B [0]*/ +static const char * const rt5682_alg_dac1_src[] = { + "Stereo1 DAC Mixer", "DAC1" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_alg_dac_l1_enum, RT5682_A_DAC1_MUX, + RT5682_A_DACL1_SFT, rt5682_alg_dac1_src); + +static const struct snd_kcontrol_new rt5682_alg_dac_l1_mux = + SOC_DAPM_ENUM("Analog DAC L1 Source", rt5682_alg_dac_l1_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_alg_dac_r1_enum, RT5682_A_DAC1_MUX, + RT5682_A_DACR1_SFT, rt5682_alg_dac1_src); + +static const struct snd_kcontrol_new rt5682_alg_dac_r1_mux = + SOC_DAPM_ENUM("Analog DAC R1 Source", rt5682_alg_dac_r1_enum); + +/* Out Switch */ +static const struct snd_kcontrol_new hpol_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1, + RT5682_L_MUTE_SFT, 1, 1); +static const struct snd_kcontrol_new hpor_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1, + RT5682_R_MUTE_SFT, 1, 1); + +static int rt5682_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_write(component, + RT5682_HP_LOGIC_CTRL_2, 0x0012); + snd_soc_component_write(component, + RT5682_HP_CTRL_2, 0x6000); + snd_soc_component_update_bits(component, RT5682_STO_NG2_CTRL_1, + RT5682_NG2_EN_MASK, RT5682_NG2_EN); + snd_soc_component_update_bits(component, + RT5682_DEPOP_1, 0x60, 0x60); + break; + + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, + RT5682_DEPOP_1, 0x60, 0x0); + snd_soc_component_write(component, + RT5682_HP_CTRL_2, 0x0000); + break; + + default: + return 0; + } + + return 0; + +} + +static int set_dmic_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /*Add delay to avoid pop noise*/ + msleep(150); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5655_set_verf(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (w->shift) { + case RT5682_PWR_VREF1_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV1, 0); + break; + + case RT5682_PWR_VREF2_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); + break; + + default: + break; + } + break; + + case SND_SOC_DAPM_POST_PMU: + usleep_range(15000, 20000); + switch (w->shift) { + case RT5682_PWR_VREF1_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV1, + RT5682_PWR_FV1); + break; + + case RT5682_PWR_VREF2_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, + RT5682_PWR_FV2); + break; + + default: + break; + } + break; + + default: + return 0; + } + + return 0; +} + +static const unsigned int rt5682_adcdat_pin_values[] = { + 1, + 3, +}; + +static const char * const rt5682_adcdat_pin_select[] = { + "ADCDAT1", + "ADCDAT2", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_adcdat_pin_enum, + RT5682_GPIO_CTRL_1, RT5682_GP4_PIN_SFT, RT5682_GP4_PIN_MASK, + rt5682_adcdat_pin_select, rt5682_adcdat_pin_values); + +static const struct snd_kcontrol_new rt5682_adcdat_pin_ctrl = + SOC_DAPM_ENUM("ADCDAT", rt5682_adcdat_pin_enum); + +static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5682_PWR_ANLG_3, RT5682_PWR_LDO2_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", RT5682_PWR_ANLG_3, RT5682_PWR_PLL_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2B", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2B_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2F", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2F_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, + rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, + rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_DAC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_ADC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AD ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_AD_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DA ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_DA_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_DMIC_ASRC_SFT, 0, NULL, 0), + + /* Input Side */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5682_PWR_ANLG_2, RT5682_PWR_MB1_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5682_PWR_ANLG_2, RT5682_PWR_MB2_BIT, + 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + + SND_SOC_DAPM_INPUT("IN1P"), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5682_DMIC_CTRL_1, + RT5682_DMIC_1_EN_SFT, 0, set_dmic_power, SND_SOC_DAPM_POST_PMU), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, + 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("CBJ Power", RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ_BIT, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5682_rec1_l_mix, + ARRAY_SIZE(rt5682_rec1_l_mix)), + SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5682_PWR_ANLG_2, + RT5682_PWR_RM1_L_BIT, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1 L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC1 R", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SUPPLY("ADC1 L Power", RT5682_PWR_DIG_1, + RT5682_PWR_ADC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 R Power", RT5682_PWR_DIG_1, + RT5682_PWR_ADC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 clock", RT5682_CHOP_ADC, + RT5682_CKGEN_ADC1_SFT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc1l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc1r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc2l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc2r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adcl_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adcr_mux), + SND_SOC_DAPM_MUX("IF1_ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_adc_slot_mux), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5682_PWR_DIG_2, + RT5682_PWR_ADC_S1F_BIT, 0, set_filter_clk, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", RT5682_STO1_ADC_DIG_VOL, + RT5682_L_MUTE_SFT, 1, rt5682_sto1_adc_l_mix, + ARRAY_SIZE(rt5682_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682_STO1_ADC_DIG_VOL, + RT5682_R_MUTE_SFT, 1, rt5682_sto1_adc_r_mix, + ARRAY_SIZE(rt5682_sto1_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("BTN Detection Mode", RT5682_SAR_IL_CMD_1, + 14, 1, NULL, 0), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5682_PWR_DIG_1, RT5682_PWR_I2S1_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5682_PWR_DIG_1, RT5682_PWR_I2S2_BIT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_01_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 23 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_23_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 45 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_45_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 67 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_67_adc_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if2_adc_swap_mux), + + SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0, + &rt5682_adcdat_pin_ctrl), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, + RT5682_I2S1_SDP, RT5682_SEL_ADCDAT_SFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, + RT5682_I2S2_SDP, RT5682_I2S2_PIN_CFG_SFT, 1), + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5682_dac_l_mix, ARRAY_SIZE(rt5682_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5682_dac_r_mix, ARRAY_SIZE(rt5682_dac_r_mix)), + + /* DAC channel Mux */ + SND_SOC_DAPM_MUX("DAC L1 Source", SND_SOC_NOPM, 0, 0, + &rt5682_alg_dac_l1_mux), + SND_SOC_DAPM_MUX("DAC R1 Source", SND_SOC_NOPM, 0, 0, + &rt5682_alg_dac_r1_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5682_PWR_DIG_2, + RT5682_PWR_DAC_S1F_BIT, 0, set_filter_clk, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5682_sto1_dac_l_mix, ARRAY_SIZE(rt5682_sto1_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5682_sto1_dac_r_mix, ARRAY_SIZE(rt5682_sto1_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5682_PWR_DIG_1, + RT5682_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5682_PWR_DIG_1, + RT5682_PWR_DAC_R1_BIT, 0), + SND_SOC_DAPM_SUPPLY_S("DAC 1 Clock", 3, RT5682_CHOP_DAC, + RT5682_CKGEN_DAC1_SFT, 0, NULL, 0), + + /* HPO */ + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5682_hp_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU), + + SND_SOC_DAPM_SUPPLY("HP Amp L", RT5682_PWR_ANLG_1, + RT5682_PWR_HA_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1, + RT5682_PWR_HA_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1, + RT5682_PUMP_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1, + RT5682_CAPLESS_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("HPOL Playback", SND_SOC_NOPM, 0, 0, + &hpol_switch), + SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0, + &hpor_switch), + + /* CLK DET */ + SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682_CLK_DET, + RT5682_SYS_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET PLL1", RT5682_CLK_DET, + RT5682_PLL1_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET PLL2", RT5682_CLK_DET, + RT5682_PLL2_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET", RT5682_CLK_DET, + RT5682_POW_CLK_DET_SFT, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + +}; + +static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { + /*PLL*/ + {"ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + {"DAC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + + /*ASRC*/ + {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, + {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc}, + {"ADC STO1 ASRC", NULL, "AD ASRC"}, + {"ADC STO1 ASRC", NULL, "CLKDET"}, + {"DAC STO1 ASRC", NULL, "DA ASRC"}, + {"DAC STO1 ASRC", NULL, "CLKDET"}, + + /*Vref*/ + {"MICBIAS1", NULL, "Vref1"}, + {"MICBIAS1", NULL, "Vref2"}, + {"MICBIAS2", NULL, "Vref1"}, + {"MICBIAS2", NULL, "Vref2"}, + + {"CLKDET SYS", NULL, "CLKDET"}, + + {"IN1P", NULL, "LDO2"}, + + {"BST1 CBJ", NULL, "IN1P"}, + {"BST1 CBJ", NULL, "CBJ Power"}, + {"CBJ Power", NULL, "Vref2"}, + + {"RECMIX1L", "CBJ Switch", "BST1 CBJ"}, + {"RECMIX1L", NULL, "RECMIX1L Power"}, + + {"ADC1 L", NULL, "RECMIX1L"}, + {"ADC1 L", NULL, "ADC1 L Power"}, + {"ADC1 L", NULL, "ADC1 clock"}, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC1 Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC1 Power"}, + {"DMIC CLK", NULL, "DMIC ASRC"}, + + {"Stereo1 ADC L Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC L Mux", "ADC1 R", "ADC1 R"}, + {"Stereo1 ADC R Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC R Mux", "ADC1 R", "ADC1 R"}, + + {"Stereo1 ADC L1 Mux", "ADC", "Stereo1 ADC L Mux"}, + {"Stereo1 ADC L1 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + {"Stereo1 ADC L2 Mux", "DMIC", "DMIC L1"}, + {"Stereo1 ADC L2 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + + {"Stereo1 ADC R1 Mux", "ADC", "Stereo1 ADC R Mux"}, + {"Stereo1 ADC R1 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + {"Stereo1 ADC R2 Mux", "DMIC", "DMIC R1"}, + {"Stereo1 ADC R2 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + + {"Stereo1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux"}, + {"Stereo1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux"}, + {"Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux"}, + {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, + {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"}, + + {"ADC Stereo1 Filter", NULL, "BTN Detection Mode"}, + + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"}, + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"}, + + {"IF1 01 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + + {"IF1_ADC Mux", "Slot 0", "IF1 01 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"}, + {"IF1_ADC Mux", NULL, "I2S1"}, + {"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"}, + {"AIF1TX", NULL, "ADCDAT Mux"}, + {"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"}, + {"AIF2TX", NULL, "ADCDAT Mux"}, + + {"IF1 DAC1 L", NULL, "AIF1RX"}, + {"IF1 DAC1 L", NULL, "I2S1"}, + {"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"}, + {"IF1 DAC1 R", NULL, "AIF1RX"}, + {"IF1 DAC1 R", NULL, "I2S1"}, + {"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"}, + + {"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"}, + {"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"}, + {"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"}, + {"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"}, + + {"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"}, + {"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"}, + + {"Stereo1 DAC MIXR", "DAC R1 Switch", "DAC1 MIXR"}, + {"Stereo1 DAC MIXR", "DAC L1 Switch", "DAC1 MIXL"}, + + {"DAC L1 Source", "DAC1", "DAC1 MIXL"}, + {"DAC L1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXL"}, + {"DAC R1 Source", "DAC1", "DAC1 MIXR"}, + {"DAC R1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXR"}, + + {"DAC L1", NULL, "DAC L1 Source"}, + {"DAC R1", NULL, "DAC R1 Source"}, + + {"DAC L1", NULL, "DAC 1 Clock"}, + {"DAC R1", NULL, "DAC 1 Clock"}, + + {"HP Amp", NULL, "DAC L1"}, + {"HP Amp", NULL, "DAC R1"}, + {"HP Amp", NULL, "HP Amp L"}, + {"HP Amp", NULL, "HP Amp R"}, + {"HP Amp", NULL, "Capless"}, + {"HP Amp", NULL, "Charge Pump"}, + {"HP Amp", NULL, "CLKDET SYS"}, + {"HP Amp", NULL, "CBJ Power"}, + {"HP Amp", NULL, "Vref2"}, + {"HPOL Playback", "Switch", "HP Amp"}, + {"HPOR Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HPOL Playback"}, + {"HPOR", NULL, "HPOR Playback"}, +}; + +static int rt5682_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int cl, val = 0; + + if (tx_mask || rx_mask) + snd_soc_component_update_bits(component, RT5682_TDM_ADDA_CTRL_2, + RT5682_TDM_EN, RT5682_TDM_EN); + else + snd_soc_component_update_bits(component, RT5682_TDM_ADDA_CTRL_2, + RT5682_TDM_EN, 0); + + switch (slots) { + case 4: + val |= RT5682_TDM_TX_CH_4; + val |= RT5682_TDM_RX_CH_4; + break; + case 6: + val |= RT5682_TDM_TX_CH_6; + val |= RT5682_TDM_RX_CH_6; + break; + case 8: + val |= RT5682_TDM_TX_CH_8; + val |= RT5682_TDM_RX_CH_8; + break; + case 2: + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682_TDM_CTRL, + RT5682_TDM_TX_CH_MASK | RT5682_TDM_RX_CH_MASK, val); + + switch (slot_width) { + case 8: + if (tx_mask || rx_mask) + return -EINVAL; + cl = RT5682_I2S1_TX_CHL_8 | RT5682_I2S1_RX_CHL_8; + break; + case 16: + val = RT5682_TDM_CL_16; + cl = RT5682_I2S1_TX_CHL_16 | RT5682_I2S1_RX_CHL_16; + break; + case 20: + val = RT5682_TDM_CL_20; + cl = RT5682_I2S1_TX_CHL_20 | RT5682_I2S1_RX_CHL_20; + break; + case 24: + val = RT5682_TDM_CL_24; + cl = RT5682_I2S1_TX_CHL_24 | RT5682_I2S1_RX_CHL_24; + break; + case 32: + val = RT5682_TDM_CL_32; + cl = RT5682_I2S1_TX_CHL_32 | RT5682_I2S1_RX_CHL_32; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_CL_MASK, val); + snd_soc_component_update_bits(component, RT5682_I2S1_SDP, + RT5682_I2S1_TX_CHL_MASK | RT5682_I2S1_RX_CHL_MASK, cl); + + return 0; +} + + +static int rt5682_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int len_1 = 0, len_2 = 0; + int pre_div, frame_size; + + rt5682->lrck[dai->id] = params_rate(params); + pre_div = rl6231_get_clk_info(rt5682->sysclk, rt5682->lrck[dai->id]); + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(component->dev, "Unsupported frame size: %d\n", + frame_size); + return -EINVAL; + } + + dev_dbg(dai->dev, "lrck is %dHz and pre_div is %d for iis %d\n", + rt5682->lrck[dai->id], pre_div, dai->id); + + switch (params_width(params)) { + case 16: + break; + case 20: + len_1 |= RT5682_I2S1_DL_20; + len_2 |= RT5682_I2S2_DL_20; + break; + case 24: + len_1 |= RT5682_I2S1_DL_24; + len_2 |= RT5682_I2S2_DL_24; + break; + case 32: + len_1 |= RT5682_I2S1_DL_32; + len_2 |= RT5682_I2S2_DL_24; + break; + case 8: + len_1 |= RT5682_I2S2_DL_8; + len_2 |= RT5682_I2S2_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5682_AIF1: + snd_soc_component_update_bits(component, RT5682_I2S1_SDP, + RT5682_I2S1_DL_MASK, len_1); + if (rt5682->master[RT5682_AIF1]) { + snd_soc_component_update_bits(component, + RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK, + pre_div << RT5682_I2S_M_DIV_SFT); + } + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, + RT5682_I2S1_SDP, RT5682_I2S1_MONO_MASK, + RT5682_I2S1_MONO_EN); + else + snd_soc_component_update_bits(component, + RT5682_I2S1_SDP, RT5682_I2S1_MONO_MASK, + RT5682_I2S1_MONO_DIS); + break; + case RT5682_AIF2: + snd_soc_component_update_bits(component, RT5682_I2S2_SDP, + RT5682_I2S2_DL_MASK, len_2); + if (rt5682->master[RT5682_AIF2]) { + snd_soc_component_update_bits(component, + RT5682_I2S_M_CLK_CTRL_1, RT5682_I2S2_M_PD_MASK, + pre_div << RT5682_I2S2_M_PD_SFT); + } + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, + RT5682_I2S2_SDP, RT5682_I2S2_MONO_MASK, + RT5682_I2S2_MONO_EN); + else + snd_soc_component_update_bits(component, + RT5682_I2S2_SDP, RT5682_I2S2_MONO_MASK, + RT5682_I2S2_MONO_DIS); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5682_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, tdm_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5682->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt5682->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5682_I2S_BP_INV; + tdm_ctrl |= RT5682_TDM_S_BP_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + if (dai->id == RT5682_AIF1) + tdm_ctrl |= RT5682_TDM_S_LP_INV | RT5682_TDM_M_BP_INV; + else + return -EINVAL; + break; + case SND_SOC_DAIFMT_IB_IF: + if (dai->id == RT5682_AIF1) + tdm_ctrl |= RT5682_TDM_S_BP_INV | RT5682_TDM_S_LP_INV | + RT5682_TDM_M_BP_INV | RT5682_TDM_M_LP_INV; + else + return -EINVAL; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5682_I2S_DF_LEFT; + tdm_ctrl |= RT5682_TDM_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5682_I2S_DF_PCM_A; + tdm_ctrl |= RT5682_TDM_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5682_I2S_DF_PCM_B; + tdm_ctrl |= RT5682_TDM_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5682_AIF1: + snd_soc_component_update_bits(component, RT5682_I2S1_SDP, + RT5682_I2S_DF_MASK, reg_val); + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_MS_MASK | RT5682_TDM_S_BP_MASK | + RT5682_TDM_DF_MASK | RT5682_TDM_M_BP_MASK | + RT5682_TDM_M_LP_MASK | RT5682_TDM_S_LP_MASK, + tdm_ctrl | rt5682->master[dai->id]); + break; + case RT5682_AIF2: + if (rt5682->master[dai->id] == 0) + reg_val |= RT5682_I2S2_MS_S; + snd_soc_component_update_bits(component, RT5682_I2S2_SDP, + RT5682_I2S2_MS_MASK | RT5682_I2S_BP_MASK | + RT5682_I2S_DF_MASK, reg_val); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5682_set_component_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, src = 0; + + if (freq == rt5682->sysclk && clk_id == rt5682->sysclk_src) + return 0; + + switch (clk_id) { + case RT5682_SCLK_S_MCLK: + reg_val |= RT5682_SCLK_SRC_MCLK; + src = RT5682_CLK_SRC_MCLK; + break; + case RT5682_SCLK_S_PLL1: + reg_val |= RT5682_SCLK_SRC_PLL1; + src = RT5682_CLK_SRC_PLL1; + break; + case RT5682_SCLK_S_PLL2: + reg_val |= RT5682_SCLK_SRC_PLL2; + src = RT5682_CLK_SRC_PLL2; + break; + case RT5682_SCLK_S_RCCLK: + reg_val |= RT5682_SCLK_SRC_RCCLK; + src = RT5682_CLK_SRC_RCCLK; + break; + default: + dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_SCLK_SRC_MASK, reg_val); + + if (rt5682->master[RT5682_AIF2]) { + snd_soc_component_update_bits(component, + RT5682_I2S_M_CLK_CTRL_1, RT5682_I2S2_SRC_MASK, + src << RT5682_I2S2_SRC_SFT); + } + + rt5682->sysclk = freq; + rt5682->sysclk_src = clk_id; + + dev_dbg(component->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; +} + +static int rt5682_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt5682->pll_src && freq_in == rt5682->pll_in && + freq_out == rt5682->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(component->dev, "PLL disabled\n"); + + rt5682->pll_in = 0; + rt5682->pll_out = 0; + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_SCLK_SRC_MASK, RT5682_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5682_PLL1_S_MCLK: + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_MCLK); + break; + case RT5682_PLL1_S_BCLK1: + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_BCLK1); + break; + default: + dev_err(component->dev, "Unknown PLL Source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_component_write(component, RT5682_PLL_CTRL_1, + pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code); + snd_soc_component_write(component, RT5682_PLL_CTRL_2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT | + pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST); + + rt5682->pll_in = freq_in; + rt5682->pll_out = freq_out; + rt5682->pll_src = source; + + return 0; +} + +static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + rt5682->bclk[dai->id] = ratio; + + switch (ratio) { + case 64: + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_2, + RT5682_I2S2_BCLK_MS2_MASK, + RT5682_I2S2_BCLK_MS2_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_2, + RT5682_I2S2_BCLK_MS2_MASK, + RT5682_I2S2_BCLK_MS2_32); + break; + default: + dev_err(dai->dev, "Invalid bclk ratio %d\n", ratio); + return -EINVAL; + } + + return 0; +} + +static int rt5682_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_PWR_MB | RT5682_PWR_BG, + RT5682_PWR_MB | RT5682_PWR_BG); + regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1, + RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO, + RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO); + break; + + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_PWR_MB, RT5682_PWR_MB); + regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1, + RT5682_DIG_GATE_CTRL, RT5682_DIG_GATE_CTRL); + break; + case SND_SOC_BIAS_OFF: + regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1, + RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO, 0); + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_PWR_MB | RT5682_PWR_BG, 0); + break; + + default: + break; + } + + return 0; +} + +static int rt5682_probe(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + rt5682->component = component; + + return 0; +} + +static void rt5682_remove(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + rt5682_reset(rt5682->regmap); +} + +#ifdef CONFIG_PM +static int rt5682_suspend(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt5682->regmap, true); + regcache_mark_dirty(rt5682->regmap); + return 0; +} + +static int rt5682_resume(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt5682->regmap, false); + regcache_sync(rt5682->regmap); + + return 0; +} +#else +#define rt5682_suspend NULL +#define rt5682_resume NULL +#endif + +#define RT5682_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5682_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt5682_aif1_dai_ops = { + .hw_params = rt5682_hw_params, + .set_fmt = rt5682_set_dai_fmt, + .set_tdm_slot = rt5682_set_tdm_slot, +}; + +static const struct snd_soc_dai_ops rt5682_aif2_dai_ops = { + .hw_params = rt5682_hw_params, + .set_fmt = rt5682_set_dai_fmt, + .set_bclk_ratio = rt5682_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt5682_dai[] = { + { + .name = "rt5682-aif1", + .id = RT5682_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .ops = &rt5682_aif1_dai_ops, + }, + { + .name = "rt5682-aif2", + .id = RT5682_AIF2, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .ops = &rt5682_aif2_dai_ops, + }, +}; + +static const struct snd_soc_component_driver soc_component_dev_rt5682 = { + .probe = rt5682_probe, + .remove = rt5682_remove, + .suspend = rt5682_suspend, + .resume = rt5682_resume, + .set_bias_level = rt5682_set_bias_level, + .controls = rt5682_snd_controls, + .num_controls = ARRAY_SIZE(rt5682_snd_controls), + .dapm_widgets = rt5682_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5682_dapm_widgets), + .dapm_routes = rt5682_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5682_dapm_routes), + .set_sysclk = rt5682_set_component_sysclk, + .set_pll = rt5682_set_component_pll, + .set_jack = rt5682_set_jack_detect, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config rt5682_regmap = { + .reg_bits = 16, + .val_bits = 16, + .max_register = RT5682_I2C_MODE, + .volatile_reg = rt5682_volatile_register, + .readable_reg = rt5682_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5682_reg, + .num_reg_defaults = ARRAY_SIZE(rt5682_reg), + .use_single_rw = true, +}; + +static const struct i2c_device_id rt5682_i2c_id[] = { + {"rt5682", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt5682_i2c_id); + +static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) +{ + + device_property_read_u32(dev, "realtek,dmic1-data-pin", + &rt5682->pdata.dmic1_data_pin); + device_property_read_u32(dev, "realtek,dmic1-clk-pin", + &rt5682->pdata.dmic1_clk_pin); + device_property_read_u32(dev, "realtek,jd-src", + &rt5682->pdata.jd_src); + + rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node, + "realtek,ldo1-en-gpios", 0); + + return 0; +} + +static void rt5682_calibrate(struct rt5682_priv *rt5682) +{ + int value, count; + + mutex_lock(&rt5682->calibrate_mutex); + + rt5682_reset(rt5682->regmap); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf); + usleep_range(15000, 20000); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); + regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8001); + regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + regmap_write(rt5682->regmap, RT5682_STO1_DAC_MIXER, 0x2080); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x4040); + regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0069); + regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); + regmap_write(rt5682->regmap, RT5682_HP_CTRL_2, 0x6000); + regmap_write(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, 0x0f26); + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7f05); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); + regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_9, 0x000f); + regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8d01); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); + regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1); + regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311); + regmap_write(rt5682->regmap, RT5682_RESET_HPF_CTRL, 0x0000); + regmap_write(rt5682->regmap, RT5682_ADC_STO1_HP_CTRL_1, 0x3320); + + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00); + + for (count = 0; count < 60; count++) { + regmap_read(rt5682->regmap, RT5682_HP_CALIB_STA_1, &value); + if (!(value & 0x8000)) + break; + + usleep_range(10000, 10005); + } + + if (count >= 60) + pr_err("HP Calibration Failure\n"); + + /* restore settings */ + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); + regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); + + mutex_unlock(&rt5682->calibrate_mutex); + +} + +static int rt5682_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5682_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5682_priv *rt5682; + int i, ret; + unsigned int val; + + rt5682 = devm_kzalloc(&i2c->dev, sizeof(struct rt5682_priv), + GFP_KERNEL); + + if (rt5682 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5682); + + if (pdata) + rt5682->pdata = *pdata; + else + rt5682_parse_dt(rt5682, &i2c->dev); + + rt5682->regmap = devm_regmap_init_i2c(i2c, &rt5682_regmap); + if (IS_ERR(rt5682->regmap)) { + ret = PTR_ERR(rt5682->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(rt5682->supplies); i++) + rt5682->supplies[i].supply = rt5682_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(rt5682->supplies), + rt5682->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(rt5682->supplies), + rt5682->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + if (gpio_is_valid(rt5682->pdata.ldo1_en)) { + if (devm_gpio_request_one(&i2c->dev, rt5682->pdata.ldo1_en, + GPIOF_OUT_INIT_HIGH, "rt5682")) + dev_err(&i2c->dev, "Fail gpio_request gpio_ldo\n"); + } + + /* Sleep for 300 ms miniumum */ + usleep_range(300000, 350000); + + regmap_write(rt5682->regmap, RT5682_I2C_MODE, 0x1); + usleep_range(10000, 15000); + + regmap_read(rt5682->regmap, RT5682_DEVICE_ID, &val); + if (val != DEVICE_ID) { + pr_err("Device with ID register %x is not rt5682\n", val); + return -ENODEV; + } + + rt5682_reset(rt5682->regmap); + + rt5682_calibrate(rt5682); + + ret = regmap_register_patch(rt5682->regmap, patch_list, + ARRAY_SIZE(patch_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0000); + + /* DMIC pin*/ + if (rt5682->pdata.dmic1_data_pin != RT5682_DMIC1_NULL) { + switch (rt5682->pdata.dmic1_data_pin) { + case RT5682_DMIC1_DATA_GPIO2: /* share with LRCK2 */ + regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1, + RT5682_DMIC_1_DP_MASK, RT5682_DMIC_1_DP_GPIO2); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP2_PIN_MASK, RT5682_GP2_PIN_DMIC_SDA); + break; + + case RT5682_DMIC1_DATA_GPIO5: /* share with DACDAT1 */ + regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1, + RT5682_DMIC_1_DP_MASK, RT5682_DMIC_1_DP_GPIO5); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP5_PIN_MASK, RT5682_GP5_PIN_DMIC_SDA); + break; + + default: + dev_warn(&i2c->dev, "invalid DMIC_DAT pin\n"); + break; + } + + switch (rt5682->pdata.dmic1_clk_pin) { + case RT5682_DMIC1_CLK_GPIO1: /* share with IRQ */ + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_DMIC_CLK); + break; + + case RT5682_DMIC1_CLK_GPIO3: /* share with BCLK2 */ + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP3_PIN_MASK, RT5682_GP3_PIN_DMIC_CLK); + break; + + default: + dev_warn(&i2c->dev, "invalid DMIC_CLK pin\n"); + break; + } + } + + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_LDO1_DVO_MASK | RT5682_HP_DRIVER_MASK, + RT5682_LDO1_DVO_12 | RT5682_HP_DRIVER_5X); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK, + RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1); + regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + + INIT_DELAYED_WORK(&rt5682->jack_detect_work, + rt5682_jack_detect_handler); + INIT_DELAYED_WORK(&rt5682->jd_check_work, + rt5682_jd_check_handler); + + mutex_init(&rt5682->calibrate_mutex); + + if (i2c->irq) { + ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, + rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5682", rt5682); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + + } + + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_rt5682, + rt5682_dai, ARRAY_SIZE(rt5682_dai)); +} + +static void rt5682_i2c_shutdown(struct i2c_client *client) +{ + struct rt5682_priv *rt5682 = i2c_get_clientdata(client); + + rt5682_reset(rt5682->regmap); +} + +#ifdef CONFIG_OF +static const struct of_device_id rt5682_of_match[] = { + {.compatible = "realtek,rt5682i"}, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5682_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt5682_acpi_match[] = { + {"10EC5682", 0,}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5682_acpi_match); +#endif + +static struct i2c_driver rt5682_i2c_driver = { + .driver = { + .name = "rt5682", + .of_match_table = of_match_ptr(rt5682_of_match), + .acpi_match_table = ACPI_PTR(rt5682_acpi_match), + }, + .probe = rt5682_i2c_probe, + .shutdown = rt5682_i2c_shutdown, + .id_table = rt5682_i2c_id, +}; +module_i2c_driver(rt5682_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5682 driver"); +MODULE_AUTHOR("Bard Liao <bardliao@realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h new file mode 100644 index 000000000000..8068140ebe3f --- /dev/null +++ b/sound/soc/codecs/rt5682.h @@ -0,0 +1,1324 @@ +/* + * rt5682.h -- RT5682/RT5658 ALSA SoC audio driver + * + * Copyright 2018 Realtek Microelectronics + * Author: Bard Liao <bardliao@realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5682_H__ +#define __RT5682_H__ + +#include <sound/rt5682.h> + +#define DEVICE_ID 0x6530 + +/* Info */ +#define RT5682_RESET 0x0000 +#define RT5682_VERSION_ID 0x00fd +#define RT5682_VENDOR_ID 0x00fe +#define RT5682_DEVICE_ID 0x00ff +/* I/O - Output */ +#define RT5682_HP_CTRL_1 0x0002 +#define RT5682_HP_CTRL_2 0x0003 +#define RT5682_HPL_GAIN 0x0005 +#define RT5682_HPR_GAIN 0x0006 + +#define RT5682_I2C_CTRL 0x0008 + +/* I/O - Input */ +#define RT5682_CBJ_BST_CTRL 0x000b +#define RT5682_CBJ_CTRL_1 0x0010 +#define RT5682_CBJ_CTRL_2 0x0011 +#define RT5682_CBJ_CTRL_3 0x0012 +#define RT5682_CBJ_CTRL_4 0x0013 +#define RT5682_CBJ_CTRL_5 0x0014 +#define RT5682_CBJ_CTRL_6 0x0015 +#define RT5682_CBJ_CTRL_7 0x0016 +/* I/O - ADC/DAC/DMIC */ +#define RT5682_DAC1_DIG_VOL 0x0019 +#define RT5682_STO1_ADC_DIG_VOL 0x001c +#define RT5682_STO1_ADC_BOOST 0x001f +#define RT5682_HP_IMP_GAIN_1 0x0022 +#define RT5682_HP_IMP_GAIN_2 0x0023 +/* Mixer - D-D */ +#define RT5682_SIDETONE_CTRL 0x0024 +#define RT5682_STO1_ADC_MIXER 0x0026 +#define RT5682_AD_DA_MIXER 0x0029 +#define RT5682_STO1_DAC_MIXER 0x002a +#define RT5682_A_DAC1_MUX 0x002b +#define RT5682_DIG_INF2_DATA 0x0030 +/* Mixer - ADC */ +#define RT5682_REC_MIXER 0x003c +#define RT5682_CAL_REC 0x0044 +#define RT5682_ALC_BACK_GAIN 0x0049 +/* Power */ +#define RT5682_PWR_DIG_1 0x0061 +#define RT5682_PWR_DIG_2 0x0062 +#define RT5682_PWR_ANLG_1 0x0063 +#define RT5682_PWR_ANLG_2 0x0064 +#define RT5682_PWR_ANLG_3 0x0065 +#define RT5682_PWR_MIXER 0x0066 +#define RT5682_PWR_VOL 0x0067 +/* Clock Detect */ +#define RT5682_CLK_DET 0x006b +/* Filter Auto Reset */ +#define RT5682_RESET_LPF_CTRL 0x006c +#define RT5682_RESET_HPF_CTRL 0x006d +/* DMIC */ +#define RT5682_DMIC_CTRL_1 0x006e +/* Format - ADC/DAC */ +#define RT5682_I2S1_SDP 0x0070 +#define RT5682_I2S2_SDP 0x0071 +#define RT5682_ADDA_CLK_1 0x0073 +#define RT5682_ADDA_CLK_2 0x0074 +#define RT5682_I2S1_F_DIV_CTRL_1 0x0075 +#define RT5682_I2S1_F_DIV_CTRL_2 0x0076 +/* Format - TDM Control */ +#define RT5682_TDM_CTRL 0x0079 +#define RT5682_TDM_ADDA_CTRL_1 0x007a +#define RT5682_TDM_ADDA_CTRL_2 0x007b +#define RT5682_DATA_SEL_CTRL_1 0x007c +#define RT5682_TDM_TCON_CTRL 0x007e +/* Function - Analog */ +#define RT5682_GLB_CLK 0x0080 +#define RT5682_PLL_CTRL_1 0x0081 +#define RT5682_PLL_CTRL_2 0x0082 +#define RT5682_PLL_TRACK_1 0x0083 +#define RT5682_PLL_TRACK_2 0x0084 +#define RT5682_PLL_TRACK_3 0x0085 +#define RT5682_PLL_TRACK_4 0x0086 +#define RT5682_PLL_TRACK_5 0x0087 +#define RT5682_PLL_TRACK_6 0x0088 +#define RT5682_PLL_TRACK_11 0x008c +#define RT5682_SDW_REF_CLK 0x008d +#define RT5682_DEPOP_1 0x008e +#define RT5682_DEPOP_2 0x008f +#define RT5682_HP_CHARGE_PUMP_1 0x0091 +#define RT5682_HP_CHARGE_PUMP_2 0x0092 +#define RT5682_MICBIAS_1 0x0093 +#define RT5682_MICBIAS_2 0x0094 +#define RT5682_PLL_TRACK_12 0x0098 +#define RT5682_PLL_TRACK_14 0x009a +#define RT5682_PLL2_CTRL_1 0x009b +#define RT5682_PLL2_CTRL_2 0x009c +#define RT5682_PLL2_CTRL_3 0x009d +#define RT5682_PLL2_CTRL_4 0x009e +#define RT5682_RC_CLK_CTRL 0x009f +#define RT5682_I2S_M_CLK_CTRL_1 0x00a0 +#define RT5682_I2S2_F_DIV_CTRL_1 0x00a3 +#define RT5682_I2S2_F_DIV_CTRL_2 0x00a4 +/* Function - Digital */ +#define RT5682_EQ_CTRL_1 0x00ae +#define RT5682_EQ_CTRL_2 0x00af +#define RT5682_IRQ_CTRL_1 0x00b6 +#define RT5682_IRQ_CTRL_2 0x00b7 +#define RT5682_IRQ_CTRL_3 0x00b8 +#define RT5682_IRQ_CTRL_4 0x00b9 +#define RT5682_INT_ST_1 0x00be +#define RT5682_GPIO_CTRL_1 0x00c0 +#define RT5682_GPIO_CTRL_2 0x00c1 +#define RT5682_GPIO_CTRL_3 0x00c2 +#define RT5682_HP_AMP_DET_CTRL_1 0x00d0 +#define RT5682_HP_AMP_DET_CTRL_2 0x00d1 +#define RT5682_MID_HP_AMP_DET 0x00d2 +#define RT5682_LOW_HP_AMP_DET 0x00d3 +#define RT5682_DELAY_BUF_CTRL 0x00d4 +#define RT5682_SV_ZCD_1 0x00d9 +#define RT5682_SV_ZCD_2 0x00da +#define RT5682_IL_CMD_1 0x00db +#define RT5682_IL_CMD_2 0x00dc +#define RT5682_IL_CMD_3 0x00dd +#define RT5682_IL_CMD_4 0x00de +#define RT5682_IL_CMD_5 0x00df +#define RT5682_IL_CMD_6 0x00e0 +#define RT5682_4BTN_IL_CMD_1 0x00e2 +#define RT5682_4BTN_IL_CMD_2 0x00e3 +#define RT5682_4BTN_IL_CMD_3 0x00e4 +#define RT5682_4BTN_IL_CMD_4 0x00e5 +#define RT5682_4BTN_IL_CMD_5 0x00e6 +#define RT5682_4BTN_IL_CMD_6 0x00e7 +#define RT5682_4BTN_IL_CMD_7 0x00e8 + +#define RT5682_ADC_STO1_HP_CTRL_1 0x00ea +#define RT5682_ADC_STO1_HP_CTRL_2 0x00eb +#define RT5682_AJD1_CTRL 0x00f0 +#define RT5682_JD1_THD 0x00f1 +#define RT5682_JD2_THD 0x00f2 +#define RT5682_JD_CTRL_1 0x00f6 +/* General Control */ +#define RT5682_DUMMY_1 0x00fa +#define RT5682_DUMMY_2 0x00fb +#define RT5682_DUMMY_3 0x00fc + +#define RT5682_DAC_ADC_DIG_VOL1 0x0100 +#define RT5682_BIAS_CUR_CTRL_2 0x010b +#define RT5682_BIAS_CUR_CTRL_3 0x010c +#define RT5682_BIAS_CUR_CTRL_4 0x010d +#define RT5682_BIAS_CUR_CTRL_5 0x010e +#define RT5682_BIAS_CUR_CTRL_6 0x010f +#define RT5682_BIAS_CUR_CTRL_7 0x0110 +#define RT5682_BIAS_CUR_CTRL_8 0x0111 +#define RT5682_BIAS_CUR_CTRL_9 0x0112 +#define RT5682_BIAS_CUR_CTRL_10 0x0113 +#define RT5682_VREF_REC_OP_FB_CAP_CTRL 0x0117 +#define RT5682_CHARGE_PUMP_1 0x0125 +#define RT5682_DIG_IN_CTRL_1 0x0132 +#define RT5682_PAD_DRIVING_CTRL 0x0136 +#define RT5682_SOFT_RAMP_DEPOP 0x0138 +#define RT5682_CHOP_DAC 0x013a +#define RT5682_CHOP_ADC 0x013b +#define RT5682_CALIB_ADC_CTRL 0x013c +#define RT5682_VOL_TEST 0x013f +#define RT5682_SPKVDD_DET_STA 0x0142 +#define RT5682_TEST_MODE_CTRL_1 0x0145 +#define RT5682_TEST_MODE_CTRL_2 0x0146 +#define RT5682_TEST_MODE_CTRL_3 0x0147 +#define RT5682_TEST_MODE_CTRL_4 0x0148 +#define RT5682_TEST_MODE_CTRL_5 0x0149 +#define RT5682_PLL1_INTERNAL 0x0150 +#define RT5682_PLL2_INTERNAL 0x0151 +#define RT5682_STO_NG2_CTRL_1 0x0160 +#define RT5682_STO_NG2_CTRL_2 0x0161 +#define RT5682_STO_NG2_CTRL_3 0x0162 +#define RT5682_STO_NG2_CTRL_4 0x0163 +#define RT5682_STO_NG2_CTRL_5 0x0164 +#define RT5682_STO_NG2_CTRL_6 0x0165 +#define RT5682_STO_NG2_CTRL_7 0x0166 +#define RT5682_STO_NG2_CTRL_8 0x0167 +#define RT5682_STO_NG2_CTRL_9 0x0168 +#define RT5682_STO_NG2_CTRL_10 0x0169 +#define RT5682_STO1_DAC_SIL_DET 0x0190 +#define RT5682_SIL_PSV_CTRL1 0x0194 +#define RT5682_SIL_PSV_CTRL2 0x0195 +#define RT5682_SIL_PSV_CTRL3 0x0197 +#define RT5682_SIL_PSV_CTRL4 0x0198 +#define RT5682_SIL_PSV_CTRL5 0x0199 +#define RT5682_HP_IMP_SENS_CTRL_01 0x01af +#define RT5682_HP_IMP_SENS_CTRL_02 0x01b0 +#define RT5682_HP_IMP_SENS_CTRL_03 0x01b1 +#define RT5682_HP_IMP_SENS_CTRL_04 0x01b2 +#define RT5682_HP_IMP_SENS_CTRL_05 0x01b3 +#define RT5682_HP_IMP_SENS_CTRL_06 0x01b4 +#define RT5682_HP_IMP_SENS_CTRL_07 0x01b5 +#define RT5682_HP_IMP_SENS_CTRL_08 0x01b6 +#define RT5682_HP_IMP_SENS_CTRL_09 0x01b7 +#define RT5682_HP_IMP_SENS_CTRL_10 0x01b8 +#define RT5682_HP_IMP_SENS_CTRL_11 0x01b9 +#define RT5682_HP_IMP_SENS_CTRL_12 0x01ba +#define RT5682_HP_IMP_SENS_CTRL_13 0x01bb +#define RT5682_HP_IMP_SENS_CTRL_14 0x01bc +#define RT5682_HP_IMP_SENS_CTRL_15 0x01bd +#define RT5682_HP_IMP_SENS_CTRL_16 0x01be +#define RT5682_HP_IMP_SENS_CTRL_17 0x01bf +#define RT5682_HP_IMP_SENS_CTRL_18 0x01c0 +#define RT5682_HP_IMP_SENS_CTRL_19 0x01c1 +#define RT5682_HP_IMP_SENS_CTRL_20 0x01c2 +#define RT5682_HP_IMP_SENS_CTRL_21 0x01c3 +#define RT5682_HP_IMP_SENS_CTRL_22 0x01c4 +#define RT5682_HP_IMP_SENS_CTRL_23 0x01c5 +#define RT5682_HP_IMP_SENS_CTRL_24 0x01c6 +#define RT5682_HP_IMP_SENS_CTRL_25 0x01c7 +#define RT5682_HP_IMP_SENS_CTRL_26 0x01c8 +#define RT5682_HP_IMP_SENS_CTRL_27 0x01c9 +#define RT5682_HP_IMP_SENS_CTRL_28 0x01ca +#define RT5682_HP_IMP_SENS_CTRL_29 0x01cb +#define RT5682_HP_IMP_SENS_CTRL_30 0x01cc +#define RT5682_HP_IMP_SENS_CTRL_31 0x01cd +#define RT5682_HP_IMP_SENS_CTRL_32 0x01ce +#define RT5682_HP_IMP_SENS_CTRL_33 0x01cf +#define RT5682_HP_IMP_SENS_CTRL_34 0x01d0 +#define RT5682_HP_IMP_SENS_CTRL_35 0x01d1 +#define RT5682_HP_IMP_SENS_CTRL_36 0x01d2 +#define RT5682_HP_IMP_SENS_CTRL_37 0x01d3 +#define RT5682_HP_IMP_SENS_CTRL_38 0x01d4 +#define RT5682_HP_IMP_SENS_CTRL_39 0x01d5 +#define RT5682_HP_IMP_SENS_CTRL_40 0x01d6 +#define RT5682_HP_IMP_SENS_CTRL_41 0x01d7 +#define RT5682_HP_IMP_SENS_CTRL_42 0x01d8 +#define RT5682_HP_IMP_SENS_CTRL_43 0x01d9 +#define RT5682_HP_LOGIC_CTRL_1 0x01da +#define RT5682_HP_LOGIC_CTRL_2 0x01db +#define RT5682_HP_LOGIC_CTRL_3 0x01dc +#define RT5682_HP_CALIB_CTRL_1 0x01de +#define RT5682_HP_CALIB_CTRL_2 0x01df +#define RT5682_HP_CALIB_CTRL_3 0x01e0 +#define RT5682_HP_CALIB_CTRL_4 0x01e1 +#define RT5682_HP_CALIB_CTRL_5 0x01e2 +#define RT5682_HP_CALIB_CTRL_6 0x01e3 +#define RT5682_HP_CALIB_CTRL_7 0x01e4 +#define RT5682_HP_CALIB_CTRL_9 0x01e6 +#define RT5682_HP_CALIB_CTRL_10 0x01e7 +#define RT5682_HP_CALIB_CTRL_11 0x01e8 +#define RT5682_HP_CALIB_STA_1 0x01ea +#define RT5682_HP_CALIB_STA_2 0x01eb +#define RT5682_HP_CALIB_STA_3 0x01ec +#define RT5682_HP_CALIB_STA_4 0x01ed +#define RT5682_HP_CALIB_STA_5 0x01ee +#define RT5682_HP_CALIB_STA_6 0x01ef +#define RT5682_HP_CALIB_STA_7 0x01f0 +#define RT5682_HP_CALIB_STA_8 0x01f1 +#define RT5682_HP_CALIB_STA_9 0x01f2 +#define RT5682_HP_CALIB_STA_10 0x01f3 +#define RT5682_HP_CALIB_STA_11 0x01f4 +#define RT5682_SAR_IL_CMD_1 0x0210 +#define RT5682_SAR_IL_CMD_2 0x0211 +#define RT5682_SAR_IL_CMD_3 0x0212 +#define RT5682_SAR_IL_CMD_4 0x0213 +#define RT5682_SAR_IL_CMD_5 0x0214 +#define RT5682_SAR_IL_CMD_6 0x0215 +#define RT5682_SAR_IL_CMD_7 0x0216 +#define RT5682_SAR_IL_CMD_8 0x0217 +#define RT5682_SAR_IL_CMD_9 0x0218 +#define RT5682_SAR_IL_CMD_10 0x0219 +#define RT5682_SAR_IL_CMD_11 0x021a +#define RT5682_SAR_IL_CMD_12 0x021b +#define RT5682_SAR_IL_CMD_13 0x021c +#define RT5682_EFUSE_CTRL_1 0x0250 +#define RT5682_EFUSE_CTRL_2 0x0251 +#define RT5682_EFUSE_CTRL_3 0x0252 +#define RT5682_EFUSE_CTRL_4 0x0253 +#define RT5682_EFUSE_CTRL_5 0x0254 +#define RT5682_EFUSE_CTRL_6 0x0255 +#define RT5682_EFUSE_CTRL_7 0x0256 +#define RT5682_EFUSE_CTRL_8 0x0257 +#define RT5682_EFUSE_CTRL_9 0x0258 +#define RT5682_EFUSE_CTRL_10 0x0259 +#define RT5682_EFUSE_CTRL_11 0x025a +#define RT5682_JD_TOP_VC_VTRL 0x0270 +#define RT5682_DRC1_CTRL_0 0x02ff +#define RT5682_DRC1_CTRL_1 0x0300 +#define RT5682_DRC1_CTRL_2 0x0301 +#define RT5682_DRC1_CTRL_3 0x0302 +#define RT5682_DRC1_CTRL_4 0x0303 +#define RT5682_DRC1_CTRL_5 0x0304 +#define RT5682_DRC1_CTRL_6 0x0305 +#define RT5682_DRC1_HARD_LMT_CTRL_1 0x0306 +#define RT5682_DRC1_HARD_LMT_CTRL_2 0x0307 +#define RT5682_DRC1_PRIV_1 0x0310 +#define RT5682_DRC1_PRIV_2 0x0311 +#define RT5682_DRC1_PRIV_3 0x0312 +#define RT5682_DRC1_PRIV_4 0x0313 +#define RT5682_DRC1_PRIV_5 0x0314 +#define RT5682_DRC1_PRIV_6 0x0315 +#define RT5682_DRC1_PRIV_7 0x0316 +#define RT5682_DRC1_PRIV_8 0x0317 +#define RT5682_EQ_AUTO_RCV_CTRL1 0x03c0 +#define RT5682_EQ_AUTO_RCV_CTRL2 0x03c1 +#define RT5682_EQ_AUTO_RCV_CTRL3 0x03c2 +#define RT5682_EQ_AUTO_RCV_CTRL4 0x03c3 +#define RT5682_EQ_AUTO_RCV_CTRL5 0x03c4 +#define RT5682_EQ_AUTO_RCV_CTRL6 0x03c5 +#define RT5682_EQ_AUTO_RCV_CTRL7 0x03c6 +#define RT5682_EQ_AUTO_RCV_CTRL8 0x03c7 +#define RT5682_EQ_AUTO_RCV_CTRL9 0x03c8 +#define RT5682_EQ_AUTO_RCV_CTRL10 0x03c9 +#define RT5682_EQ_AUTO_RCV_CTRL11 0x03ca +#define RT5682_EQ_AUTO_RCV_CTRL12 0x03cb +#define RT5682_EQ_AUTO_RCV_CTRL13 0x03cc +#define RT5682_ADC_L_EQ_LPF1_A1 0x03d0 +#define RT5682_R_EQ_LPF1_A1 0x03d1 +#define RT5682_L_EQ_LPF1_H0 0x03d2 +#define RT5682_R_EQ_LPF1_H0 0x03d3 +#define RT5682_L_EQ_BPF1_A1 0x03d4 +#define RT5682_R_EQ_BPF1_A1 0x03d5 +#define RT5682_L_EQ_BPF1_A2 0x03d6 +#define RT5682_R_EQ_BPF1_A2 0x03d7 +#define RT5682_L_EQ_BPF1_H0 0x03d8 +#define RT5682_R_EQ_BPF1_H0 0x03d9 +#define RT5682_L_EQ_BPF2_A1 0x03da +#define RT5682_R_EQ_BPF2_A1 0x03db +#define RT5682_L_EQ_BPF2_A2 0x03dc +#define RT5682_R_EQ_BPF2_A2 0x03dd +#define RT5682_L_EQ_BPF2_H0 0x03de +#define RT5682_R_EQ_BPF2_H0 0x03df +#define RT5682_L_EQ_BPF3_A1 0x03e0 +#define RT5682_R_EQ_BPF3_A1 0x03e1 +#define RT5682_L_EQ_BPF3_A2 0x03e2 +#define RT5682_R_EQ_BPF3_A2 0x03e3 +#define RT5682_L_EQ_BPF3_H0 0x03e4 +#define RT5682_R_EQ_BPF3_H0 0x03e5 +#define RT5682_L_EQ_BPF4_A1 0x03e6 +#define RT5682_R_EQ_BPF4_A1 0x03e7 +#define RT5682_L_EQ_BPF4_A2 0x03e8 +#define RT5682_R_EQ_BPF4_A2 0x03e9 +#define RT5682_L_EQ_BPF4_H0 0x03ea +#define RT5682_R_EQ_BPF4_H0 0x03eb +#define RT5682_L_EQ_HPF1_A1 0x03ec +#define RT5682_R_EQ_HPF1_A1 0x03ed +#define RT5682_L_EQ_HPF1_H0 0x03ee +#define RT5682_R_EQ_HPF1_H0 0x03ef +#define RT5682_L_EQ_PRE_VOL 0x03f0 +#define RT5682_R_EQ_PRE_VOL 0x03f1 +#define RT5682_L_EQ_POST_VOL 0x03f2 +#define RT5682_R_EQ_POST_VOL 0x03f3 +#define RT5682_I2C_MODE 0xffff + + +/* global definition */ +#define RT5682_L_MUTE (0x1 << 15) +#define RT5682_L_MUTE_SFT 15 +#define RT5682_VOL_L_MUTE (0x1 << 14) +#define RT5682_VOL_L_SFT 14 +#define RT5682_R_MUTE (0x1 << 7) +#define RT5682_R_MUTE_SFT 7 +#define RT5682_VOL_R_MUTE (0x1 << 6) +#define RT5682_VOL_R_SFT 6 +#define RT5682_L_VOL_MASK (0x3f << 8) +#define RT5682_L_VOL_SFT 8 +#define RT5682_R_VOL_MASK (0x3f) +#define RT5682_R_VOL_SFT 0 + +/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ +#define RT5682_G_HP (0xf << 8) +#define RT5682_G_HP_SFT 8 +#define RT5682_G_STO_DA_DMIX (0xf) +#define RT5682_G_STO_DA_SFT 0 + +/* CBJ Control (0x000b) */ +#define RT5682_BST_CBJ_MASK (0xf << 8) +#define RT5682_BST_CBJ_SFT 8 + +/* Embeeded Jack and Type Detection Control 1 (0x0010) */ +#define RT5682_EMB_JD_EN (0x1 << 15) +#define RT5682_EMB_JD_EN_SFT 15 +#define RT5682_EMB_JD_RST (0x1 << 14) +#define RT5682_JD_MODE (0x1 << 13) +#define RT5682_JD_MODE_SFT 13 +#define RT5682_DET_TYPE (0x1 << 12) +#define RT5682_DET_TYPE_SFT 12 +#define RT5682_POLA_EXT_JD_MASK (0x1 << 11) +#define RT5682_POLA_EXT_JD_LOW (0x1 << 11) +#define RT5682_POLA_EXT_JD_HIGH (0x0 << 11) +#define RT5682_EXT_JD_DIG (0x1 << 9) +#define RT5682_POL_FAST_OFF_MASK (0x1 << 8) +#define RT5682_POL_FAST_OFF_HIGH (0x1 << 8) +#define RT5682_POL_FAST_OFF_LOW (0x0 << 8) +#define RT5682_FAST_OFF_MASK (0x1 << 7) +#define RT5682_FAST_OFF_EN (0x1 << 7) +#define RT5682_FAST_OFF_DIS (0x0 << 7) +#define RT5682_VREF_POW_MASK (0x1 << 6) +#define RT5682_VREF_POW_FSM (0x0 << 6) +#define RT5682_VREF_POW_REG (0x1 << 6) +#define RT5682_MB1_PATH_MASK (0x1 << 5) +#define RT5682_CTRL_MB1_REG (0x1 << 5) +#define RT5682_CTRL_MB1_FSM (0x0 << 5) +#define RT5682_MB2_PATH_MASK (0x1 << 4) +#define RT5682_CTRL_MB2_REG (0x1 << 4) +#define RT5682_CTRL_MB2_FSM (0x0 << 4) +#define RT5682_TRIG_JD_MASK (0x1 << 3) +#define RT5682_TRIG_JD_HIGH (0x1 << 3) +#define RT5682_TRIG_JD_LOW (0x0 << 3) +#define RT5682_MIC_CAP_MASK (0x1 << 1) +#define RT5682_MIC_CAP_HS (0x1 << 1) +#define RT5682_MIC_CAP_HP (0x0 << 1) +#define RT5682_MIC_CAP_SRC_MASK (0x1) +#define RT5682_MIC_CAP_SRC_REG (0x1) +#define RT5682_MIC_CAP_SRC_ANA (0x0) + +/* Embeeded Jack and Type Detection Control 2 (0x0011) */ +#define RT5682_EXT_JD_SRC (0x7 << 4) +#define RT5682_EXT_JD_SRC_SFT 4 +#define RT5682_EXT_JD_SRC_GPIO_JD1 (0x0 << 4) +#define RT5682_EXT_JD_SRC_GPIO_JD2 (0x1 << 4) +#define RT5682_EXT_JD_SRC_JDH (0x2 << 4) +#define RT5682_EXT_JD_SRC_JDL (0x3 << 4) +#define RT5682_EXT_JD_SRC_MANUAL (0x4 << 4) +#define RT5682_JACK_TYPE_MASK (0x3) + +/* Combo Jack and Type Detection Control 3 (0x0012) */ +#define RT5682_CBJ_IN_BUF_EN (0x1 << 7) + +/* Combo Jack and Type Detection Control 4 (0x0013) */ +#define RT5682_SEL_SHT_MID_TON_MASK (0x3 << 12) +#define RT5682_SEL_SHT_MID_TON_2 (0x0 << 12) +#define RT5682_SEL_SHT_MID_TON_3 (0x1 << 12) +#define RT5682_CBJ_JD_TEST_MASK (0x1 << 6) +#define RT5682_CBJ_JD_TEST_NORM (0x0 << 6) +#define RT5682_CBJ_JD_TEST_MODE (0x1 << 6) + +/* DAC1 Digital Volume (0x0019) */ +#define RT5682_DAC_L1_VOL_MASK (0xff << 8) +#define RT5682_DAC_L1_VOL_SFT 8 +#define RT5682_DAC_R1_VOL_MASK (0xff) +#define RT5682_DAC_R1_VOL_SFT 0 + +/* ADC Digital Volume Control (0x001c) */ +#define RT5682_ADC_L_VOL_MASK (0x7f << 8) +#define RT5682_ADC_L_VOL_SFT 8 +#define RT5682_ADC_R_VOL_MASK (0x7f) +#define RT5682_ADC_R_VOL_SFT 0 + +/* Stereo1 ADC Boost Gain Control (0x001f) */ +#define RT5682_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5682_STO1_ADC_L_BST_SFT 14 +#define RT5682_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5682_STO1_ADC_R_BST_SFT 12 + +/* Sidetone Control (0x0024) */ +#define RT5682_ST_SRC_SEL (0x1 << 8) +#define RT5682_ST_SRC_SFT 8 +#define RT5682_ST_EN_MASK (0x1 << 6) +#define RT5682_ST_DIS (0x0 << 6) +#define RT5682_ST_EN (0x1 << 6) +#define RT5682_ST_EN_SFT 6 + +/* Stereo1 ADC Mixer Control (0x0026) */ +#define RT5682_M_STO1_ADC_L1 (0x1 << 15) +#define RT5682_M_STO1_ADC_L1_SFT 15 +#define RT5682_M_STO1_ADC_L2 (0x1 << 14) +#define RT5682_M_STO1_ADC_L2_SFT 14 +#define RT5682_STO1_ADC1L_SRC_MASK (0x1 << 13) +#define RT5682_STO1_ADC1L_SRC_SFT 13 +#define RT5682_STO1_ADC1_SRC_ADC (0x1 << 13) +#define RT5682_STO1_ADC1_SRC_DACMIX (0x0 << 13) +#define RT5682_STO1_ADC2L_SRC_MASK (0x1 << 12) +#define RT5682_STO1_ADC2L_SRC_SFT 12 +#define RT5682_STO1_ADCL_SRC_MASK (0x3 << 10) +#define RT5682_STO1_ADCL_SRC_SFT 10 +#define RT5682_STO1_DD_L_SRC_MASK (0x1 << 9) +#define RT5682_STO1_DD_L_SRC_SFT 9 +#define RT5682_STO1_DMIC_SRC_MASK (0x1 << 8) +#define RT5682_STO1_DMIC_SRC_SFT 8 +#define RT5682_STO1_DMIC_SRC_DMIC2 (0x1 << 8) +#define RT5682_STO1_DMIC_SRC_DMIC1 (0x0 << 8) +#define RT5682_M_STO1_ADC_R1 (0x1 << 7) +#define RT5682_M_STO1_ADC_R1_SFT 7 +#define RT5682_M_STO1_ADC_R2 (0x1 << 6) +#define RT5682_M_STO1_ADC_R2_SFT 6 +#define RT5682_STO1_ADC1R_SRC_MASK (0x1 << 5) +#define RT5682_STO1_ADC1R_SRC_SFT 5 +#define RT5682_STO1_ADC2R_SRC_MASK (0x1 << 4) +#define RT5682_STO1_ADC2R_SRC_SFT 4 +#define RT5682_STO1_ADCR_SRC_MASK (0x3 << 2) +#define RT5682_STO1_ADCR_SRC_SFT 2 + +/* ADC Mixer to DAC Mixer Control (0x0029) */ +#define RT5682_M_ADCMIX_L (0x1 << 15) +#define RT5682_M_ADCMIX_L_SFT 15 +#define RT5682_M_DAC1_L (0x1 << 14) +#define RT5682_M_DAC1_L_SFT 14 +#define RT5682_DAC1_R_SEL_MASK (0x1 << 10) +#define RT5682_DAC1_R_SEL_SFT 10 +#define RT5682_DAC1_L_SEL_MASK (0x1 << 8) +#define RT5682_DAC1_L_SEL_SFT 8 +#define RT5682_M_ADCMIX_R (0x1 << 7) +#define RT5682_M_ADCMIX_R_SFT 7 +#define RT5682_M_DAC1_R (0x1 << 6) +#define RT5682_M_DAC1_R_SFT 6 + +/* Stereo1 DAC Mixer Control (0x002a) */ +#define RT5682_M_DAC_L1_STO_L (0x1 << 15) +#define RT5682_M_DAC_L1_STO_L_SFT 15 +#define RT5682_G_DAC_L1_STO_L_MASK (0x1 << 14) +#define RT5682_G_DAC_L1_STO_L_SFT 14 +#define RT5682_M_DAC_R1_STO_L (0x1 << 13) +#define RT5682_M_DAC_R1_STO_L_SFT 13 +#define RT5682_G_DAC_R1_STO_L_MASK (0x1 << 12) +#define RT5682_G_DAC_R1_STO_L_SFT 12 +#define RT5682_M_DAC_L1_STO_R (0x1 << 7) +#define RT5682_M_DAC_L1_STO_R_SFT 7 +#define RT5682_G_DAC_L1_STO_R_MASK (0x1 << 6) +#define RT5682_G_DAC_L1_STO_R_SFT 6 +#define RT5682_M_DAC_R1_STO_R (0x1 << 5) +#define RT5682_M_DAC_R1_STO_R_SFT 5 +#define RT5682_G_DAC_R1_STO_R_MASK (0x1 << 4) +#define RT5682_G_DAC_R1_STO_R_SFT 4 + +/* Analog DAC1 Input Source Control (0x002b) */ +#define RT5682_M_ST_STO_L (0x1 << 9) +#define RT5682_M_ST_STO_L_SFT 9 +#define RT5682_M_ST_STO_R (0x1 << 8) +#define RT5682_M_ST_STO_R_SFT 8 +#define RT5682_DAC_L1_SRC_MASK (0x3 << 4) +#define RT5682_A_DACL1_SFT 4 +#define RT5682_DAC_R1_SRC_MASK (0x3) +#define RT5682_A_DACR1_SFT 0 + +/* Digital Interface Data Control (0x0030) */ +#define RT5682_IF2_ADC_SEL_MASK (0x3 << 0) +#define RT5682_IF2_ADC_SEL_SFT 0 + +/* REC Left Mixer Control 2 (0x003c) */ +#define RT5682_G_CBJ_RM1_L (0x7 << 10) +#define RT5682_G_CBJ_RM1_L_SFT 10 +#define RT5682_M_CBJ_RM1_L (0x1 << 7) +#define RT5682_M_CBJ_RM1_L_SFT 7 + +/* Power Management for Digital 1 (0x0061) */ +#define RT5682_PWR_I2S1 (0x1 << 15) +#define RT5682_PWR_I2S1_BIT 15 +#define RT5682_PWR_I2S2 (0x1 << 14) +#define RT5682_PWR_I2S2_BIT 14 +#define RT5682_PWR_DAC_L1 (0x1 << 11) +#define RT5682_PWR_DAC_L1_BIT 11 +#define RT5682_PWR_DAC_R1 (0x1 << 10) +#define RT5682_PWR_DAC_R1_BIT 10 +#define RT5682_PWR_LDO (0x1 << 8) +#define RT5682_PWR_LDO_BIT 8 +#define RT5682_PWR_ADC_L1 (0x1 << 4) +#define RT5682_PWR_ADC_L1_BIT 4 +#define RT5682_PWR_ADC_R1 (0x1 << 3) +#define RT5682_PWR_ADC_R1_BIT 3 +#define RT5682_DIG_GATE_CTRL (0x1 << 0) +#define RT5682_DIG_GATE_CTRL_SFT 0 + + +/* Power Management for Digital 2 (0x0062) */ +#define RT5682_PWR_ADC_S1F (0x1 << 15) +#define RT5682_PWR_ADC_S1F_BIT 15 +#define RT5682_PWR_DAC_S1F (0x1 << 10) +#define RT5682_PWR_DAC_S1F_BIT 10 + +/* Power Management for Analog 1 (0x0063) */ +#define RT5682_PWR_VREF1 (0x1 << 15) +#define RT5682_PWR_VREF1_BIT 15 +#define RT5682_PWR_FV1 (0x1 << 14) +#define RT5682_PWR_FV1_BIT 14 +#define RT5682_PWR_VREF2 (0x1 << 13) +#define RT5682_PWR_VREF2_BIT 13 +#define RT5682_PWR_FV2 (0x1 << 12) +#define RT5682_PWR_FV2_BIT 12 +#define RT5682_LDO1_DBG_MASK (0x3 << 10) +#define RT5682_PWR_MB (0x1 << 9) +#define RT5682_PWR_MB_BIT 9 +#define RT5682_PWR_BG (0x1 << 7) +#define RT5682_PWR_BG_BIT 7 +#define RT5682_LDO1_BYPASS_MASK (0x1 << 6) +#define RT5682_LDO1_BYPASS (0x1 << 6) +#define RT5682_LDO1_NOT_BYPASS (0x0 << 6) +#define RT5682_PWR_MA_BIT 6 +#define RT5682_LDO1_DVO_MASK (0x3 << 4) +#define RT5682_LDO1_DVO_09 (0x0 << 4) +#define RT5682_LDO1_DVO_10 (0x1 << 4) +#define RT5682_LDO1_DVO_12 (0x2 << 4) +#define RT5682_LDO1_DVO_14 (0x3 << 4) +#define RT5682_HP_DRIVER_MASK (0x3 << 2) +#define RT5682_HP_DRIVER_1X (0x0 << 2) +#define RT5682_HP_DRIVER_3X (0x1 << 2) +#define RT5682_HP_DRIVER_5X (0x3 << 2) +#define RT5682_PWR_HA_L (0x1 << 1) +#define RT5682_PWR_HA_L_BIT 1 +#define RT5682_PWR_HA_R (0x1 << 0) +#define RT5682_PWR_HA_R_BIT 0 + +/* Power Management for Analog 2 (0x0064) */ +#define RT5682_PWR_MB1 (0x1 << 11) +#define RT5682_PWR_MB1_PWR_DOWN (0x0 << 11) +#define RT5682_PWR_MB1_BIT 11 +#define RT5682_PWR_MB2 (0x1 << 10) +#define RT5682_PWR_MB2_PWR_DOWN (0x0 << 10) +#define RT5682_PWR_MB2_BIT 10 +#define RT5682_PWR_JDH (0x1 << 3) +#define RT5682_PWR_JDH_BIT 3 +#define RT5682_PWR_JDL (0x1 << 2) +#define RT5682_PWR_JDL_BIT 2 +#define RT5682_PWR_RM1_L (0x1 << 1) +#define RT5682_PWR_RM1_L_BIT 1 + +/* Power Management for Analog 3 (0x0065) */ +#define RT5682_PWR_CBJ (0x1 << 9) +#define RT5682_PWR_CBJ_BIT 9 +#define RT5682_PWR_PLL (0x1 << 6) +#define RT5682_PWR_PLL_BIT 6 +#define RT5682_PWR_PLL2B (0x1 << 5) +#define RT5682_PWR_PLL2B_BIT 5 +#define RT5682_PWR_PLL2F (0x1 << 4) +#define RT5682_PWR_PLL2F_BIT 4 +#define RT5682_PWR_LDO2 (0x1 << 2) +#define RT5682_PWR_LDO2_BIT 2 +#define RT5682_PWR_DET_SPKVDD (0x1 << 1) +#define RT5682_PWR_DET_SPKVDD_BIT 1 + +/* Power Management for Mixer (0x0066) */ +#define RT5682_PWR_STO1_DAC_L (0x1 << 5) +#define RT5682_PWR_STO1_DAC_L_BIT 5 +#define RT5682_PWR_STO1_DAC_R (0x1 << 4) +#define RT5682_PWR_STO1_DAC_R_BIT 4 + +/* MCLK and System Clock Detection Control (0x006b) */ +#define RT5682_SYS_CLK_DET (0x1 << 15) +#define RT5682_SYS_CLK_DET_SFT 15 +#define RT5682_PLL1_CLK_DET (0x1 << 14) +#define RT5682_PLL1_CLK_DET_SFT 14 +#define RT5682_PLL2_CLK_DET (0x1 << 13) +#define RT5682_PLL2_CLK_DET_SFT 13 +#define RT5682_POW_CLK_DET2_SFT 8 +#define RT5682_POW_CLK_DET_SFT 0 + +/* Digital Microphone Control 1 (0x006e) */ +#define RT5682_DMIC_1_EN_MASK (0x1 << 15) +#define RT5682_DMIC_1_EN_SFT 15 +#define RT5682_DMIC_1_DIS (0x0 << 15) +#define RT5682_DMIC_1_EN (0x1 << 15) +#define RT5682_DMIC_1_DP_MASK (0x3 << 4) +#define RT5682_DMIC_1_DP_SFT 4 +#define RT5682_DMIC_1_DP_GPIO2 (0x0 << 4) +#define RT5682_DMIC_1_DP_GPIO5 (0x1 << 4) +#define RT5682_DMIC_CLK_MASK (0xf << 0) +#define RT5682_DMIC_CLK_SFT 0 + +/* I2S1 Audio Serial Data Port Control (0x0070) */ +#define RT5682_SEL_ADCDAT_MASK (0x1 << 15) +#define RT5682_SEL_ADCDAT_OUT (0x0 << 15) +#define RT5682_SEL_ADCDAT_IN (0x1 << 15) +#define RT5682_SEL_ADCDAT_SFT 15 +#define RT5682_I2S1_TX_CHL_MASK (0x7 << 12) +#define RT5682_I2S1_TX_CHL_SFT 12 +#define RT5682_I2S1_TX_CHL_16 (0x0 << 12) +#define RT5682_I2S1_TX_CHL_20 (0x1 << 12) +#define RT5682_I2S1_TX_CHL_24 (0x2 << 12) +#define RT5682_I2S1_TX_CHL_32 (0x3 << 12) +#define RT5682_I2S1_TX_CHL_8 (0x4 << 12) +#define RT5682_I2S1_RX_CHL_MASK (0x7 << 8) +#define RT5682_I2S1_RX_CHL_SFT 8 +#define RT5682_I2S1_RX_CHL_16 (0x0 << 8) +#define RT5682_I2S1_RX_CHL_20 (0x1 << 8) +#define RT5682_I2S1_RX_CHL_24 (0x2 << 8) +#define RT5682_I2S1_RX_CHL_32 (0x3 << 8) +#define RT5682_I2S1_RX_CHL_8 (0x4 << 8) +#define RT5682_I2S1_MONO_MASK (0x1 << 7) +#define RT5682_I2S1_MONO_EN (0x1 << 7) +#define RT5682_I2S1_MONO_DIS (0x0 << 7) +#define RT5682_I2S2_MONO_MASK (0x1 << 6) +#define RT5682_I2S2_MONO_EN (0x1 << 6) +#define RT5682_I2S2_MONO_DIS (0x0 << 6) +#define RT5682_I2S1_DL_MASK (0x7 << 4) +#define RT5682_I2S1_DL_SFT 4 +#define RT5682_I2S1_DL_16 (0x0 << 4) +#define RT5682_I2S1_DL_20 (0x1 << 4) +#define RT5682_I2S1_DL_24 (0x2 << 4) +#define RT5682_I2S1_DL_32 (0x3 << 4) +#define RT5682_I2S1_DL_8 (0x4 << 4) + +/* I2S1/2 Audio Serial Data Port Control (0x0070)(0x0071) */ +#define RT5682_I2S2_MS_MASK (0x1 << 15) +#define RT5682_I2S2_MS_SFT 15 +#define RT5682_I2S2_MS_M (0x0 << 15) +#define RT5682_I2S2_MS_S (0x1 << 15) +#define RT5682_I2S2_PIN_CFG_MASK (0x1 << 14) +#define RT5682_I2S2_PIN_CFG_SFT 14 +#define RT5682_I2S2_CLK_SEL_MASK (0x1 << 11) +#define RT5682_I2S2_CLK_SEL_SFT 11 +#define RT5682_I2S2_OUT_MASK (0x1 << 9) +#define RT5682_I2S2_OUT_SFT 9 +#define RT5682_I2S2_OUT_UM (0x0 << 9) +#define RT5682_I2S2_OUT_M (0x1 << 9) +#define RT5682_I2S_BP_MASK (0x1 << 8) +#define RT5682_I2S_BP_SFT 8 +#define RT5682_I2S_BP_NOR (0x0 << 8) +#define RT5682_I2S_BP_INV (0x1 << 8) +#define RT5682_I2S2_MONO_EN (0x1 << 6) +#define RT5682_I2S2_MONO_DIS (0x0 << 6) +#define RT5682_I2S2_DL_MASK (0x3 << 4) +#define RT5682_I2S2_DL_SFT 4 +#define RT5682_I2S2_DL_16 (0x0 << 4) +#define RT5682_I2S2_DL_20 (0x1 << 4) +#define RT5682_I2S2_DL_24 (0x2 << 4) +#define RT5682_I2S2_DL_8 (0x3 << 4) +#define RT5682_I2S_DF_MASK (0x7) +#define RT5682_I2S_DF_SFT 0 +#define RT5682_I2S_DF_I2S (0x0) +#define RT5682_I2S_DF_LEFT (0x1) +#define RT5682_I2S_DF_PCM_A (0x2) +#define RT5682_I2S_DF_PCM_B (0x3) +#define RT5682_I2S_DF_PCM_A_N (0x6) +#define RT5682_I2S_DF_PCM_B_N (0x7) + +/* ADC/DAC Clock Control 1 (0x0073) */ +#define RT5682_ADC_OSR_MASK (0xf << 12) +#define RT5682_ADC_OSR_SFT 12 +#define RT5682_ADC_OSR_D_1 (0x0 << 12) +#define RT5682_ADC_OSR_D_2 (0x1 << 12) +#define RT5682_ADC_OSR_D_4 (0x2 << 12) +#define RT5682_ADC_OSR_D_6 (0x3 << 12) +#define RT5682_ADC_OSR_D_8 (0x4 << 12) +#define RT5682_ADC_OSR_D_12 (0x5 << 12) +#define RT5682_ADC_OSR_D_16 (0x6 << 12) +#define RT5682_ADC_OSR_D_24 (0x7 << 12) +#define RT5682_ADC_OSR_D_32 (0x8 << 12) +#define RT5682_ADC_OSR_D_48 (0x9 << 12) +#define RT5682_I2S_M_DIV_MASK (0xf << 12) +#define RT5682_I2S_M_DIV_SFT 8 +#define RT5682_I2S_M_D_1 (0x0 << 8) +#define RT5682_I2S_M_D_2 (0x1 << 8) +#define RT5682_I2S_M_D_3 (0x2 << 8) +#define RT5682_I2S_M_D_4 (0x3 << 8) +#define RT5682_I2S_M_D_6 (0x4 << 8) +#define RT5682_I2S_M_D_8 (0x5 << 8) +#define RT5682_I2S_M_D_12 (0x6 << 8) +#define RT5682_I2S_M_D_16 (0x7 << 8) +#define RT5682_I2S_M_D_24 (0x8 << 8) +#define RT5682_I2S_M_D_32 (0x9 << 8) +#define RT5682_I2S_M_D_48 (0x10 << 8) +#define RT5682_I2S_CLK_SRC_MASK (0x7 << 4) +#define RT5682_I2S_CLK_SRC_SFT 4 +#define RT5682_I2S_CLK_SRC_MCLK (0x0 << 4) +#define RT5682_I2S_CLK_SRC_PLL1 (0x1 << 4) +#define RT5682_I2S_CLK_SRC_PLL2 (0x2 << 4) +#define RT5682_I2S_CLK_SRC_SDW (0x3 << 4) +#define RT5682_I2S_CLK_SRC_RCCLK (0x4 << 4) /* 25M */ +#define RT5682_DAC_OSR_MASK (0xf << 0) +#define RT5682_DAC_OSR_SFT 0 +#define RT5682_DAC_OSR_D_1 (0x0 << 0) +#define RT5682_DAC_OSR_D_2 (0x1 << 0) +#define RT5682_DAC_OSR_D_4 (0x2 << 0) +#define RT5682_DAC_OSR_D_6 (0x3 << 0) +#define RT5682_DAC_OSR_D_8 (0x4 << 0) +#define RT5682_DAC_OSR_D_12 (0x5 << 0) +#define RT5682_DAC_OSR_D_16 (0x6 << 0) +#define RT5682_DAC_OSR_D_24 (0x7 << 0) +#define RT5682_DAC_OSR_D_32 (0x8 << 0) +#define RT5682_DAC_OSR_D_48 (0x9 << 0) + +/* ADC/DAC Clock Control 2 (0x0074) */ +#define RT5682_I2S2_BCLK_MS2_MASK (0x1 << 11) +#define RT5682_I2S2_BCLK_MS2_SFT 11 +#define RT5682_I2S2_BCLK_MS2_32 (0x0 << 11) +#define RT5682_I2S2_BCLK_MS2_64 (0x1 << 11) + + +/* TDM control 1 (0x0079) */ +#define RT5682_TDM_TX_CH_MASK (0x3 << 12) +#define RT5682_TDM_TX_CH_2 (0x0 << 12) +#define RT5682_TDM_TX_CH_4 (0x1 << 12) +#define RT5682_TDM_TX_CH_6 (0x2 << 12) +#define RT5682_TDM_TX_CH_8 (0x3 << 12) +#define RT5682_TDM_RX_CH_MASK (0x3 << 8) +#define RT5682_TDM_RX_CH_2 (0x0 << 8) +#define RT5682_TDM_RX_CH_4 (0x1 << 8) +#define RT5682_TDM_RX_CH_6 (0x2 << 8) +#define RT5682_TDM_RX_CH_8 (0x3 << 8) +#define RT5682_TDM_ADC_LCA_MASK (0xf << 4) +#define RT5682_TDM_ADC_LCA_SFT 4 +#define RT5682_TDM_ADC_DL_SFT 0 + +/* TDM control 2 (0x007a) */ +#define RT5682_IF1_ADC1_SEL_SFT 14 +#define RT5682_IF1_ADC2_SEL_SFT 12 +#define RT5682_IF1_ADC3_SEL_SFT 10 +#define RT5682_IF1_ADC4_SEL_SFT 8 +#define RT5682_TDM_ADC_SEL_SFT 4 + +/* TDM control 3 (0x007b) */ +#define RT5682_TDM_EN (0x1 << 7) + +/* TDM/I2S control (0x007e) */ +#define RT5682_TDM_S_BP_MASK (0x1 << 15) +#define RT5682_TDM_S_BP_SFT 15 +#define RT5682_TDM_S_BP_NOR (0x0 << 15) +#define RT5682_TDM_S_BP_INV (0x1 << 15) +#define RT5682_TDM_S_LP_MASK (0x1 << 14) +#define RT5682_TDM_S_LP_SFT 14 +#define RT5682_TDM_S_LP_NOR (0x0 << 14) +#define RT5682_TDM_S_LP_INV (0x1 << 14) +#define RT5682_TDM_DF_MASK (0x7 << 11) +#define RT5682_TDM_DF_SFT 11 +#define RT5682_TDM_DF_I2S (0x0 << 11) +#define RT5682_TDM_DF_LEFT (0x1 << 11) +#define RT5682_TDM_DF_PCM_A (0x2 << 11) +#define RT5682_TDM_DF_PCM_B (0x3 << 11) +#define RT5682_TDM_DF_PCM_A_N (0x6 << 11) +#define RT5682_TDM_DF_PCM_B_N (0x7 << 11) +#define RT5682_TDM_CL_MASK (0x3 << 4) +#define RT5682_TDM_CL_16 (0x0 << 4) +#define RT5682_TDM_CL_20 (0x1 << 4) +#define RT5682_TDM_CL_24 (0x2 << 4) +#define RT5682_TDM_CL_32 (0x3 << 4) +#define RT5682_TDM_M_BP_MASK (0x1 << 2) +#define RT5682_TDM_M_BP_SFT 2 +#define RT5682_TDM_M_BP_NOR (0x0 << 2) +#define RT5682_TDM_M_BP_INV (0x1 << 2) +#define RT5682_TDM_M_LP_MASK (0x1 << 1) +#define RT5682_TDM_M_LP_SFT 1 +#define RT5682_TDM_M_LP_NOR (0x0 << 1) +#define RT5682_TDM_M_LP_INV (0x1 << 1) +#define RT5682_TDM_MS_MASK (0x1 << 0) +#define RT5682_TDM_MS_SFT 0 +#define RT5682_TDM_MS_M (0x0 << 0) +#define RT5682_TDM_MS_S (0x1 << 0) + +/* Global Clock Control (0x0080) */ +#define RT5682_SCLK_SRC_MASK (0x7 << 13) +#define RT5682_SCLK_SRC_SFT 13 +#define RT5682_SCLK_SRC_MCLK (0x0 << 13) +#define RT5682_SCLK_SRC_PLL1 (0x1 << 13) +#define RT5682_SCLK_SRC_PLL2 (0x2 << 13) +#define RT5682_SCLK_SRC_SDW (0x3 << 13) +#define RT5682_SCLK_SRC_RCCLK (0x4 << 13) +#define RT5682_PLL1_SRC_MASK (0x3 << 10) +#define RT5682_PLL1_SRC_SFT 10 +#define RT5682_PLL1_SRC_MCLK (0x0 << 10) +#define RT5682_PLL1_SRC_BCLK1 (0x1 << 10) +#define RT5682_PLL1_SRC_SDW (0x2 << 10) +#define RT5682_PLL1_SRC_RC (0x3 << 10) +#define RT5682_PLL2_SRC_MASK (0x3 << 8) +#define RT5682_PLL2_SRC_SFT 8 +#define RT5682_PLL2_SRC_MCLK (0x0 << 8) +#define RT5682_PLL2_SRC_BCLK1 (0x1 << 8) +#define RT5682_PLL2_SRC_SDW (0x2 << 8) +#define RT5682_PLL2_SRC_RC (0x3 << 8) + + + +#define RT5682_PLL_INP_MAX 40000000 +#define RT5682_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x0081) */ +#define RT5682_PLL_N_MAX 0x001ff +#define RT5682_PLL_N_MASK (RT5682_PLL_N_MAX << 7) +#define RT5682_PLL_N_SFT 7 +#define RT5682_PLL_K_MAX 0x001f +#define RT5682_PLL_K_MASK (RT5682_PLL_K_MAX) +#define RT5682_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x0082) */ +#define RT5682_PLL_M_MAX 0x00f +#define RT5682_PLL_M_MASK (RT5682_PLL_M_MAX << 12) +#define RT5682_PLL_M_SFT 12 +#define RT5682_PLL_M_BP (0x1 << 11) +#define RT5682_PLL_M_BP_SFT 11 +#define RT5682_PLL_K_BP (0x1 << 10) +#define RT5682_PLL_K_BP_SFT 10 +#define RT5682_PLL_RST (0x1 << 1) + +/* PLL tracking mode 1 (0x0083) */ +#define RT5682_DA_ASRC_MASK (0x1 << 13) +#define RT5682_DA_ASRC_SFT 13 +#define RT5682_DAC_STO1_ASRC_MASK (0x1 << 12) +#define RT5682_DAC_STO1_ASRC_SFT 12 +#define RT5682_AD_ASRC_MASK (0x1 << 8) +#define RT5682_AD_ASRC_SFT 8 +#define RT5682_AD_ASRC_SEL_MASK (0x1 << 4) +#define RT5682_AD_ASRC_SEL_SFT 4 +#define RT5682_DMIC_ASRC_MASK (0x1 << 3) +#define RT5682_DMIC_ASRC_SFT 3 +#define RT5682_ADC_STO1_ASRC_MASK (0x1 << 2) +#define RT5682_ADC_STO1_ASRC_SFT 2 +#define RT5682_DA_ASRC_SEL_MASK (0x1 << 0) +#define RT5682_DA_ASRC_SEL_SFT 0 + +/* PLL tracking mode 2 3 (0x0084)(0x0085)*/ +#define RT5682_FILTER_CLK_SEL_MASK (0x7 << 12) +#define RT5682_FILTER_CLK_SEL_SFT 12 +#define RT5682_FILTER_CLK_DIV_MASK (0xf << 8) +#define RT5682_FILTER_CLK_DIV_SFT 8 + +/* ASRC Control 4 (0x0086) */ +#define RT5682_ASRCIN_FTK_N1_MASK (0x3 << 14) +#define RT5682_ASRCIN_FTK_N1_SFT 14 +#define RT5682_ASRCIN_FTK_N2_MASK (0x3 << 12) +#define RT5682_ASRCIN_FTK_N2_SFT 12 +#define RT5682_ASRCIN_FTK_M1_MASK (0x7 << 8) +#define RT5682_ASRCIN_FTK_M1_SFT 8 +#define RT5682_ASRCIN_FTK_M2_MASK (0x7 << 4) +#define RT5682_ASRCIN_FTK_M2_SFT 4 + +/* SoundWire reference clk (0x008d) */ +#define RT5682_PLL2_OUT_MASK (0x1 << 8) +#define RT5682_PLL2_OUT_98M (0x0 << 8) +#define RT5682_PLL2_OUT_49M (0x1 << 8) +#define RT5682_SDW_REF_2_MASK (0xf << 4) +#define RT5682_SDW_REF_2_SFT 4 +#define RT5682_SDW_REF_2_48K (0x0 << 4) +#define RT5682_SDW_REF_2_96K (0x1 << 4) +#define RT5682_SDW_REF_2_192K (0x2 << 4) +#define RT5682_SDW_REF_2_32K (0x3 << 4) +#define RT5682_SDW_REF_2_24K (0x4 << 4) +#define RT5682_SDW_REF_2_16K (0x5 << 4) +#define RT5682_SDW_REF_2_12K (0x6 << 4) +#define RT5682_SDW_REF_2_8K (0x7 << 4) +#define RT5682_SDW_REF_2_44K (0x8 << 4) +#define RT5682_SDW_REF_2_88K (0x9 << 4) +#define RT5682_SDW_REF_2_176K (0xa << 4) +#define RT5682_SDW_REF_2_353K (0xb << 4) +#define RT5682_SDW_REF_2_22K (0xc << 4) +#define RT5682_SDW_REF_2_384K (0xd << 4) +#define RT5682_SDW_REF_2_11K (0xe << 4) +#define RT5682_SDW_REF_1_MASK (0xf << 0) +#define RT5682_SDW_REF_1_SFT 0 +#define RT5682_SDW_REF_1_48K (0x0 << 0) +#define RT5682_SDW_REF_1_96K (0x1 << 0) +#define RT5682_SDW_REF_1_192K (0x2 << 0) +#define RT5682_SDW_REF_1_32K (0x3 << 0) +#define RT5682_SDW_REF_1_24K (0x4 << 0) +#define RT5682_SDW_REF_1_16K (0x5 << 0) +#define RT5682_SDW_REF_1_12K (0x6 << 0) +#define RT5682_SDW_REF_1_8K (0x7 << 0) +#define RT5682_SDW_REF_1_44K (0x8 << 0) +#define RT5682_SDW_REF_1_88K (0x9 << 0) +#define RT5682_SDW_REF_1_176K (0xa << 0) +#define RT5682_SDW_REF_1_353K (0xb << 0) +#define RT5682_SDW_REF_1_22K (0xc << 0) +#define RT5682_SDW_REF_1_384K (0xd << 0) +#define RT5682_SDW_REF_1_11K (0xe << 0) + +/* Depop Mode Control 1 (0x008e) */ +#define RT5682_PUMP_EN (0x1 << 3) +#define RT5682_PUMP_EN_SFT 3 +#define RT5682_CAPLESS_EN (0x1 << 0) +#define RT5682_CAPLESS_EN_SFT 0 + +/* Depop Mode Control 2 (0x8f) */ +#define RT5682_RAMP_MASK (0x1 << 12) +#define RT5682_RAMP_SFT 12 +#define RT5682_RAMP_DIS (0x0 << 12) +#define RT5682_RAMP_EN (0x1 << 12) +#define RT5682_BPS_MASK (0x1 << 11) +#define RT5682_BPS_SFT 11 +#define RT5682_BPS_DIS (0x0 << 11) +#define RT5682_BPS_EN (0x1 << 11) +#define RT5682_FAST_UPDN_MASK (0x1 << 10) +#define RT5682_FAST_UPDN_SFT 10 +#define RT5682_FAST_UPDN_DIS (0x0 << 10) +#define RT5682_FAST_UPDN_EN (0x1 << 10) +#define RT5682_VLO_MASK (0x1 << 7) +#define RT5682_VLO_SFT 7 +#define RT5682_VLO_3V (0x0 << 7) +#define RT5682_VLO_33V (0x1 << 7) + +/* HPOUT charge pump 1 (0x0091) */ +#define RT5682_OSW_L_MASK (0x1 << 11) +#define RT5682_OSW_L_SFT 11 +#define RT5682_OSW_L_DIS (0x0 << 11) +#define RT5682_OSW_L_EN (0x1 << 11) +#define RT5682_OSW_R_MASK (0x1 << 10) +#define RT5682_OSW_R_SFT 10 +#define RT5682_OSW_R_DIS (0x0 << 10) +#define RT5682_OSW_R_EN (0x1 << 10) +#define RT5682_PM_HP_MASK (0x3 << 8) +#define RT5682_PM_HP_SFT 8 +#define RT5682_PM_HP_LV (0x0 << 8) +#define RT5682_PM_HP_MV (0x1 << 8) +#define RT5682_PM_HP_HV (0x2 << 8) +#define RT5682_IB_HP_MASK (0x3 << 6) +#define RT5682_IB_HP_SFT 6 +#define RT5682_IB_HP_125IL (0x0 << 6) +#define RT5682_IB_HP_25IL (0x1 << 6) +#define RT5682_IB_HP_5IL (0x2 << 6) +#define RT5682_IB_HP_1IL (0x3 << 6) + +/* Micbias Control1 (0x93) */ +#define RT5682_MIC1_OV_MASK (0x3 << 14) +#define RT5682_MIC1_OV_SFT 14 +#define RT5682_MIC1_OV_2V7 (0x0 << 14) +#define RT5682_MIC1_OV_2V4 (0x1 << 14) +#define RT5682_MIC1_OV_2V25 (0x3 << 14) +#define RT5682_MIC1_OV_1V8 (0x4 << 14) +#define RT5682_MIC1_CLK_MASK (0x1 << 13) +#define RT5682_MIC1_CLK_SFT 13 +#define RT5682_MIC1_CLK_DIS (0x0 << 13) +#define RT5682_MIC1_CLK_EN (0x1 << 13) +#define RT5682_MIC1_OVCD_MASK (0x1 << 12) +#define RT5682_MIC1_OVCD_SFT 12 +#define RT5682_MIC1_OVCD_DIS (0x0 << 12) +#define RT5682_MIC1_OVCD_EN (0x1 << 12) +#define RT5682_MIC1_OVTH_MASK (0x3 << 10) +#define RT5682_MIC1_OVTH_SFT 10 +#define RT5682_MIC1_OVTH_768UA (0x0 << 10) +#define RT5682_MIC1_OVTH_960UA (0x1 << 10) +#define RT5682_MIC1_OVTH_1152UA (0x2 << 10) +#define RT5682_MIC1_OVTH_1960UA (0x3 << 10) +#define RT5682_MIC2_OV_MASK (0x3 << 8) +#define RT5682_MIC2_OV_SFT 8 +#define RT5682_MIC2_OV_2V7 (0x0 << 8) +#define RT5682_MIC2_OV_2V4 (0x1 << 8) +#define RT5682_MIC2_OV_2V25 (0x3 << 8) +#define RT5682_MIC2_OV_1V8 (0x4 << 8) +#define RT5682_MIC2_CLK_MASK (0x1 << 7) +#define RT5682_MIC2_CLK_SFT 7 +#define RT5682_MIC2_CLK_DIS (0x0 << 7) +#define RT5682_MIC2_CLK_EN (0x1 << 7) +#define RT5682_MIC2_OVTH_MASK (0x3 << 4) +#define RT5682_MIC2_OVTH_SFT 4 +#define RT5682_MIC2_OVTH_768UA (0x0 << 4) +#define RT5682_MIC2_OVTH_960UA (0x1 << 4) +#define RT5682_MIC2_OVTH_1152UA (0x2 << 4) +#define RT5682_MIC2_OVTH_1960UA (0x3 << 4) +#define RT5682_PWR_MB_MASK (0x1 << 3) +#define RT5682_PWR_MB_SFT 3 +#define RT5682_PWR_MB_PD (0x0 << 3) +#define RT5682_PWR_MB_PU (0x1 << 3) + +/* Micbias Control2 (0x0094) */ +#define RT5682_PWR_CLK25M_MASK (0x1 << 9) +#define RT5682_PWR_CLK25M_SFT 9 +#define RT5682_PWR_CLK25M_PD (0x0 << 9) +#define RT5682_PWR_CLK25M_PU (0x1 << 9) +#define RT5682_PWR_CLK1M_MASK (0x1 << 8) +#define RT5682_PWR_CLK1M_SFT 8 +#define RT5682_PWR_CLK1M_PD (0x0 << 8) +#define RT5682_PWR_CLK1M_PU (0x1 << 8) + +/* RC Clock Control (0x009f) */ +#define RT5682_POW_IRQ (0x1 << 15) +#define RT5682_POW_JDH (0x1 << 14) +#define RT5682_POW_JDL (0x1 << 13) +#define RT5682_POW_ANA (0x1 << 12) + +/* I2S Master Mode Clock Control 1 (0x00a0) */ +#define RT5682_CLK_SRC_MCLK (0x0) +#define RT5682_CLK_SRC_PLL1 (0x1) +#define RT5682_CLK_SRC_PLL2 (0x2) +#define RT5682_CLK_SRC_SDW (0x3) +#define RT5682_CLK_SRC_RCCLK (0x4) +#define RT5682_I2S_PD_1 (0x0) +#define RT5682_I2S_PD_2 (0x1) +#define RT5682_I2S_PD_3 (0x2) +#define RT5682_I2S_PD_4 (0x3) +#define RT5682_I2S_PD_6 (0x4) +#define RT5682_I2S_PD_8 (0x5) +#define RT5682_I2S_PD_12 (0x6) +#define RT5682_I2S_PD_16 (0x7) +#define RT5682_I2S_PD_24 (0x8) +#define RT5682_I2S_PD_32 (0x9) +#define RT5682_I2S_PD_48 (0xa) +#define RT5682_I2S2_SRC_MASK (0x3 << 4) +#define RT5682_I2S2_SRC_SFT 4 +#define RT5682_I2S2_M_PD_MASK (0xf << 0) +#define RT5682_I2S2_M_PD_SFT 0 + +/* IRQ Control 1 (0x00b6) */ +#define RT5682_JD1_PULSE_EN_MASK (0x1 << 10) +#define RT5682_JD1_PULSE_EN_SFT 10 +#define RT5682_JD1_PULSE_DIS (0x0 << 10) +#define RT5682_JD1_PULSE_EN (0x1 << 10) + +/* IRQ Control 2 (0x00b7) */ +#define RT5682_JD1_EN_MASK (0x1 << 15) +#define RT5682_JD1_EN_SFT 15 +#define RT5682_JD1_DIS (0x0 << 15) +#define RT5682_JD1_EN (0x1 << 15) +#define RT5682_JD1_POL_MASK (0x1 << 13) +#define RT5682_JD1_POL_NOR (0x0 << 13) +#define RT5682_JD1_POL_INV (0x1 << 13) + +/* IRQ Control 3 (0x00b8) */ +#define RT5682_IL_IRQ_MASK (0x1 << 7) +#define RT5682_IL_IRQ_DIS (0x0 << 7) +#define RT5682_IL_IRQ_EN (0x1 << 7) + +/* GPIO Control 1 (0x00c0) */ +#define RT5682_GP1_PIN_MASK (0x3 << 14) +#define RT5682_GP1_PIN_SFT 14 +#define RT5682_GP1_PIN_GPIO1 (0x0 << 14) +#define RT5682_GP1_PIN_IRQ (0x1 << 14) +#define RT5682_GP1_PIN_DMIC_CLK (0x2 << 14) +#define RT5682_GP2_PIN_MASK (0x3 << 12) +#define RT5682_GP2_PIN_SFT 12 +#define RT5682_GP2_PIN_GPIO2 (0x0 << 12) +#define RT5682_GP2_PIN_LRCK2 (0x1 << 12) +#define RT5682_GP2_PIN_DMIC_SDA (0x2 << 12) +#define RT5682_GP3_PIN_MASK (0x3 << 10) +#define RT5682_GP3_PIN_SFT 10 +#define RT5682_GP3_PIN_GPIO3 (0x0 << 10) +#define RT5682_GP3_PIN_BCLK2 (0x1 << 10) +#define RT5682_GP3_PIN_DMIC_CLK (0x2 << 10) +#define RT5682_GP4_PIN_MASK (0x3 << 8) +#define RT5682_GP4_PIN_SFT 8 +#define RT5682_GP4_PIN_GPIO4 (0x0 << 8) +#define RT5682_GP4_PIN_ADCDAT1 (0x1 << 8) +#define RT5682_GP4_PIN_DMIC_CLK (0x2 << 8) +#define RT5682_GP4_PIN_ADCDAT2 (0x3 << 8) +#define RT5682_GP5_PIN_MASK (0x3 << 6) +#define RT5682_GP5_PIN_SFT 6 +#define RT5682_GP5_PIN_GPIO5 (0x0 << 6) +#define RT5682_GP5_PIN_DACDAT1 (0x1 << 6) +#define RT5682_GP5_PIN_DMIC_SDA (0x2 << 6) +#define RT5682_GP6_PIN_MASK (0x1 << 5) +#define RT5682_GP6_PIN_SFT 5 +#define RT5682_GP6_PIN_GPIO6 (0x0 << 5) +#define RT5682_GP6_PIN_LRCK1 (0x1 << 5) + +/* GPIO Control 2 (0x00c1)*/ +#define RT5682_GP1_PF_MASK (0x1 << 15) +#define RT5682_GP1_PF_IN (0x0 << 15) +#define RT5682_GP1_PF_OUT (0x1 << 15) +#define RT5682_GP1_OUT_MASK (0x1 << 14) +#define RT5682_GP1_OUT_L (0x0 << 14) +#define RT5682_GP1_OUT_H (0x1 << 14) +#define RT5682_GP2_PF_MASK (0x1 << 13) +#define RT5682_GP2_PF_IN (0x0 << 13) +#define RT5682_GP2_PF_OUT (0x1 << 13) +#define RT5682_GP2_OUT_MASK (0x1 << 12) +#define RT5682_GP2_OUT_L (0x0 << 12) +#define RT5682_GP2_OUT_H (0x1 << 12) +#define RT5682_GP3_PF_MASK (0x1 << 11) +#define RT5682_GP3_PF_IN (0x0 << 11) +#define RT5682_GP3_PF_OUT (0x1 << 11) +#define RT5682_GP3_OUT_MASK (0x1 << 10) +#define RT5682_GP3_OUT_L (0x0 << 10) +#define RT5682_GP3_OUT_H (0x1 << 10) +#define RT5682_GP4_PF_MASK (0x1 << 9) +#define RT5682_GP4_PF_IN (0x0 << 9) +#define RT5682_GP4_PF_OUT (0x1 << 9) +#define RT5682_GP4_OUT_MASK (0x1 << 8) +#define RT5682_GP4_OUT_L (0x0 << 8) +#define RT5682_GP4_OUT_H (0x1 << 8) +#define RT5682_GP5_PF_MASK (0x1 << 7) +#define RT5682_GP5_PF_IN (0x0 << 7) +#define RT5682_GP5_PF_OUT (0x1 << 7) +#define RT5682_GP5_OUT_MASK (0x1 << 6) +#define RT5682_GP5_OUT_L (0x0 << 6) +#define RT5682_GP5_OUT_H (0x1 << 6) +#define RT5682_GP6_PF_MASK (0x1 << 5) +#define RT5682_GP6_PF_IN (0x0 << 5) +#define RT5682_GP6_PF_OUT (0x1 << 5) +#define RT5682_GP6_OUT_MASK (0x1 << 4) +#define RT5682_GP6_OUT_L (0x0 << 4) +#define RT5682_GP6_OUT_H (0x1 << 4) + + +/* GPIO Status (0x00c2) */ +#define RT5682_GP6_STA (0x1 << 6) +#define RT5682_GP5_STA (0x1 << 5) +#define RT5682_GP4_STA (0x1 << 4) +#define RT5682_GP3_STA (0x1 << 3) +#define RT5682_GP2_STA (0x1 << 2) +#define RT5682_GP1_STA (0x1 << 1) + +/* Soft volume and zero cross control 1 (0x00d9) */ +#define RT5682_SV_MASK (0x1 << 15) +#define RT5682_SV_SFT 15 +#define RT5682_SV_DIS (0x0 << 15) +#define RT5682_SV_EN (0x1 << 15) +#define RT5682_ZCD_MASK (0x1 << 10) +#define RT5682_ZCD_SFT 10 +#define RT5682_ZCD_PD (0x0 << 10) +#define RT5682_ZCD_PU (0x1 << 10) +#define RT5682_SV_DLY_MASK (0xf) +#define RT5682_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0x00da) */ +#define RT5682_ZCD_BST1_CBJ_MASK (0x1 << 7) +#define RT5682_ZCD_BST1_CBJ_SFT 7 +#define RT5682_ZCD_BST1_CBJ_DIS (0x0 << 7) +#define RT5682_ZCD_BST1_CBJ_EN (0x1 << 7) +#define RT5682_ZCD_RECMIX_MASK (0x1) +#define RT5682_ZCD_RECMIX_SFT 0 +#define RT5682_ZCD_RECMIX_DIS (0x0) +#define RT5682_ZCD_RECMIX_EN (0x1) + +/* 4 Button Inline Command Control 2 (0x00e3) */ +#define RT5682_4BTN_IL_MASK (0x1 << 15) +#define RT5682_4BTN_IL_EN (0x1 << 15) +#define RT5682_4BTN_IL_DIS (0x0 << 15) +#define RT5682_4BTN_IL_RST_MASK (0x1 << 14) +#define RT5682_4BTN_IL_NOR (0x1 << 14) +#define RT5682_4BTN_IL_RST (0x0 << 14) + +/* Analog JD Control (0x00f0) */ +#define RT5682_JDH_RS_MASK (0x1 << 4) +#define RT5682_JDH_NO_PLUG (0x1 << 4) +#define RT5682_JDH_PLUG (0x0 << 4) + +/* Chopper and Clock control for DAC (0x013a)*/ +#define RT5682_CKXEN_DAC1_MASK (0x1 << 13) +#define RT5682_CKXEN_DAC1_SFT 13 +#define RT5682_CKGEN_DAC1_MASK (0x1 << 12) +#define RT5682_CKGEN_DAC1_SFT 12 + +/* Chopper and Clock control for ADC (0x013b)*/ +#define RT5682_CKXEN_ADC1_MASK (0x1 << 13) +#define RT5682_CKXEN_ADC1_SFT 13 +#define RT5682_CKGEN_ADC1_MASK (0x1 << 12) +#define RT5682_CKGEN_ADC1_SFT 12 + +/* Volume test (0x013f)*/ +#define RT5682_SEL_CLK_VOL_MASK (0x1 << 15) +#define RT5682_SEL_CLK_VOL_EN (0x1 << 15) +#define RT5682_SEL_CLK_VOL_DIS (0x0 << 15) + +/* Test Mode Control 1 (0x0145) */ +#define RT5682_AD2DA_LB_MASK (0x1 << 10) +#define RT5682_AD2DA_LB_SFT 10 + +/* Stereo Noise Gate Control 1 (0x0160) */ +#define RT5682_NG2_EN_MASK (0x1 << 15) +#define RT5682_NG2_EN (0x1 << 15) +#define RT5682_NG2_DIS (0x0 << 15) + +/* Stereo1 DAC Silence Detection Control (0x0190) */ +#define RT5682_DEB_STO_DAC_MASK (0x7 << 4) +#define RT5682_DEB_80_MS (0x0 << 4) + +/* SAR ADC Inline Command Control 1 (0x0210) */ +#define RT5682_SAR_BUTT_DET_MASK (0x1 << 15) +#define RT5682_SAR_BUTT_DET_EN (0x1 << 15) +#define RT5682_SAR_BUTT_DET_DIS (0x0 << 15) +#define RT5682_SAR_BUTDET_MODE_MASK (0x1 << 14) +#define RT5682_SAR_BUTDET_POW_SAV (0x1 << 14) +#define RT5682_SAR_BUTDET_POW_NORM (0x0 << 14) +#define RT5682_SAR_BUTDET_RST_MASK (0x1 << 13) +#define RT5682_SAR_BUTDET_RST_NORMAL (0x1 << 13) +#define RT5682_SAR_BUTDET_RST (0x0 << 13) +#define RT5682_SAR_POW_MASK (0x1 << 12) +#define RT5682_SAR_POW_EN (0x1 << 12) +#define RT5682_SAR_POW_DIS (0x0 << 12) +#define RT5682_SAR_RST_MASK (0x1 << 11) +#define RT5682_SAR_RST_NORMAL (0x1 << 11) +#define RT5682_SAR_RST (0x0 << 11) +#define RT5682_SAR_BYPASS_MASK (0x1 << 10) +#define RT5682_SAR_BYPASS_EN (0x1 << 10) +#define RT5682_SAR_BYPASS_DIS (0x0 << 10) +#define RT5682_SAR_SEL_MB1_MASK (0x1 << 9) +#define RT5682_SAR_SEL_MB1_SEL (0x1 << 9) +#define RT5682_SAR_SEL_MB1_NOSEL (0x0 << 9) +#define RT5682_SAR_SEL_MB2_MASK (0x1 << 8) +#define RT5682_SAR_SEL_MB2_SEL (0x1 << 8) +#define RT5682_SAR_SEL_MB2_NOSEL (0x0 << 8) +#define RT5682_SAR_SEL_MODE_MASK (0x1 << 7) +#define RT5682_SAR_SEL_MODE_CMP (0x1 << 7) +#define RT5682_SAR_SEL_MODE_ADC (0x0 << 7) +#define RT5682_SAR_SEL_MB1_MB2_MASK (0x1 << 5) +#define RT5682_SAR_SEL_MB1_MB2_AUTO (0x1 << 5) +#define RT5682_SAR_SEL_MB1_MB2_MANU (0x0 << 5) +#define RT5682_SAR_SEL_SIGNAL_MASK (0x1 << 4) +#define RT5682_SAR_SEL_SIGNAL_AUTO (0x1 << 4) +#define RT5682_SAR_SEL_SIGNAL_MANU (0x0 << 4) + +/* SAR ADC Inline Command Control 13 (0x021c) */ +#define RT5682_SAR_SOUR_MASK (0x3f) +#define RT5682_SAR_SOUR_BTN (0x3f) +#define RT5682_SAR_SOUR_TYPE (0x0) + + +/* System Clock Source */ +enum { + RT5682_SCLK_S_MCLK, + RT5682_SCLK_S_PLL1, + RT5682_SCLK_S_PLL2, + RT5682_SCLK_S_RCCLK, +}; + +/* PLL Source */ +enum { + RT5682_PLL1_S_MCLK, + RT5682_PLL1_S_BCLK1, + RT5682_PLL1_S_RCCLK, +}; + +enum { + RT5682_AIF1, + RT5682_AIF2, + RT5682_AIFS +}; + +/* filter mask */ +enum { + RT5682_DA_STEREO1_FILTER = 0x1, + RT5682_AD_STEREO1_FILTER = (0x1 << 1), +}; + +enum { + RT5682_CLK_SEL_SYS, + RT5682_CLK_SEL_I2S1_ASRC, + RT5682_CLK_SEL_I2S2_ASRC, +}; + +int rt5682_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src); + +#endif /* __RT5682_H__ */ diff --git a/sound/soc/codecs/dio2125.c b/sound/soc/codecs/simple-amplifier.c index 09451cd44f9b..85524acf3e9c 100644 --- a/sound/soc/codecs/dio2125.c +++ b/sound/soc/codecs/simple-amplifier.c @@ -21,9 +21,9 @@ #include <linux/module.h> #include <sound/soc.h> -#define DRV_NAME "dio2125" +#define DRV_NAME "simple-amplifier" -struct dio2125 { +struct simple_amp { struct gpio_desc *gpiod_enable; }; @@ -31,7 +31,7 @@ static int drv_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event) { struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); - struct dio2125 *priv = snd_soc_component_get_drvdata(c); + struct simple_amp *priv = snd_soc_component_get_drvdata(c); int val; switch (event) { @@ -51,7 +51,7 @@ static int drv_event(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget dio2125_dapm_widgets[] = { +static const struct snd_soc_dapm_widget simple_amp_dapm_widgets[] = { SND_SOC_DAPM_INPUT("INL"), SND_SOC_DAPM_INPUT("INR"), SND_SOC_DAPM_OUT_DRV_E("DRV", SND_SOC_NOPM, 0, 0, NULL, 0, drv_event, @@ -60,24 +60,24 @@ static const struct snd_soc_dapm_widget dio2125_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("OUTR"), }; -static const struct snd_soc_dapm_route dio2125_dapm_routes[] = { +static const struct snd_soc_dapm_route simple_amp_dapm_routes[] = { { "DRV", NULL, "INL" }, { "DRV", NULL, "INR" }, { "OUTL", NULL, "DRV" }, { "OUTR", NULL, "DRV" }, }; -static const struct snd_soc_component_driver dio2125_component_driver = { - .dapm_widgets = dio2125_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(dio2125_dapm_widgets), - .dapm_routes = dio2125_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(dio2125_dapm_routes), +static const struct snd_soc_component_driver simple_amp_component_driver = { + .dapm_widgets = simple_amp_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(simple_amp_dapm_widgets), + .dapm_routes = simple_amp_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(simple_amp_dapm_routes), }; -static int dio2125_probe(struct platform_device *pdev) +static int simple_amp_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; - struct dio2125 *priv; + struct simple_amp *priv; int err; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -93,28 +93,30 @@ static int dio2125_probe(struct platform_device *pdev) return err; } - return devm_snd_soc_register_component(dev, &dio2125_component_driver, + return devm_snd_soc_register_component(dev, + &simple_amp_component_driver, NULL, 0); } #ifdef CONFIG_OF -static const struct of_device_id dio2125_ids[] = { +static const struct of_device_id simple_amp_ids[] = { { .compatible = "dioo,dio2125", }, + { .compatible = "simple-audio-amplifier", }, { } }; -MODULE_DEVICE_TABLE(of, dio2125_ids); +MODULE_DEVICE_TABLE(of, simple_amp_ids); #endif -static struct platform_driver dio2125_driver = { +static struct platform_driver simple_amp_driver = { .driver = { .name = DRV_NAME, - .of_match_table = of_match_ptr(dio2125_ids), + .of_match_table = of_match_ptr(simple_amp_ids), }, - .probe = dio2125_probe, + .probe = simple_amp_probe, }; -module_platform_driver(dio2125_driver); +module_platform_driver(simple_amp_driver); -MODULE_DESCRIPTION("ASoC DIO2125 output driver"); +MODULE_DESCRIPTION("ASoC Simple Audio Amplifier driver"); MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 52f34c94ec25..ca2dfe12344e 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -7,6 +7,9 @@ * TAS5721 support: * Copyright (C) 2016 Petr Kulhavy, Barix AG <petr@barix.com> * + * TAS5707 support: + * Copyright (C) 2018 Jerome Brunet, Baylibre SAS <jbrunet@baylibre.com> + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -444,6 +447,111 @@ static const struct tas571x_chip tas5711_chip = { .vol_reg_size = 1, }; +static const struct regmap_range tas5707_volatile_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG), + regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG), + regmap_reg_range(TAS5707_CH1_BQ0_REG, TAS5707_CH2_BQ6_REG), +}; + +static const struct regmap_access_table tas5707_volatile_regs = { + .yes_ranges = tas5707_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(tas5707_volatile_regs_range), + +}; + +static const DECLARE_TLV_DB_SCALE(tas5707_volume_tlv, -7900, 50, 1); + +static const char * const tas5707_volume_slew_step_txt[] = { + "256", "512", "1024", "2048", +}; + +static const unsigned int tas5707_volume_slew_step_values[] = { + 3, 0, 1, 2, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(tas5707_volume_slew_step_enum, + TAS571X_VOL_CFG_REG, 0, 0x3, + tas5707_volume_slew_step_txt, + tas5707_volume_slew_step_values); + +static const struct snd_kcontrol_new tas5707_controls[] = { + SOC_SINGLE_TLV("Master Volume", + TAS571X_MVOL_REG, + 0, 0xff, 1, tas5707_volume_tlv), + SOC_DOUBLE_R_TLV("Speaker Volume", + TAS571X_CH1_VOL_REG, + TAS571X_CH2_VOL_REG, + 0, 0xff, 1, tas5707_volume_tlv), + SOC_DOUBLE("Speaker Switch", + TAS571X_SOFT_MUTE_REG, + TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT, + 1, 1), + + SOC_ENUM("Slew Rate Steps", tas5707_volume_slew_step_enum), + + BIQUAD_COEFS("CH1 - Biquad 0", TAS5707_CH1_BQ0_REG), + BIQUAD_COEFS("CH1 - Biquad 1", TAS5707_CH1_BQ1_REG), + BIQUAD_COEFS("CH1 - Biquad 2", TAS5707_CH1_BQ2_REG), + BIQUAD_COEFS("CH1 - Biquad 3", TAS5707_CH1_BQ3_REG), + BIQUAD_COEFS("CH1 - Biquad 4", TAS5707_CH1_BQ4_REG), + BIQUAD_COEFS("CH1 - Biquad 5", TAS5707_CH1_BQ5_REG), + BIQUAD_COEFS("CH1 - Biquad 6", TAS5707_CH1_BQ6_REG), + + BIQUAD_COEFS("CH2 - Biquad 0", TAS5707_CH2_BQ0_REG), + BIQUAD_COEFS("CH2 - Biquad 1", TAS5707_CH2_BQ1_REG), + BIQUAD_COEFS("CH2 - Biquad 2", TAS5707_CH2_BQ2_REG), + BIQUAD_COEFS("CH2 - Biquad 3", TAS5707_CH2_BQ3_REG), + BIQUAD_COEFS("CH2 - Biquad 4", TAS5707_CH2_BQ4_REG), + BIQUAD_COEFS("CH2 - Biquad 5", TAS5707_CH2_BQ5_REG), + BIQUAD_COEFS("CH2 - Biquad 6", TAS5707_CH2_BQ6_REG), +}; + +static const struct reg_default tas5707_reg_defaults[] = { + {TAS571X_CLK_CTRL_REG, 0x6c}, + {TAS571X_DEV_ID_REG, 0x70}, + {TAS571X_ERR_STATUS_REG, 0x00}, + {TAS571X_SYS_CTRL_1_REG, 0xa0}, + {TAS571X_SDI_REG, 0x05}, + {TAS571X_SYS_CTRL_2_REG, 0x40}, + {TAS571X_SOFT_MUTE_REG, 0x00}, + {TAS571X_MVOL_REG, 0xff}, + {TAS571X_CH1_VOL_REG, 0x30}, + {TAS571X_CH2_VOL_REG, 0x30}, + {TAS571X_VOL_CFG_REG, 0x91}, + {TAS571X_MODULATION_LIMIT_REG, 0x02}, + {TAS571X_IC_DELAY_CH1_REG, 0xac}, + {TAS571X_IC_DELAY_CH2_REG, 0x54}, + {TAS571X_IC_DELAY_CH3_REG, 0xac}, + {TAS571X_IC_DELAY_CH4_REG, 0x54}, + {TAS571X_START_STOP_PERIOD_REG, 0x0f}, + {TAS571X_OSC_TRIM_REG, 0x82}, + {TAS571X_BKND_ERR_REG, 0x02}, + {TAS571X_INPUT_MUX_REG, 0x17772}, + {TAS571X_PWM_MUX_REG, 0x1021345}, +}; + +static const struct regmap_config tas5707_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .max_register = 0xff, + .reg_read = tas571x_reg_read, + .reg_write = tas571x_reg_write, + .reg_defaults = tas5707_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5707_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas5707_volatile_regs, +}; + +static const struct tas571x_chip tas5707_chip = { + .supply_names = tas5711_supply_names, + .num_supply_names = ARRAY_SIZE(tas5711_supply_names), + .controls = tas5707_controls, + .num_controls = ARRAY_SIZE(tas5707_controls), + .regmap_config = &tas5707_regmap_config, + .vol_reg_size = 1, +}; + static const char *const tas5717_supply_names[] = { "AVDD", "DVDD", @@ -775,6 +883,7 @@ static int tas571x_i2c_remove(struct i2c_client *client) } static const struct of_device_id tas571x_of_match[] = { + { .compatible = "ti,tas5707", .data = &tas5707_chip, }, { .compatible = "ti,tas5711", .data = &tas5711_chip, }, { .compatible = "ti,tas5717", .data = &tas5717_chip, }, { .compatible = "ti,tas5719", .data = &tas5717_chip, }, @@ -784,6 +893,7 @@ static const struct of_device_id tas571x_of_match[] = { MODULE_DEVICE_TABLE(of, tas571x_of_match); static const struct i2c_device_id tas571x_i2c_id[] = { + { "tas5707", (kernel_ulong_t) &tas5707_chip }, { "tas5711", (kernel_ulong_t) &tas5711_chip }, { "tas5717", (kernel_ulong_t) &tas5717_chip }, { "tas5719", (kernel_ulong_t) &tas5717_chip }, diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index c45677bc26ad..bd23e89cfe79 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -53,6 +53,22 @@ #define TAS571X_PWM_MUX_REG 0x25 /* 20-byte biquad registers */ +#define TAS5707_CH1_BQ0_REG 0x29 +#define TAS5707_CH1_BQ1_REG 0x2a +#define TAS5707_CH1_BQ2_REG 0x2b +#define TAS5707_CH1_BQ3_REG 0x2c +#define TAS5707_CH1_BQ4_REG 0x2d +#define TAS5707_CH1_BQ5_REG 0x2e +#define TAS5707_CH1_BQ6_REG 0x2f + +#define TAS5707_CH2_BQ0_REG 0x30 +#define TAS5707_CH2_BQ1_REG 0x31 +#define TAS5707_CH2_BQ2_REG 0x32 +#define TAS5707_CH2_BQ3_REG 0x33 +#define TAS5707_CH2_BQ4_REG 0x34 +#define TAS5707_CH2_BQ5_REG 0x35 +#define TAS5707_CH2_BQ6_REG 0x36 + #define TAS5717_CH1_BQ0_REG 0x26 #define TAS5717_CH1_BQ1_REG 0x27 #define TAS5717_CH1_BQ2_REG 0x28 diff --git a/sound/soc/codecs/tda7419.c b/sound/soc/codecs/tda7419.c index 225c210ac38f..7f3b79c5a563 100644 --- a/sound/soc/codecs/tda7419.c +++ b/sound/soc/codecs/tda7419.c @@ -142,9 +142,9 @@ struct tda7419_vol_control { static inline bool tda7419_vol_is_stereo(struct tda7419_vol_control *tvc) { if (tvc->reg == tvc->rreg) - return 0; + return false; - return 1; + return true; } static int tda7419_vol_info(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index d18ff17719cc..7396a6e5277e 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -625,25 +625,34 @@ static int bytes_info_ext(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { /* Volumes */ - SOC_DOUBLE_R_TLV("Headphone Playback Volume", R_HPVOLL, R_HPVOLR, + SOC_DOUBLE_R_TLV("Headphone Volume", R_HPVOLL, R_HPVOLR, FB_HPVOLL, 0x7F, 0, hpvol_scale), - SOC_DOUBLE_R_TLV("Speaker Playback Volume", R_SPKVOLL, R_SPKVOLR, + SOC_DOUBLE_R_TLV("Speaker Volume", R_SPKVOLL, R_SPKVOLR, FB_SPKVOLL, 0x7F, 0, spkvol_scale), - SOC_DOUBLE_R_TLV("Master Playback Volume", R_DACVOLL, R_DACVOLR, + SOC_DOUBLE_R_TLV("Master Volume", R_DACVOLL, R_DACVOLR, FB_DACVOLL, 0xFF, 0, dacvol_scale), - SOC_DOUBLE_R_TLV("PCM Capture Volume", R_ADCVOLL, R_ADCVOLR, + SOC_DOUBLE_R_TLV("PCM Volume", R_ADCVOLL, R_ADCVOLR, FB_ADCVOLL, 0xFF, 0, adcvol_scale), - SOC_DOUBLE_R_TLV("Master Capture Volume", R_INVOLL, R_INVOLR, + SOC_DOUBLE_R_TLV("Input Volume", R_INVOLL, R_INVOLR, FB_INVOLL, 0x3F, 0, invol_scale), /* INSEL */ - SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", R_INSELL, R_INSELR, + SOC_DOUBLE_R_TLV("Mic Boost Volume", R_INSELL, R_INSELR, FB_INSELL_MICBSTL, FV_INSELL_MICBSTL_30DB, 0, mic_boost_scale), /* Input Channel Map */ SOC_ENUM("Input Channel Map", ch_map_select_enum), + /* Mic Bias */ + SOC_SINGLE("Mic Bias Boost Switch", 0x71, 0x07, 1, 0), + + /* Headphone Auto Switching */ + SOC_SINGLE("Headphone Auto Switching Switch", + R_CTL, FB_CTL_HPSWEN, 1, 0), + SOC_SINGLE("Headphone Detect Polarity Toggle Switch", + R_CTL, FB_CTL_HPSWPOL, 1, 0), + /* Coefficient Ram */ COEFF_RAM_CTL("Cascade1L BiQuad1", BIQUAD_SIZE, 0x00), COEFF_RAM_CTL("Cascade1L BiQuad2", BIQUAD_SIZE, 0x05), @@ -733,9 +742,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { R_CLECTL, FB_CLECTL_LIMIT_EN, 1, 0), SOC_SINGLE("Comp Switch", R_CLECTL, FB_CLECTL_COMP_EN, 1, 0), - SOC_SINGLE_TLV("CLE Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("CLE Make-Up Gain Volume", R_MUGAIN, FB_MUGAIN_CLEMUG, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("Comp Thresh Playback Volume", + SOC_SINGLE_TLV("Comp Thresh Volume", R_COMPTH, FB_COMPTH, 0xff, 0, compth_scale), SOC_ENUM("Comp Ratio", compressor_ratio_enum), SND_SOC_BYTES("Comp Atk Time", R_CATKTCL, 2), @@ -766,9 +775,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { SOC_SINGLE("MBC1 Phase Invert Switch", R_DACMBCMUG1, FB_DACMBCMUG1_PHASE, 1, 0), - SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Volume", R_DACMBCMUG1, FB_DACMBCMUG1_MUGAIN, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Playback Volume", + SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Volume", R_DACMBCTHR1, FB_DACMBCTHR1_THRESH, 0xff, 0, compth_scale), SOC_ENUM("DAC MBC1 Comp Ratio", dac_mbc1_compressor_ratio_enum), @@ -778,9 +787,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { SOC_SINGLE("MBC2 Phase Invert Switch", R_DACMBCMUG2, FB_DACMBCMUG2_PHASE, 1, 0), - SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Volume", R_DACMBCMUG2, FB_DACMBCMUG2_MUGAIN, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Playback Volume", + SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Volume", R_DACMBCTHR2, FB_DACMBCTHR2_THRESH, 0xff, 0, compth_scale), SOC_ENUM("DAC MBC2 Comp Ratio", dac_mbc2_compressor_ratio_enum), @@ -790,9 +799,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { SOC_SINGLE("MBC3 Phase Invert Switch", R_DACMBCMUG3, FB_DACMBCMUG3_PHASE, 1, 0), - SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Volume", R_DACMBCMUG3, FB_DACMBCMUG3_MUGAIN, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Playback Volume", + SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Volume", R_DACMBCTHR3, FB_DACMBCTHR3_THRESH, 0xff, 0, compth_scale), SOC_ENUM("DAC MBC3 Comp Ratio", dac_mbc3_compressor_ratio_enum), diff --git a/sound/soc/codecs/tscs42xx.h b/sound/soc/codecs/tscs42xx.h index 814c8f3c4a68..6b3a21081635 100644 --- a/sound/soc/codecs/tscs42xx.h +++ b/sound/soc/codecs/tscs42xx.h @@ -34,6 +34,7 @@ enum { #define R_DACSR 0x19 #define R_PWRM1 0x1A #define R_PWRM2 0x1B +#define R_CTL 0x1C #define R_CONFIG0 0x1F #define R_CONFIG1 0x20 #define R_DMICCTL 0x24 @@ -1110,6 +1111,13 @@ enum { #define RV_PWRM2_VREF_DISABLE \ RV(FV_PWRM2_VREF_DISABLE, FB_PWRM2_VREF) +/****************************** + * R_CTL (0x1C) * + ******************************/ + +/* Fiel Offsets */ +#define FB_CTL_HPSWEN 7 +#define FB_CTL_HPSWPOL 6 /****************************** * R_CONFIG0 (0x1F) * diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index bfd1abd72253..94675da514c8 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -148,7 +148,7 @@ static bool twl6040_can_write_to_chip(struct snd_soc_component *component, case TWL6040_REG_HFRCTL: return priv->dl2_unmuted; default: - return 1; + return true; } } diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 3663b9fd4d65..deff65161504 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1180,6 +1180,9 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L, SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT, WM2200_SPK1R_MUTE_SHIFT, 1, 1), SOC_ENUM("RxANC Src", wm2200_rxanc_input_sel), + +WM_ADSP_FW_CONTROL("DSP1", 0), +WM_ADSP_FW_CONTROL("DSP2", 1), }; WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE); @@ -1553,15 +1556,10 @@ static const struct snd_soc_dapm_route wm2200_dapm_routes[] = { static int wm2200_probe(struct snd_soc_component *component) { struct wm2200_priv *wm2200 = snd_soc_component_get_drvdata(component); - int ret; wm2200->component = component; - ret = snd_soc_add_component_controls(component, wm_adsp_fw_controls, 2); - if (ret != 0) - return ret; - - return ret; + return 0; } static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index e239f4bf2460..9e987cf07450 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -30,7 +30,7 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg) case WM5100_OUTPUT_STATUS_2: case WM5100_INPUT_ENABLES_STATUS: case WM5100_MIC_DETECT_3: - return 1; + return true; default: if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || @@ -41,9 +41,9 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg) (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) - return 1; + return true; else - return 0; + return false; } } @@ -798,7 +798,7 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) case WM5100_DSP3_CONTROL_28: case WM5100_DSP3_CONTROL_29: case WM5100_DSP3_CONTROL_30: - return 1; + return true; default: if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || @@ -809,9 +809,9 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) - return 1; + return true; else - return 0; + return false; } } diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 1ac83388d1b8..7e817e1877c2 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -985,6 +985,8 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP1", 0), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); @@ -2094,6 +2096,12 @@ static int wm5102_probe(struct platform_device *pdev) return ret; } + ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1); + if (ret != 0) + dev_warn(&pdev->dev, + "Failed to set compressed IRQ as a wake source: %d\n", + ret); + arizona_init_common(arizona); ret = arizona_init_vol_limit(arizona); @@ -2117,6 +2125,7 @@ static int wm5102_probe(struct platform_device *pdev) err_spk_irqs: arizona_free_spk_irqs(arizona); err_dsp_irq: + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5102); return ret; @@ -2133,6 +2142,7 @@ static int wm5102_remove(struct platform_device *pdev) arizona_free_spk_irqs(arizona); + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5102); return 0; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index fb9835dcd836..b0789a03d699 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -927,6 +927,11 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP1", 0), +WM_ADSP_FW_CONTROL("DSP2", 1), +WM_ADSP_FW_CONTROL("DSP3", 2), +WM_ADSP_FW_CONTROL("DSP4", 3), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); @@ -2455,6 +2460,12 @@ static int wm5110_probe(struct platform_device *pdev) return ret; } + ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1); + if (ret != 0) + dev_warn(&pdev->dev, + "Failed to set compressed IRQ as a wake source: %d\n", + ret); + arizona_init_common(arizona); ret = arizona_init_vol_limit(arizona); @@ -2478,6 +2489,7 @@ static int wm5110_probe(struct platform_device *pdev) err_spk_irqs: arizona_free_spk_irqs(arizona); err_dsp_irq: + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5110); return ret; @@ -2496,6 +2508,7 @@ static int wm5110_remove(struct platform_device *pdev) arizona_free_spk_irqs(arizona); + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5110); return 0; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 7b8b6ef2f632..6cb3c153ba19 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -251,10 +251,10 @@ static bool wm8903_volatile_register(struct device *dev, unsigned int reg) case WM8903_DC_SERVO_READBACK_2: case WM8903_DC_SERVO_READBACK_3: case WM8903_DC_SERVO_READBACK_4: - return 1; + return true; default: - return 0; + return false; } } diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 9037a35b931d..1965635ec07c 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1455,6 +1455,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= 0x3 | WM8904_AIF_LRCLK_INV; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x3; break; diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index ba44e3d6c1e0..cd204f79647d 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -686,6 +686,7 @@ static int wm8955_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif |= WM8955_LRP; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif |= 0x3; break; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index c30f5aa392c6..8dc1f3d6a988 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -839,6 +839,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, iface |= 0x000c; break; } + /* fall through */ default: dev_err(component->dev, "unsupported width %d\n", params_width(params)); diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index f70f563d59f3..68b4cadc308f 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -653,6 +653,7 @@ static int wm8961_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_B: aif |= WM8961_LRP; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif |= 3; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index a11e9d6bf950..efd8910b1ff7 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2649,6 +2649,7 @@ static int wm8962_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif0 |= WM8962_LRCLK_INV | 3; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif0 |= 3; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 62200117444b..6e52c6a8bab3 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -522,7 +522,7 @@ static inline int get_coeff(int mclk, int rate) /* The set of rates we can generate from the above for each SYSCLK */ static const unsigned int rates_12288[] = { - 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000, + 8000, 12000, 16000, 24000, 32000, 48000, 96000, }; static const struct snd_pcm_hw_constraint_list constraints_12288 = { @@ -540,7 +540,7 @@ static const struct snd_pcm_hw_constraint_list constraints_112896 = { }; static const unsigned int rates_12[] = { - 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000, + 8000, 11025, 12000, 16000, 22050, 24000, 32000, 41100, 48000, 48000, 88235, 96000, }; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 411b9eee88c2..457bc437ce54 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -40,9 +40,9 @@ static bool wm8990_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM8990_RESET: - return 1; + return true; default: - return 0; + return false; } } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7fdfdf3f6e67..14f1b0c0d286 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2432,6 +2432,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, snd_soc_component_update_bits(component, WM8994_POWER_MANAGEMENT_2, WM8994_OPCLK_ENA, 0); } + break; default: return -EINVAL; diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 60e227832331..68c99fe37097 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1465,6 +1465,7 @@ static int wm8995_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif |= WM8995_AIF1_LRCLK_INV; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif |= (0x3 << WM8995_AIF1_FMT_SHIFT); break; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index d9d206046f8c..91711f8958c5 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1498,9 +1498,9 @@ static bool wm8996_readable_register(struct device *dev, unsigned int reg) case WM8996_RIGHT_PDM_SPEAKER: case WM8996_PDM_SPEAKER_MUTE_SEQUENCE: case WM8996_PDM_SPEAKER_VOLUME: - return 1; + return true; default: - return 0; + return false; } } @@ -1522,9 +1522,9 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg) case WM8996_MIC_DETECT_3: case WM8996_HEADPHONE_DETECT_1: case WM8996_HEADPHONE_DETECT_2: - return 1; + return true; default: - return 0; + return false; } } @@ -1858,6 +1858,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, case 24576000: ratediv = WM8996_SYSCLK_DIV; wm8996->sysclk /= 2; + /* fall through */ case 11289600: case 12288000: snd_soc_component_update_bits(component, WM8996_AIF_RATE, diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 5a0ea7b3c149..399255d1f78a 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -933,6 +933,7 @@ static int wm9081_set_dai_fmt(struct snd_soc_dai *dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif2 |= WM9081_AIF_LRCLK_INV; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif2 |= 0x3; break; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 2fcdd84021a5..f61656070225 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -10,6 +10,7 @@ * published by the Free Software Foundation. */ +#include <linux/ctype.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -35,15 +36,15 @@ #include "wm_adsp.h" #define adsp_crit(_dsp, fmt, ...) \ - dev_crit(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__) + dev_crit(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) #define adsp_err(_dsp, fmt, ...) \ - dev_err(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__) + dev_err(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) #define adsp_warn(_dsp, fmt, ...) \ - dev_warn(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__) + dev_warn(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) #define adsp_info(_dsp, fmt, ...) \ - dev_info(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__) + dev_info(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) #define adsp_dbg(_dsp, fmt, ...) \ - dev_dbg(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__) + dev_dbg(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) #define ADSP1_CONTROL_1 0x00 #define ADSP1_CONTROL_2 0x02 @@ -418,7 +419,7 @@ static const struct wm_adsp_fw_caps ctrl_caps[] = { { .id = SND_AUDIOCODEC_BESPOKE, .desc = { - .max_ch = 1, + .max_ch = 8, .sample_rates = { 16000 }, .num_sample_rates = 1, .formats = SNDRV_PCM_FMTBIT_S16_LE, @@ -608,7 +609,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, struct snd_soc_component *component) { struct dentry *root = NULL; - char *root_name; int i; if (!component->debugfs_root) { @@ -616,13 +616,7 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, goto err; } - root_name = kmalloc(PAGE_SIZE, GFP_KERNEL); - if (!root_name) - goto err; - - snprintf(root_name, PAGE_SIZE, "dsp%d", dsp->num); - root = debugfs_create_dir(root_name, component->debugfs_root); - kfree(root_name); + root = debugfs_create_dir(dsp->name, component->debugfs_root); if (!root) goto err; @@ -684,8 +678,8 @@ static inline void wm_adsp_debugfs_clear(struct wm_adsp *dsp) } #endif -static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -695,9 +689,10 @@ static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, return 0; } +EXPORT_SYMBOL_GPL(wm_adsp_fw_get); -static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -721,8 +716,9 @@ static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, return ret; } +EXPORT_SYMBOL_GPL(wm_adsp_fw_put); -static const struct soc_enum wm_adsp_fw_enum[] = { +const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), SOC_ENUM_SINGLE(0, 1, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), SOC_ENUM_SINGLE(0, 2, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), @@ -731,24 +727,7 @@ static const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 5, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), SOC_ENUM_SINGLE(0, 6, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), }; - -const struct snd_kcontrol_new wm_adsp_fw_controls[] = { - SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP5 Firmware", wm_adsp_fw_enum[4], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP6 Firmware", wm_adsp_fw_enum[5], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP7 Firmware", wm_adsp_fw_enum[6], - wm_adsp_fw_get, wm_adsp_fw_put), -}; -EXPORT_SYMBOL_GPL(wm_adsp_fw_controls); +EXPORT_SYMBOL_GPL(wm_adsp_fw_enum); static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, int type) @@ -1330,12 +1309,12 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, switch (dsp->fw_ver) { case 0: case 1: - snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "DSP%d %s %x", - dsp->num, region_name, alg_region->alg); + snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s %x", + dsp->name, region_name, alg_region->alg); break; default: ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "DSP%d%c %.12s %x", dsp->num, *region_name, + "%s%c %.12s %x", dsp->name, *region_name, wm_adsp_fw_text[dsp->fw], alg_region->alg); /* Truncate the subname from the start if it is too long */ @@ -1343,6 +1322,9 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, int avail = SNDRV_CTL_ELEM_ID_NAME_MAXLEN - ret - 2; int skip = 0; + if (dsp->component->name_prefix) + avail -= strlen(dsp->component->name_prefix) + 1; + if (subname_len > avail) skip = subname_len - avail; @@ -1604,6 +1586,15 @@ static int wm_adsp_parse_coeff(struct wm_adsp *dsp, if (ret) return -EINVAL; break; + case WMFW_CTL_TYPE_HOST_BUFFER: + ret = wm_adsp_check_coeff_flags(dsp, &coeff_blk, + WMFW_CTL_FLAG_SYS | + WMFW_CTL_FLAG_VOLATILE | + WMFW_CTL_FLAG_READABLE, + 0); + if (ret) + return -EINVAL; + break; default: adsp_err(dsp, "Unknown control type: %d\n", coeff_blk.ctl_type); @@ -1651,7 +1642,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) if (file == NULL) return -ENOMEM; - snprintf(file, PAGE_SIZE, "%s-dsp%d-%s.wmfw", dsp->part, dsp->num, + snprintf(file, PAGE_SIZE, "%s-%s-%s.wmfw", dsp->part, dsp->fwf_name, wm_adsp_fw[dsp->fw].file); file[PAGE_SIZE - 1] = '\0'; @@ -1871,9 +1862,11 @@ static void wm_adsp_ctl_fixup_base(struct wm_adsp *dsp, } static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, + const struct wm_adsp_region *mem, unsigned int pos, unsigned int len) { void *alg; + unsigned int reg; int ret; __be32 val; @@ -1888,7 +1881,9 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, } /* Read the terminator first to validate the length */ - ret = regmap_raw_read(dsp->regmap, pos + len, &val, sizeof(val)); + reg = wm_adsp_region_to_reg(mem, pos + len); + + ret = regmap_raw_read(dsp->regmap, reg, &val, sizeof(val)); if (ret != 0) { adsp_err(dsp, "Failed to read algorithm list end: %d\n", ret); @@ -1897,13 +1892,18 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, if (be32_to_cpu(val) != 0xbedead) adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbedead\n", - pos + len, be32_to_cpu(val)); + reg, be32_to_cpu(val)); + + /* Convert length from DSP words to bytes */ + len *= sizeof(u32); - alg = kcalloc(len, 2, GFP_KERNEL | GFP_DMA); + alg = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!alg) return ERR_PTR(-ENOMEM); - ret = regmap_raw_read(dsp->regmap, pos, alg, len * 2); + reg = wm_adsp_region_to_reg(mem, pos); + + ret = regmap_raw_read(dsp->regmap, reg, alg, len); if (ret != 0) { adsp_err(dsp, "Failed to read algorithm list: %d\n", ret); kfree(alg); @@ -2002,10 +2002,11 @@ static int wm_adsp1_setup_algs(struct wm_adsp *dsp) if (IS_ERR(alg_region)) return PTR_ERR(alg_region); - pos = sizeof(adsp1_id) / 2; - len = (sizeof(*adsp1_alg) * n_algs) / 2; + /* Calculate offset and length in DSP words */ + pos = sizeof(adsp1_id) / sizeof(u32); + len = (sizeof(*adsp1_alg) * n_algs) / sizeof(u32); - adsp1_alg = wm_adsp_read_algs(dsp, n_algs, mem->base + pos, len); + adsp1_alg = wm_adsp_read_algs(dsp, n_algs, mem, pos, len); if (IS_ERR(adsp1_alg)) return PTR_ERR(adsp1_alg); @@ -2113,10 +2114,11 @@ static int wm_adsp2_setup_algs(struct wm_adsp *dsp) if (IS_ERR(alg_region)) return PTR_ERR(alg_region); - pos = sizeof(adsp2_id) / 2; - len = (sizeof(*adsp2_alg) * n_algs) / 2; + /* Calculate offset and length in DSP words */ + pos = sizeof(adsp2_id) / sizeof(u32); + len = (sizeof(*adsp2_alg) * n_algs) / sizeof(u32); - adsp2_alg = wm_adsp_read_algs(dsp, n_algs, mem->base + pos, len); + adsp2_alg = wm_adsp_read_algs(dsp, n_algs, mem, pos, len); if (IS_ERR(adsp2_alg)) return PTR_ERR(adsp2_alg); @@ -2218,7 +2220,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) if (file == NULL) return -ENOMEM; - snprintf(file, PAGE_SIZE, "%s-dsp%d-%s.bin", dsp->part, dsp->num, + snprintf(file, PAGE_SIZE, "%s-%s-%s.bin", dsp->part, dsp->fwf_name, wm_adsp_fw[dsp->fw].file); file[PAGE_SIZE - 1] = '\0'; @@ -2390,8 +2392,38 @@ out: return ret; } +static int wm_adsp_create_name(struct wm_adsp *dsp) +{ + char *p; + + if (!dsp->name) { + dsp->name = devm_kasprintf(dsp->dev, GFP_KERNEL, "DSP%d", + dsp->num); + if (!dsp->name) + return -ENOMEM; + } + + if (!dsp->fwf_name) { + p = devm_kstrdup(dsp->dev, dsp->name, GFP_KERNEL); + if (!p) + return -ENOMEM; + + dsp->fwf_name = p; + for (; *p != 0; ++p) + *p = tolower(*p); + } + + return 0; +} + int wm_adsp1_init(struct wm_adsp *dsp) { + int ret; + + ret = wm_adsp_create_name(dsp); + if (ret) + return ret; + INIT_LIST_HEAD(&dsp->alg_regions); mutex_init(&dsp->pwr_lock); @@ -2642,7 +2674,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); - struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); + struct wm_adsp *dsps = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct wm_adsp *dsp = &dsps[mc->shift - 1]; ucontrol->value.integer.value[0] = dsp->preloaded; @@ -2654,13 +2689,14 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); - struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); + struct wm_adsp *dsps = snd_soc_component_get_drvdata(component); struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + struct wm_adsp *dsp = &dsps[mc->shift - 1]; char preload[32]; - snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift); + snprintf(preload, ARRAY_SIZE(preload), "%s Preload", dsp->name); dsp->preloaded = ucontrol->value.integer.value[0]; @@ -2671,6 +2707,8 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, snd_soc_dapm_sync(dapm); + flush_work(&dsp->boot_work); + return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put); @@ -2853,17 +2891,14 @@ int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *comp { char preload[32]; - snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", dsp->num); - + snprintf(preload, ARRAY_SIZE(preload), "%s Preload", dsp->name); snd_soc_component_disable_pin(component, preload); wm_adsp2_init_debugfs(dsp, component); dsp->component = component; - return snd_soc_add_component_controls(component, - &wm_adsp_fw_controls[dsp->num - 1], - 1); + return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_component_probe); @@ -2879,6 +2914,10 @@ int wm_adsp2_init(struct wm_adsp *dsp) { int ret; + ret = wm_adsp_create_name(dsp); + if (ret) + return ret; + switch (dsp->rev) { case 0: /* @@ -3186,7 +3225,7 @@ static inline int wm_adsp_buffer_write(struct wm_adsp_compr_buf *buf, buf->host_buf_ptr + field_offset, data); } -static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf) +static int wm_adsp_legacy_host_buf_addr(struct wm_adsp_compr_buf *buf) { struct wm_adsp_alg_region *alg_region; struct wm_adsp *dsp = buf->dsp; @@ -3225,6 +3264,61 @@ static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf) return 0; } +static struct wm_coeff_ctl * +wm_adsp_find_host_buffer_ctrl(struct wm_adsp_compr_buf *buf) +{ + struct wm_adsp *dsp = buf->dsp; + struct wm_coeff_ctl *ctl; + + list_for_each_entry(ctl, &dsp->ctl_list, list) { + if (ctl->type != WMFW_CTL_TYPE_HOST_BUFFER) + continue; + + if (!ctl->enabled) + continue; + + return ctl; + } + + return NULL; +} + +static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf) +{ + struct wm_adsp *dsp = buf->dsp; + struct wm_coeff_ctl *ctl; + unsigned int reg; + u32 val; + int i, ret; + + ctl = wm_adsp_find_host_buffer_ctrl(buf); + if (!ctl) + return wm_adsp_legacy_host_buf_addr(buf); + + ret = wm_coeff_base_reg(ctl, ®); + if (ret) + return ret; + + for (i = 0; i < 5; ++i) { + ret = regmap_raw_read(dsp->regmap, reg, &val, sizeof(val)); + if (ret < 0) + return ret; + + if (val) + break; + + usleep_range(1000, 2000); + } + + if (!val) + return -EIO; + + buf->host_buf_ptr = be32_to_cpu(val); + adsp_dbg(dsp, "host_buf_ptr=%x\n", buf->host_buf_ptr); + + return 0; +} + static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) { const struct wm_adsp_fw_caps *caps = wm_adsp_fw[buf->dsp->fw].caps; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index bc6d359f0533..4b8778b0b06c 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -57,6 +57,8 @@ struct wm_adsp_compr_buf; struct wm_adsp { const char *part; + const char *name; + const char *fwf_name; int rev; int num; int type; @@ -121,7 +123,11 @@ struct wm_adsp { .reg = SND_SOC_NOPM, .shift = num, .event = wm_adsp2_event, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD } -extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; +#define WM_ADSP_FW_CONTROL(dspname, num) \ + SOC_ENUM_EXT(dspname " Firmware", wm_adsp_fw_enum[num], \ + wm_adsp_fw_get, wm_adsp_fw_put) + +extern const struct soc_enum wm_adsp_fw_enum[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); @@ -144,6 +150,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream); int wm_adsp_compr_free(struct snd_compr_stream *stream); diff --git a/sound/soc/codecs/wmfw.h b/sound/soc/codecs/wmfw.h index ec78b9da020f..0c3f50acb8b1 100644 --- a/sound/soc/codecs/wmfw.h +++ b/sound/soc/codecs/wmfw.h @@ -29,6 +29,7 @@ /* Non-ALSA coefficient types start at 0x1000 */ #define WMFW_CTL_TYPE_ACKED 0x1000 /* acked control */ #define WMFW_CTL_TYPE_HOSTEVENT 0x1001 /* event control */ +#define WMFW_CTL_TYPE_HOST_BUFFER 0x1002 /* host buffer pointer */ struct wmfw_header { char magic[4]; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 807040bb3921..a3206e65e5e5 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -340,6 +340,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, * rate is lowered. */ inv_fs = true; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: dev->mode = MOD_DSP_A; break; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 47c0c821d325..f70db8412c7c 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -320,12 +320,8 @@ static irqreturn_t davinci_mcasp_tx_irq_handler(int irq, void *data) handled_mask |= XUNDRN; substream = mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK]; - if (substream) { - snd_pcm_stream_lock_irq(substream); - if (snd_pcm_running(substream)) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irq(substream); - } + if (substream) + snd_pcm_stop_xrun(substream); } if (!handled_mask) @@ -355,12 +351,8 @@ static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data) handled_mask |= ROVRN; substream = mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE]; - if (substream) { - snd_pcm_stream_lock_irq(substream); - if (snd_pcm_running(substream)) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irq(substream); - } + if (substream) + snd_pcm_stop_xrun(substream); } if (!handled_mask) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4a6750aa3637..44433b20435c 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -1,14 +1,10 @@ -/* - * Freescale Generic ASoC Sound Card driver with ASRC - * - * Copyright (C) 2014 Freescale Semiconductor, Inc. - * - * Author: Nicolin Chen <nicoleotsuka@gmail.com> - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale Generic ASoC Sound Card driver with ASRC +// +// Copyright (C) 2014 Freescale Semiconductor, Inc. +// +// Author: Nicolin Chen <nicoleotsuka@gmail.com> #include <linux/clk.h> #include <linux/i2c.h> @@ -199,7 +195,7 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); snd_mask_none(mask); - snd_mask_set(mask, (__force int)priv->asrc_format); + snd_mask_set_format(mask, priv->asrc_format); return 0; } diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index adfb8135d739..528e8b108422 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -1,14 +1,10 @@ -/* - * Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver - * - * Copyright (C) 2014 Freescale Semiconductor, Inc. - * - * Author: Nicolin Chen <nicoleotsuka@gmail.com> - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver +// +// Copyright (C) 2014 Freescale Semiconductor, Inc. +// +// Author: Nicolin Chen <nicoleotsuka@gmail.com> #include <linux/clk.h> #include <linux/delay.h> diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index d558dd5499a5..c60075112570 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -1,13 +1,10 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* * fsl_asrc.h - Freescale ASRC ALSA SoC header file * * Copyright (C) 2014 Freescale Semiconductor, Inc. * * Author: Nicolin Chen <nicoleotsuka@gmail.com> - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. */ #ifndef _FSL_ASRC_H diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 565e16d8fe85..1033ac6631b0 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -1,14 +1,10 @@ -/* - * Freescale ASRC ALSA SoC Platform (DMA) driver - * - * Copyright (C) 2014 Freescale Semiconductor, Inc. - * - * Author: Nicolin Chen <nicoleotsuka@gmail.com> - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale ASRC ALSA SoC Platform (DMA) driver +// +// Copyright (C) 2014 Freescale Semiconductor, Inc. +// +// Author: Nicolin Chen <nicoleotsuka@gmail.com> #include <linux/dma-mapping.h> #include <linux/module.h> diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 8f43110373b8..c1d1d06783e5 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -249,6 +249,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, break; case ESAI_HCKT_EXTAL: ecr |= ESAI_ECR_ETI; + /* fall through */ case ESAI_HCKR_EXTAL: ecr |= ESAI_ECR_ERI; break; diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 9b59d87b61bf..740b90df44bb 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1118,7 +1118,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) { for (txclk_df = 1; txclk_df <= 128; txclk_df++) { - rate_ideal = rate[index] * txclk_df * 64; + rate_ideal = rate[index] * txclk_df * 64ULL; if (round) rate_actual = clk_round_rate(clk, rate_ideal); else diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 7592b0406370..7f0fa4b52223 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -1,14 +1,10 @@ -/** - * Freescale ALSA SoC Machine driver utility - * - * Author: Timur Tabi <timur@freescale.com> - * - * Copyright 2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale ALSA SoC Machine driver utility +// +// Author: Timur Tabi <timur@freescale.com> +// +// Copyright 2010 Freescale Semiconductor, Inc. #include <linux/module.h> #include <linux/of_address.h> diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h index 1687b66ef18e..c5dc2a14b492 100644 --- a/sound/soc/fsl/fsl_utils.h +++ b/sound/soc/fsl/fsl_utils.h @@ -1,13 +1,10 @@ -/** +/* SPDX-License-Identifier: GPL-2.0 */ +/* * Freescale ALSA SoC Machine driver utility * * Author: Timur Tabi <timur@freescale.com> * * Copyright 2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. */ #ifndef _FSL_UTILS_H diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index b99e0b5e00e9..c29200cf755a 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -1,14 +1,7 @@ -/* - * Copyright 2012 Freescale Semiconductor, Inc. - * Copyright 2012 Linaro Ltd. - * - * The code contained herein is licensed under the GNU General Public - * License. You may obtain a copy of the GNU General Public License - * Version 2 or later at the following locations: - * - * http://www.opensource.org/licenses/gpl-license.html - * http://www.gnu.org/copyleft/gpl.html - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright 2012 Freescale Semiconductor, Inc. +// Copyright 2012 Linaro Ltd. #include <linux/module.h> #include <linux/of.h> diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index d93bacacbd5b..2094d2c8919f 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -1,15 +1,12 @@ -/* - * ASoC audio graph sound card support - * - * Copyright (C) 2016 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * based on ${LINUX}/sound/soc/generic/simple-card.c - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC audio graph sound card support +// +// Copyright (C) 2016 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> +// +// based on ${LINUX}/sound/soc/generic/simple-card.c + #include <linux/clk.h> #include <linux/device.h> #include <linux/gpio.h> @@ -21,7 +18,6 @@ #include <linux/of_graph.h> #include <linux/platform_device.h> #include <linux/string.h> -#include <sound/jack.h> #include <sound/simple_card_utils.h> struct graph_card_data { @@ -32,6 +28,8 @@ struct graph_card_data { unsigned int mclk_fs; } *dai_props; unsigned int mclk_fs; + struct asoc_simple_jack hp_jack; + struct asoc_simple_jack mic_jack; struct snd_soc_dai_link *dai_link; struct gpio_desc *pa_gpio; }; @@ -278,6 +276,22 @@ static int asoc_graph_get_dais_count(struct device *dev) return count; } +static int asoc_graph_soc_card_probe(struct snd_soc_card *card) +{ + struct graph_card_data *priv = snd_soc_card_get_drvdata(card); + int ret; + + ret = asoc_simple_card_init_hp(card, &priv->hp_jack, NULL); + if (ret < 0) + return ret; + + ret = asoc_simple_card_init_mic(card, &priv->mic_jack, NULL); + if (ret < 0) + return ret; + + return 0; +} + static int asoc_graph_card_probe(struct platform_device *pdev) { struct graph_card_data *priv; @@ -319,6 +333,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) card->num_links = num; card->dapm_widgets = asoc_graph_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets); + card->probe = asoc_graph_soc_card_probe; ret = asoc_graph_card_parse_of(priv); if (ret < 0) { diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 095ef6426d42..92882e392d6c 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -1,17 +1,14 @@ -/* - * ASoC audio graph SCU sound card support - * - * Copyright (C) 2017 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * based on - * ${LINUX}/sound/soc/generic/simple-scu-card.c - * ${LINUX}/sound/soc/generic/audio-graph-card.c - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC audio graph SCU sound card support +// +// Copyright (C) 2017 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> +// +// based on +// ${LINUX}/sound/soc/generic/simple-scu-card.c +// ${LINUX}/sound/soc/generic/audio-graph-card.c + #include <linux/clk.h> #include <linux/device.h> #include <linux/gpio.h> diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 3751a07de6aa..d3f3f0fec74c 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -1,16 +1,17 @@ -/* - * simple-card-utils.c - * - * Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// simple-card-utils.c +// +// Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #include <linux/clk.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> +#include <linux/of_gpio.h> #include <linux/of_graph.h> +#include <sound/jack.h> #include <sound/simple_card_utils.h> void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, @@ -419,6 +420,61 @@ int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_widgets); +int asoc_simple_card_init_jack(struct snd_soc_card *card, + struct asoc_simple_jack *sjack, + int is_hp, char *prefix) +{ + struct device *dev = card->dev; + enum of_gpio_flags flags; + char prop[128]; + char *pin_name; + char *gpio_name; + int mask; + int det; + + if (!prefix) + prefix = ""; + + sjack->gpio.gpio = -ENOENT; + + if (is_hp) { + snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix); + pin_name = "Headphones"; + gpio_name = "Headphone detection"; + mask = SND_JACK_HEADPHONE; + } else { + snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix); + pin_name = "Mic Jack"; + gpio_name = "Mic detection"; + mask = SND_JACK_MICROPHONE; + } + + det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags); + if (det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + if (gpio_is_valid(det)) { + sjack->pin.pin = pin_name; + sjack->pin.mask = mask; + + sjack->gpio.name = gpio_name; + sjack->gpio.report = mask; + sjack->gpio.gpio = det; + sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); + sjack->gpio.debounce_time = 150; + + snd_soc_card_jack_new(card, pin_name, mask, + &sjack->jack, + &sjack->pin, 1); + + snd_soc_jack_add_gpios(&sjack->jack, 1, + &sjack->gpio); + } + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_card_init_jack); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>"); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 8b374af86a6e..64bf3560c1d1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -1,32 +1,20 @@ -/* - * ASoC simple sound card support - * - * Copyright (C) 2012 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC simple sound card support +// +// Copyright (C) 2012 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #include <linux/clk.h> #include <linux/device.h> -#include <linux/gpio.h> #include <linux/module.h> #include <linux/of.h> -#include <linux/of_gpio.h> #include <linux/platform_device.h> #include <linux/string.h> -#include <sound/jack.h> #include <sound/simple_card.h> #include <sound/soc-dai.h> #include <sound/soc.h> -struct asoc_simple_jack { - struct snd_soc_jack jack; - struct snd_soc_jack_pin pin; - struct snd_soc_jack_gpio gpio; -}; - struct simple_card_data { struct snd_soc_card snd_card; struct simple_dai_props { @@ -49,61 +37,6 @@ struct simple_card_data { #define CELL "#sound-dai-cells" #define PREFIX "simple-audio-card," -#define asoc_simple_card_init_hp(card, sjack, prefix)\ - asoc_simple_card_init_jack(card, sjack, 1, prefix) -#define asoc_simple_card_init_mic(card, sjack, prefix)\ - asoc_simple_card_init_jack(card, sjack, 0, prefix) -static int asoc_simple_card_init_jack(struct snd_soc_card *card, - struct asoc_simple_jack *sjack, - int is_hp, char *prefix) -{ - struct device *dev = card->dev; - enum of_gpio_flags flags; - char prop[128]; - char *pin_name; - char *gpio_name; - int mask; - int det; - - sjack->gpio.gpio = -ENOENT; - - if (is_hp) { - snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix); - pin_name = "Headphones"; - gpio_name = "Headphone detection"; - mask = SND_JACK_HEADPHONE; - } else { - snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix); - pin_name = "Mic Jack"; - gpio_name = "Mic detection"; - mask = SND_JACK_MICROPHONE; - } - - det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags); - if (det == -EPROBE_DEFER) - return -EPROBE_DEFER; - - if (gpio_is_valid(det)) { - sjack->pin.pin = pin_name; - sjack->pin.mask = mask; - - sjack->gpio.name = gpio_name; - sjack->gpio.report = mask; - sjack->gpio.gpio = det; - sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); - sjack->gpio.debounce_time = 150; - - snd_soc_card_jack_new(card, pin_name, mask, - &sjack->jack, - &sjack->pin, 1); - - snd_soc_jack_add_gpios(&sjack->jack, 1, - &sjack->gpio); - } - - return 0; -} - static int asoc_simple_card_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -213,14 +146,6 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; - ret = asoc_simple_card_init_hp(rtd->card, &priv->hp_jack, PREFIX); - if (ret < 0) - return ret; - - ret = asoc_simple_card_init_mic(rtd->card, &priv->mic_jack, PREFIX); - if (ret < 0) - return ret; - return 0; } @@ -414,6 +339,22 @@ card_parse_end: return ret; } +static int asoc_simple_soc_card_probe(struct snd_soc_card *card) +{ + struct simple_card_data *priv = snd_soc_card_get_drvdata(card); + int ret; + + ret = asoc_simple_card_init_hp(card, &priv->hp_jack, PREFIX); + if (ret < 0) + return ret; + + ret = asoc_simple_card_init_mic(card, &priv->mic_jack, PREFIX); + if (ret < 0) + return ret; + + return 0; +} + static int asoc_simple_card_probe(struct platform_device *pdev) { struct simple_card_data *priv; @@ -449,6 +390,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) card->dev = dev; card->dai_link = priv->dai_link; card->num_links = num; + card->probe = asoc_simple_soc_card_probe; if (np && of_device_is_available(np)) { diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 487716559deb..16a83bc51e0e 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -1,15 +1,12 @@ -/* - * ASoC simple SCU sound card support - * - * Copyright (C) 2015 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * based on ${LINUX}/sound/soc/generic/simple-card.c - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC simple SCU sound card support +// +// Copyright (C) 2015 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> +// +// based on ${LINUX}/sound/soc/generic/simple-card.c + #include <linux/clk.h> #include <linux/device.h> #include <linux/module.h> diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 6a8b253c58d2..5455d6e0ab53 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -266,17 +266,15 @@ static int sst_cdev_ack(struct device *dev, unsigned int str_id, stream->cumm_bytes += bytes; dev_dbg(dev, "bytes copied %d inc by %ld\n", stream->cumm_bytes, bytes); - memcpy_fromio(&fw_tstamp, - ((void *)(ctx->mailbox + ctx->tstamp) - +(str_id * sizeof(fw_tstamp))), - sizeof(fw_tstamp)); + addr = ((void __iomem *)(ctx->mailbox + ctx->tstamp)) + + (str_id * sizeof(fw_tstamp)); + + memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp)); fw_tstamp.bytes_copied = stream->cumm_bytes; dev_dbg(dev, "bytes sent to fw %llu inc by %ld\n", fw_tstamp.bytes_copied, bytes); - addr = ((void *)(ctx->mailbox + ctx->tstamp)) + - (str_id * sizeof(fw_tstamp)); offset = offsetof(struct snd_sst_tstamp, bytes_copied); sst_shim_write(addr, offset, fw_tstamp.bytes_copied); return 0; @@ -360,11 +358,12 @@ static int sst_cdev_tstamp(struct device *dev, unsigned int str_id, struct snd_sst_tstamp fw_tstamp = {0,}; struct stream_info *stream; struct intel_sst_drv *ctx = dev_get_drvdata(dev); + void __iomem *addr; + + addr = (void __iomem *)(ctx->mailbox + ctx->tstamp) + + (str_id * sizeof(fw_tstamp)); - memcpy_fromio(&fw_tstamp, - ((void *)(ctx->mailbox + ctx->tstamp) - +(str_id * sizeof(fw_tstamp))), - sizeof(fw_tstamp)); + memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp)); stream = get_stream_info(ctx, str_id); if (!stream) @@ -530,6 +529,7 @@ static int sst_read_timestamp(struct device *dev, struct pcm_stream_info *info) struct snd_sst_tstamp fw_tstamp; unsigned int str_id; struct intel_sst_drv *ctx = dev_get_drvdata(dev); + void __iomem *addr; str_id = info->str_id; stream = get_stream_info(ctx, str_id); @@ -540,10 +540,11 @@ static int sst_read_timestamp(struct device *dev, struct pcm_stream_info *info) return -EINVAL; substream = stream->pcm_substream; - memcpy_fromio(&fw_tstamp, - ((void *)(ctx->mailbox + ctx->tstamp) - + (str_id * sizeof(fw_tstamp))), - sizeof(fw_tstamp)); + addr = (void __iomem *)(ctx->mailbox + ctx->tstamp) + + (str_id * sizeof(fw_tstamp)); + + memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp)); + return sst_calc_tstamp(ctx, info, substream, &fw_tstamp); } diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index a686eef2cf7f..27413ebae956 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -44,15 +44,15 @@ void memcpy32_toio(void __iomem *dst, const void *src, int count) /* __iowrite32_copy uses 32-bit count values so divide by 4 for * right count in words */ - __iowrite32_copy(dst, src, count/4); + __iowrite32_copy(dst, src, count / 4); } void memcpy32_fromio(void *dst, const void __iomem *src, int count) { - /* __iowrite32_copy uses 32-bit count values so divide by 4 for + /* __ioread32_copy uses 32-bit count values so divide by 4 for * right count in words */ - __iowrite32_copy(dst, src, count/4); + __ioread32_copy(dst, src, count / 4); } /** diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 24797482a3d2..cccda87f4b34 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -281,6 +281,20 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH + tristate "GLK with RT5682 and MAX98357A in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + select SND_SOC_RT5682 + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_HDA_DSP_LOADER + help + This adds support for ASoC machine driver for Geminilake platforms + with RT5682 + MAX98357A I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_INTEL_SKYLAKE endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 92b5507291af..87ef8b4058e5 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -6,6 +6,7 @@ snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o snd-soc-sst-bxt-rt298-objs := bxt_rt298.o +snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o @@ -27,6 +28,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o +obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 6ea360f33575..efcfd906c856 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -154,9 +154,7 @@ static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 40eb979d5ac1..6f052fc8d1e2 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -160,7 +160,7 @@ static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } @@ -324,8 +324,22 @@ static const struct snd_pcm_hw_constraint_list constraints_16000 = { .list = rates_16000, }; +static const unsigned int ch_mono[] = { + 1, +}; + +static const struct snd_pcm_hw_constraint_list constraints_refcap = { + .count = ARRAY_SIZE(ch_mono), + .list = ch_mono, +}; + static int broxton_refcap_startup(struct snd_pcm_substream *substream) { + substream->runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_refcap); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_16000); @@ -586,7 +600,7 @@ static int broxton_audio_probe(struct platform_device *pdev) static struct platform_driver broxton_audio = { .probe = broxton_audio_probe, .driver = { - .name = "bxt_da7219_max98357a_i2s", + .name = "bxt_da7219_max98357a", .pm = &snd_soc_pm_ops, }, }; @@ -599,4 +613,4 @@ MODULE_AUTHOR("Rohit Ainapure <rohit.m.ainapure@intel.com>"); MODULE_AUTHOR("Harsha Priya <harshapriya.n@intel.com>"); MODULE_AUTHOR("Conrad Cooke <conrad.cooke@intel.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:bxt_da7219_max98357a_i2s"); +MODULE_ALIAS("platform:bxt_da7219_max98357a"); diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index b68c289558a8..27308337ab12 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -221,7 +221,7 @@ static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP5 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 33065ba294a9..d32844f94d74 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -404,7 +404,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { }, .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | BYT_RT5640_JD_SRC_JD1_IN4P | - BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_TH_1500UA | BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), @@ -464,12 +464,38 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MCLK_EN), }, { + /* Chuwi Vi10 (CWI505) */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"), + DMI_MATCH(DMI_BOARD_NAME, "BYT-PF02"), + DMI_MATCH(DMI_SYS_VENDOR, "ilife"), + DMI_MATCH(DMI_PRODUCT_NAME, "S165"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, + { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"), }, .driver_data = (void *)(BYT_RT5640_DMIC1_MAP), }, + { /* Connect Tablet 9 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Connect"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "Tablet 9"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Dell Inc."), @@ -536,6 +562,19 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Lenovo Miix 2 8 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "20326"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Hiking"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), @@ -549,6 +588,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_DIFF_MIC | BYT_RT5640_MCLK_EN), }, + { /* Nuvison/TMax TM800W560 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TMAX"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TM800W560L"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Pipo W4 */ .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 987720e203f9..f8a68bdb3885 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -26,8 +26,12 @@ #include <linux/clk.h> #include <linux/device.h> #include <linux/dmi.h> +#include <linux/input.h> +#include <linux/gpio/consumer.h> +#include <linux/gpio/machine.h> #include <linux/slab.h> #include <asm/cpu_device_id.h> +#include <asm/intel-family.h> #include <asm/platform_sst_audio.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -42,8 +46,6 @@ enum { BYT_RT5651_IN1_MAP, BYT_RT5651_IN2_MAP, BYT_RT5651_IN1_IN2_MAP, - BYT_RT5651_IN1_HS_IN3_MAP, - BYT_RT5651_IN2_HS_IN3_MAP, }; enum { @@ -76,21 +78,26 @@ enum { #define BYT_RT5651_SSP2_AIF2 BIT(19) /* default is using AIF1 */ #define BYT_RT5651_SSP0_AIF1 BIT(20) #define BYT_RT5651_SSP0_AIF2 BIT(21) +#define BYT_RT5651_HP_LR_SWAPPED BIT(22) +#define BYT_RT5651_MONO_SPEAKER BIT(23) + +#define BYT_RT5651_DEFAULT_QUIRKS (BYT_RT5651_MCLK_EN | \ + BYT_RT5651_JD1_1 | \ + BYT_RT5651_OVCD_TH_2000UA | \ + BYT_RT5651_OVCD_SF_0P75) /* jack-detect-source + dmic-en + ovcd-th + -sf + terminating empty entry */ #define MAX_NO_PROPS 5 struct byt_rt5651_private { struct clk *mclk; + struct gpio_desc *ext_amp_gpio; struct snd_soc_jack jack; }; /* Default: jack-detect on JD1_1, internal mic on in2, headsetmic on in3 */ -static unsigned long byt_rt5651_quirk = BYT_RT5651_MCLK_EN | - BYT_RT5651_JD1_1 | - BYT_RT5651_OVCD_TH_2000UA | - BYT_RT5651_OVCD_SF_0P75 | - BYT_RT5651_IN2_HS_IN3_MAP; +static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP; static void log_quirks(struct device *dev) { @@ -100,10 +107,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk IN1_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_MAP) dev_info(dev, "quirk IN2_MAP enabled"); - if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_HS_IN3_MAP) - dev_info(dev, "quirk IN1_HS_IN3_MAP enabled"); - if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_HS_IN3_MAP) - dev_info(dev, "quirk IN2_HS_IN3_MAP enabled"); + if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_IN2_MAP) + dev_info(dev, "quirk IN1_IN2_MAP enabled"); if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) { dev_info(dev, "quirk realtek,jack-detect-source %ld\n", BYT_RT5651_JDSRC(byt_rt5651_quirk)); @@ -124,6 +129,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk SSP0_AIF1 enabled\n"); if (byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2) dev_info(dev, "quirk SSP0_AIF2 enabled\n"); + if (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) + dev_info(dev, "quirk MONO_SPEAKER enabled\n"); } #define BYT_CODEC_DAI1 "rt5651-aif1" @@ -211,6 +218,20 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return 0; } +static int rt5651_ext_amp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card); + + if (SND_SOC_DAPM_EVENT_ON(event)) + gpiod_set_value_cansleep(priv->ext_amp_gpio, 1); + else + gpiod_set_value_cansleep(priv->ext_amp_gpio, 0); + + return 0; +} + static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -220,7 +241,9 @@ static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = { SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - + SND_SOC_DAPM_SUPPLY("Ext Amp Power", SND_SOC_NOPM, 0, 0, + rt5651_ext_amp_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), }; static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { @@ -228,6 +251,7 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { {"Headset Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "Platform Clock"}, {"Speaker", NULL, "Platform Clock"}, + {"Speaker", NULL, "Ext Amp Power"}, {"Line In", NULL, "Platform Clock"}, {"Headset Mic", NULL, "micbias1"}, /* lowercase for rt5651 */ @@ -241,38 +265,26 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic_map[] = { - {"IN2P", NULL, "Headset Mic"}, {"DMIC L1", NULL, "Internal Mic"}, {"DMIC R1", NULL, "Internal Mic"}, + {"IN3P", NULL, "Headset Mic"}, }; static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = { {"Internal Mic", NULL, "micbias1"}, {"IN1P", NULL, "Internal Mic"}, - {"IN2P", NULL, "Headset Mic"}, + {"IN3P", NULL, "Headset Mic"}, }; static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_map[] = { {"Internal Mic", NULL, "micbias1"}, - {"IN1P", NULL, "Headset Mic"}, - {"IN2P", NULL, "Internal Mic"}, -}; - -static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = { - {"Internal Mic", NULL, "micbias1"}, - {"IN1P", NULL, "Internal Mic"}, {"IN2P", NULL, "Internal Mic"}, {"IN3P", NULL, "Headset Mic"}, }; -static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_hs_in3_map[] = { +static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = { {"Internal Mic", NULL, "micbias1"}, {"IN1P", NULL, "Internal Mic"}, - {"IN3P", NULL, "Headset Mic"}, -}; - -static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_hs_in3_map[] = { - {"Internal Mic", NULL, "micbias1"}, {"IN2P", NULL, "Internal Mic"}, {"IN3P", NULL, "Headset Mic"}, }; @@ -357,46 +369,72 @@ static int byt_rt5651_quirk_cb(const struct dmi_system_id *id) static const struct dmi_system_id byt_rt5651_quirk_table[] = { { + /* Chuwi Hi8 Pro (CWI513) */ .callback = byt_rt5651_quirk_cb, .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), - DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"), + DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"), + DMI_MATCH(DMI_PRODUCT_NAME, "X1D3_C806N"), }, - .driver_data = (void *)(BYT_RT5651_IN1_HS_IN3_MAP), + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_HP_LR_SWAPPED | + BYT_RT5651_MONO_SPEAKER), }, { + /* Chuwi Vi8 Plus (CWI519) */ .callback = byt_rt5651_quirk_cb, .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "ADI"), - DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"), + DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"), + DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), }, - .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_IN1_HS_IN3_MAP), + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_HP_LR_SWAPPED | + BYT_RT5651_MONO_SPEAKER), + }, + { + /* I.T.Works TW701, Ployer Momo7w and Trekstor ST70416-6 + * (these all use the same mainboard) */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_BIOS_VENDOR, "INSYDE Corp."), + /* Partial match for all of itWORKS.G.WI71C.JGBMRBA, + * TREK.G.WI71C.JGBMRBA0x and MOMO.G.WI71C.MABMRBA02 */ + DMI_MATCH(DMI_BIOS_VERSION, ".G.WI71C."), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_SSP0_AIF1 | + BYT_RT5651_MONO_SPEAKER), }, { + /* KIANO SlimNote 14.2 */ .callback = byt_rt5651_quirk_cb, .matches = { DMI_MATCH(DMI_SYS_VENDOR, "KIANO"), DMI_MATCH(DMI_PRODUCT_NAME, "KIANO SlimNote 14.2"), }, - .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_JD1_1 | - BYT_RT5651_OVCD_TH_2000UA | - BYT_RT5651_OVCD_SF_0P75 | + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN1_IN2_MAP), }, { - /* Chuwi Vi8 Plus (CWI519) */ + /* Minnowboard Max B3 */ .callback = byt_rt5651_quirk_cb, .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"), - DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), + DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), + DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"), + }, + .driver_data = (void *)(BYT_RT5651_IN1_MAP), + }, + { + /* Minnowboard Turbot */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ADI"), + DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"), }, .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_JD1_1 | - BYT_RT5651_OVCD_TH_2000UA | - BYT_RT5651_OVCD_SF_0P75 | - BYT_RT5651_IN2_HS_IN3_MAP), + BYT_RT5651_IN1_MAP), }, { /* VIOS LTH17 */ @@ -405,11 +443,24 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "VIOS"), DMI_MATCH(DMI_PRODUCT_NAME, "LTH17"), }, - .driver_data = (void *)(BYT_RT5651_MCLK_EN | + .driver_data = (void *)(BYT_RT5651_IN1_IN2_MAP | BYT_RT5651_JD1_1 | BYT_RT5651_OVCD_TH_2000UA | BYT_RT5651_OVCD_SF_1P0 | - BYT_RT5651_IN1_IN2_MAP), + BYT_RT5651_MCLK_EN), + }, + { + /* Yours Y8W81 (and others using the same mainboard) */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_BIOS_VENDOR, "INSYDE Corp."), + /* Partial match for all devs with a W86C mainboard */ + DMI_MATCH(DMI_BIOS_VERSION, ".F.W86C."), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_SSP0_AIF1 | + BYT_RT5651_MONO_SPEAKER), }, {} }; @@ -418,15 +469,10 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { * Note this MUST be called before snd_soc_register_card(), so that the props * are in place before the codec component driver's probe function parses them. */ -static int byt_rt5651_add_codec_device_props(const char *i2c_dev_name) +static int byt_rt5651_add_codec_device_props(struct device *i2c_dev) { struct property_entry props[MAX_NO_PROPS] = {}; - struct device *i2c_dev; - int ret, cnt = 0; - - i2c_dev = bus_find_device_by_name(&i2c_bus_type, NULL, i2c_dev_name); - if (!i2c_dev) - return -EPROBE_DEFER; + int cnt = 0; props[cnt++] = PROPERTY_ENTRY_U32("realtek,jack-detect-source", BYT_RT5651_JDSRC(byt_rt5651_quirk)); @@ -440,10 +486,7 @@ static int byt_rt5651_add_codec_device_props(const char *i2c_dev_name) if (byt_rt5651_quirk & BYT_RT5651_DMIC_EN) props[cnt++] = PROPERTY_ENTRY_BOOL("realtek,dmic-en"); - ret = device_add_properties(i2c_dev, props); - put_device(i2c_dev); - - return ret; + return device_add_properties(i2c_dev, props); } static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) @@ -475,14 +518,6 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5651_intmic_in1_in2_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_in2_map); break; - case BYT_RT5651_IN1_HS_IN3_MAP: - custom_map = byt_rt5651_intmic_in1_hs_in3_map; - num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_hs_in3_map); - break; - case BYT_RT5651_IN2_HS_IN3_MAP: - custom_map = byt_rt5651_intmic_in2_hs_in3_map; - num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_hs_in3_map); - break; default: custom_map = byt_rt5651_intmic_dmic_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic_map); @@ -546,13 +581,17 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) { ret = snd_soc_card_jack_new(runtime->card, "Headset", - SND_JACK_HEADSET, &priv->jack, - bytcr_jack_pins, ARRAY_SIZE(bytcr_jack_pins)); + SND_JACK_HEADSET | SND_JACK_BTN_0, + &priv->jack, bytcr_jack_pins, + ARRAY_SIZE(bytcr_jack_pins)); if (ret) { dev_err(runtime->dev, "jack creation failed %d\n", ret); return ret; } + snd_jack_set_key(priv->jack.jack, SND_JACK_BTN_0, + KEY_PLAYPAUSE); + ret = snd_soc_component_set_jack(codec, &priv->jack, NULL); if (ret) return ret; @@ -691,6 +730,48 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { }; /* SoC card */ +static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; +static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ +static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ +static char byt_rt5651_long_name[50]; /* = "bytcr-rt5651-*-spk-*-mic[-swapped-hp]" */ + +static int byt_rt5651_suspend(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + + if (!BYT_RT5651_JDSRC(byt_rt5651_quirk)) + return 0; + + list_for_each_entry(component, &card->component_dev_list, card_list) { + if (!strcmp(component->name, byt_rt5651_codec_name)) { + dev_dbg(component->dev, "disabling jack detect before suspend\n"); + snd_soc_component_set_jack(component, NULL, NULL); + break; + } + } + + return 0; +} + +static int byt_rt5651_resume(struct snd_soc_card *card) +{ + struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component; + + if (!BYT_RT5651_JDSRC(byt_rt5651_quirk)) + return 0; + + list_for_each_entry(component, &card->component_dev_list, card_list) { + if (!strcmp(component->name, byt_rt5651_codec_name)) { + dev_dbg(component->dev, "re-enabling jack detect after resume\n"); + snd_soc_component_set_jack(component, &priv->jack, NULL); + break; + } + } + + return 0; +} + static struct snd_soc_card byt_rt5651_card = { .name = "bytcr-rt5651", .owner = THIS_MODULE, @@ -701,23 +782,86 @@ static struct snd_soc_card byt_rt5651_card = { .dapm_routes = byt_rt5651_audio_map, .num_dapm_routes = ARRAY_SIZE(byt_rt5651_audio_map), .fully_routed = true, + .suspend_pre = byt_rt5651_suspend, + .resume_post = byt_rt5651_resume, }; -static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; -static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ -static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ -static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-spk-*-mic" */ +static const struct x86_cpu_id baytrail_cpu_ids[] = { + { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_SILVERMONT1 }, /* Valleyview */ + {} +}; + +static const struct x86_cpu_id cherrytrail_cpu_ids[] = { + { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_AIRMONT }, /* Braswell */ + {} +}; + +static const struct acpi_gpio_params first_gpio = { 0, 0, false }; +static const struct acpi_gpio_params second_gpio = { 1, 0, false }; + +static const struct acpi_gpio_mapping byt_rt5651_amp_en_first[] = { + { "ext-amp-enable-gpios", &first_gpio, 1 }, + { }, +}; -static bool is_valleyview(void) +static const struct acpi_gpio_mapping byt_rt5651_amp_en_second[] = { + { "ext-amp-enable-gpios", &second_gpio, 1 }, + { }, +}; + +/* + * Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other + * boards may have I2cSerialBusV2, GpioInt, GpioIo instead. We want the + * GpioIo one for the ext-amp-enable-gpio and both count for the index in + * acpi_gpio_params index. So we have 2 different mappings and the code + * below figures out which one to use. + */ +struct byt_rt5651_acpi_resource_data { + int gpio_count; + int gpio_int_idx; +}; + +static int snd_byt_rt5651_acpi_resource(struct acpi_resource *ares, void *arg) { - static const struct x86_cpu_id cpu_ids[] = { - { X86_VENDOR_INTEL, 6, 55 }, /* Valleyview, Bay Trail */ - {} - }; - - if (!x86_match_cpu(cpu_ids)) - return false; - return true; + struct byt_rt5651_acpi_resource_data *data = arg; + + if (ares->type != ACPI_RESOURCE_TYPE_GPIO) + return 0; + + if (ares->data.gpio.connection_type == ACPI_RESOURCE_GPIO_TYPE_INT) + data->gpio_int_idx = data->gpio_count; + + data->gpio_count++; + return 0; +} + +static void snd_byt_rt5651_mc_add_amp_en_gpio_mapping(struct device *codec) +{ + struct byt_rt5651_acpi_resource_data data = { 0, -1 }; + LIST_HEAD(resources); + int ret; + + ret = acpi_dev_get_resources(ACPI_COMPANION(codec), &resources, + snd_byt_rt5651_acpi_resource, &data); + if (ret < 0) { + dev_warn(codec, "Failed to get ACPI resources, not adding external amplifier GPIO mapping\n"); + return; + } + + /* All info we need is gathered during the walk */ + acpi_dev_free_resource_list(&resources); + + switch (data.gpio_int_idx) { + case 0: + devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_second); + break; + case 1: + devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_first); + break; + default: + dev_warn(codec, "Unknown GpioInt index %d, not adding external amplifier GPIO mapping\n", + data.gpio_int_idx); + } } struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ @@ -727,13 +871,12 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { - const char * const intmic_name[] = - { "dmic", "in1", "in2", "in12", "in1", "in2" }; - const char * const hsmic_name[] = - { "in2", "in2", "in1", "in3", "in3", "in3" }; + const char * const mic_name[] = { "dmic", "in1", "in2", "in12" }; struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; + struct device *codec_dev; const char *i2c_name = NULL; + const char *hp_swapped; bool is_bytcr = false; int ret_val = 0; int dai_index = 0; @@ -767,11 +910,16 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) "%s%s", "i2c-", i2c_name); byt_rt5651_dais[dai_index].codec_name = byt_rt5651_codec_name; + codec_dev = bus_find_device_by_name(&i2c_bus_type, NULL, + byt_rt5651_codec_name); + if (!codec_dev) + return -EPROBE_DEFER; + /* * swap SSP0 if bytcr is detected * (will be overridden if DMI quirk is detected) */ - if (is_valleyview()) { + if (x86_match_cpu(baytrail_cpu_ids)) { struct sst_platform_info *p_info = mach->pdata; const struct sst_res_info *res_info = p_info->res_info; @@ -830,9 +978,37 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) dmi_check_system(byt_rt5651_quirk_table); /* Must be called before register_card, also see declaration comment. */ - ret_val = byt_rt5651_add_codec_device_props(byt_rt5651_codec_name); - if (ret_val) + ret_val = byt_rt5651_add_codec_device_props(codec_dev); + if (ret_val) { + put_device(codec_dev); return ret_val; + } + + /* Cherry Trail devices use an external amplifier enable gpio */ + if (x86_match_cpu(cherrytrail_cpu_ids)) { + snd_byt_rt5651_mc_add_amp_en_gpio_mapping(codec_dev); + priv->ext_amp_gpio = devm_fwnode_get_index_gpiod_from_child( + &pdev->dev, "ext-amp-enable", 0, + codec_dev->fwnode, + GPIOD_OUT_LOW, "speaker-amp"); + if (IS_ERR(priv->ext_amp_gpio)) { + ret_val = PTR_ERR(priv->ext_amp_gpio); + switch (ret_val) { + case -ENOENT: + priv->ext_amp_gpio = NULL; + break; + default: + dev_err(&pdev->dev, "Failed to get ext-amp-enable GPIO: %d\n", + ret_val); + /* fall through */ + case -EPROBE_DEFER: + put_device(codec_dev); + return ret_val; + } + } + } + + put_device(codec_dev); log_quirks(&pdev->dev); @@ -876,10 +1052,16 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) } } + if (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED) + hp_swapped = "-hp-swapped"; + else + hp_swapped = ""; + snprintf(byt_rt5651_long_name, sizeof(byt_rt5651_long_name), - "bytcr-rt5651-%s-intmic-%s-hsmic", - intmic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], - hsmic_name[BYT_RT5651_MAP(byt_rt5651_quirk)]); + "bytcr-rt5651-%s-spk-%s-mic%s", + (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) ? + "mono" : "stereo", + mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], hp_swapped); byt_rt5651_card.long_name = byt_rt5651_long_name; ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_rt5651_card); diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c new file mode 100644 index 000000000000..c4b94e2617c5 --- /dev/null +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -0,0 +1,643 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018 Intel Corporation. + +/* + * Intel Geminilake I2S Machine Driver with MAX98357A & RT5682 Codecs + * + * Modified from: + * Intel Apollolake I2S Machine driver + */ + +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../skylake/skl.h" +#include "../../codecs/rt5682.h" +#include "../../codecs/hdac_hdmi.h" + +/* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */ +#define GLK_PLAT_CLK_FREQ 19200000 +#define RT5682_PLL_FREQ (48000 * 512) +#define GLK_REALTEK_CODEC_DAI "rt5682-aif1" +#define GLK_MAXIM_CODEC_DAI "HiFi" +#define MAXIM_DEV0_NAME "MX98357A:00" +#define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 +#define NAME_SIZE 32 + +static struct snd_soc_jack geminilake_hdmi[3]; + +struct glk_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct glk_card_private { + struct snd_soc_jack geminilake_headset; + struct list_head hdmi_pcm_list; +}; + +enum { + GLK_DPCM_AUDIO_PB = 0, + GLK_DPCM_AUDIO_CP, + GLK_DPCM_AUDIO_HS_PB, + GLK_DPCM_AUDIO_ECHO_REF_CP, + GLK_DPCM_AUDIO_REF_CP, + GLK_DPCM_AUDIO_DMIC_CP, + GLK_DPCM_AUDIO_HDMI1_PB, + GLK_DPCM_AUDIO_HDMI2_PB, + GLK_DPCM_AUDIO_HDMI3_PB, +}; + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop sysclk: %d\n", ret); + } else if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(card->dev, "can't set codec pll: %d\n", ret); + return ret; + } + } + + if (ret) + dev_err(card->dev, "failed to start internal clk: %d\n", ret); + + return ret; +} + +static const struct snd_kcontrol_new geminilake_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget geminilake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SPK("HDMI1", NULL), + SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SPK("HDMI3", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route geminilake_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* speaker */ + { "Spk", NULL, "Speaker" }, + + /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, + { "IN1P", NULL, "Headset Mic" }, + + /* digital mics */ + { "DMic", NULL, "SoC DMIC" }, + + /* CODEC BE connections */ + { "HiFi Playback", NULL, "ssp1 Tx" }, + { "ssp1 Tx", NULL, "codec0_out" }, + + { "AIF1 Playback", NULL, "ssp2 Tx" }, + { "ssp2 Tx", NULL, "codec1_out" }, + + { "codec0_in", NULL, "ssp2 Rx" }, + { "ssp2 Rx", NULL, "AIF1 Capture" }, + + { "HDMI1", NULL, "hif5-0 Output" }, + { "HDMI2", NULL, "hif6-0 Output" }, + { "HDMI2", NULL, "hif7-0 Output" }, + + { "hifi3", NULL, "iDisp3 Tx" }, + { "iDisp3 Tx", NULL, "iDisp3_out" }, + { "hifi2", NULL, "iDisp2 Tx" }, + { "iDisp2 Tx", NULL, "iDisp2_out" }, + { "hifi1", NULL, "iDisp1 Tx" }, + { "iDisp1 Tx", NULL, "iDisp1_out" }, + + /* DMIC */ + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "DMIC AIF" }, +}; + +static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = DUAL_CHANNEL; + + /* set SSP to 24 bit */ + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_jack *jack; + int ret; + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->geminilake_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->geminilake_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +}; + +static int geminilake_rt5682_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* Set valid bitmask & configuration for I2S in 24 bit */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, 24); + if (ret < 0) { + dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + return ret; +} + +static struct snd_soc_ops geminilake_rt5682_ops = { + .hw_params = geminilake_rt5682_hw_params, +}; + +static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct glk_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = GLK_DPCM_AUDIO_HDMI1_PB + dai->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_dapm_context *dapm; + int ret; + + dapm = snd_soc_component_get_dapm(component); + ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); + if (ret) { + dev_err(rtd->dev, "Ref Cap ignore suspend failed %d\n", ret); + return ret; + } + + return ret; +} + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + +static int geminilake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* + * set BE channel constraint as user FE channels + */ + channels->min = channels->max = 4; + + return 0; +} + +static int geminilake_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels_quad); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +static const struct snd_soc_ops geminilake_dmic_ops = { + .startup = geminilake_dmic_startup, +}; + +static const unsigned int rates_16000[] = { + 16000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_16000 = { + .count = ARRAY_SIZE(rates_16000), + .list = rates_16000, +}; + +static int geminilake_refcap_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +}; + +static const struct snd_soc_ops geminilake_refcap_ops = { + .startup = geminilake_refcap_startup, +}; + +/* geminilake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link geminilake_dais[] = { + /* Front End DAI links */ + [GLK_DPCM_AUDIO_PB] = { + .name = "Glk Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = geminilake_rt5682_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + [GLK_DPCM_AUDIO_CP] = { + .name = "Glk Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + [GLK_DPCM_AUDIO_HS_PB] = { + .name = "Glk Audio Headset Playback", + .stream_name = "Headset Audio", + .cpu_dai_name = "System Pin2", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dpcm_playback = 1, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Glk Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [GLK_DPCM_AUDIO_REF_CP] = { + .name = "Glk Audio Reference cap", + .stream_name = "Refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_refcap_ops, + }, + [GLK_DPCM_AUDIO_DMIC_CP] = { + .name = "Glk Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_dmic_ops, + }, + [GLK_DPCM_AUDIO_HDMI1_PB] = { + .name = "Glk HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI2_PB] = { + .name = "Glk HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI3_PB] = { + .name = "Glk HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + /* Back End DAI links */ + { + /* SSP1 - Codec */ + .name = "SSP1-Codec", + .id = 0, + .cpu_dai_name = "SSP1 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = MAXIM_DEV0_NAME, + .codec_dai_name = GLK_MAXIM_CODEC_DAI, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .dpcm_playback = 1, + }, + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .id = 1, + .cpu_dai_name = "SSP2 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = "i2c-10EC5682:00", + .codec_dai_name = GLK_REALTEK_CODEC_DAI, + .init = geminilake_rt5682_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .ops = &geminilake_rt5682_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .id = 2, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:0e.0", + .ignore_suspend = 1, + .be_hw_params_fixup = geminilake_dmic_fixup, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 3, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 4, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 5, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +static int glk_card_late_probe(struct snd_soc_card *card) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = NULL; + char jack_name[NAME_SIZE]; + struct glk_hdmi_pcm *pcm; + int err = 0; + int i = 0; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &geminilake_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &geminilake_hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} + +/* geminilake audio machine driver for SPT + RT5682 */ +static struct snd_soc_card glk_audio_card_rt5682_m98357a = { + .name = "glkrt5682max", + .owner = THIS_MODULE, + .dai_link = geminilake_dais, + .num_links = ARRAY_SIZE(geminilake_dais), + .controls = geminilake_controls, + .num_controls = ARRAY_SIZE(geminilake_controls), + .dapm_widgets = geminilake_widgets, + .num_dapm_widgets = ARRAY_SIZE(geminilake_widgets), + .dapm_routes = geminilake_map, + .num_dapm_routes = ARRAY_SIZE(geminilake_map), + .fully_routed = true, + .late_probe = glk_card_late_probe, +}; + +static int geminilake_audio_probe(struct platform_device *pdev) +{ + struct glk_card_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + glk_audio_card_rt5682_m98357a.dev = &pdev->dev; + snd_soc_card_set_drvdata(&glk_audio_card_rt5682_m98357a, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &glk_audio_card_rt5682_m98357a); +} + +static const struct platform_device_id glk_board_ids[] = { + { + .name = "glk_rt5682_max98357a", + .driver_data = + (kernel_ulong_t)&glk_audio_card_rt5682_m98357a, + }, + { } +}; + +static struct platform_driver geminilake_audio = { + .probe = geminilake_audio_probe, + .driver = { + .name = "glk_rt5682_max98357a", + .pm = &snd_soc_pm_ops, + }, + .id_table = glk_board_ids, +}; +module_platform_driver(geminilake_audio) + +/* Module information */ +MODULE_DESCRIPTION("Geminilake Audio Machine driver-RT5682 & MAX98357A in I2S mode"); +MODULE_AUTHOR("Naveen Manohar <naveen.m@intel.com>"); +MODULE_AUTHOR("Harsha Priya <harshapriya.n@intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:glk_rt5682_max98357a"); diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 94294c27d1db..38f6ab74709d 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -152,7 +152,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } @@ -380,6 +380,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .trigger = { SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, + .ops = &kabylake_da7219_fe_ops, }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 3a61252fe450..21a6490746a6 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -434,14 +434,14 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, rate->min = rate->max = 48000; channels->min = channels->max = 2; snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); } /* * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ if (!strcmp(be_dai_link->name, "SSP0-Codec")) - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 92f5fb2ae0a3..a892b37eab7c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -307,7 +307,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, rate->min = rate->max = 48000; channels->min = channels->max = 2; snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) @@ -320,7 +320,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * thus changing the mask here */ if (!strcmp(be_dai_link->name, "SSP0-Codec")) - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 3ff6646cfa21..d31482b8c9bb 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -157,7 +157,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP0 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index b0610bba3cfa..e877bb60beb1 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -346,7 +346,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP0 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 38a1495c29cf..0e1818dd4cc6 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -229,7 +229,7 @@ static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP0 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 7379d8830c39..915a34cdc8ac 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -3,7 +3,11 @@ snd-soc-sst-dsp-objs := sst-dsp.o snd-soc-sst-acpi-objs := sst-acpi.o snd-soc-sst-ipc-objs := sst-ipc.o snd-soc-sst-firmware-objs := sst-firmware.o -snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o soc-acpi-intel-hsw-bdw-match.o +snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o \ + soc-acpi-intel-hsw-bdw-match.o \ + soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \ + soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \ + soc-acpi-intel-cnl-match.o obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c new file mode 100644 index 000000000000..f39386e540d3 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c @@ -0,0 +1,59 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-bxt-match.c - tables and support for BXT ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> + +static struct snd_soc_acpi_codecs bxt_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = { + { + .id = "INT343A", + .drv_name = "bxt_alc298s_i2s", + .fw_filename = "intel/dsp_fw_bxtn.bin", + }, + { + .id = "DLGS7219", + .drv_name = "bxt_da7219_max98357a", + .fw_filename = "intel/dsp_fw_bxtn.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &bxt_codecs, + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-da7219.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + { + .id = "104C5122", + .drv_name = "bxt-pcm512x", + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-pcm512x.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + { + .id = "1AEC8804", + .drv_name = "bxt-wm8804", + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-wm8804.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + { + .id = "INT34C3", + .drv_name = "bxt_tdf8532", + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-tdf8532.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_bxt_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c index bfe1ca68a542..4daa8a4f0c0c 100644 --- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c @@ -59,8 +59,8 @@ static struct snd_soc_acpi_mach byt_thinkpad_10 = { .drv_name = "cht-bsw-rt5672", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5670.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5670.tplg", .asoc_plat_name = "sst-mfld-platform", }; @@ -98,8 +98,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5640", .machine_quirk = byt_quirk, - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5640.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -107,8 +107,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcr_rt5640", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5640", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5640.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -116,8 +116,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcr_rt5640", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5640", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5640.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -125,8 +125,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcr_rt5651", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5651", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5651.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5651.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -134,8 +134,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-da7213.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -143,8 +143,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-da7213.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* some Baytrail platforms rely on RT5645, use CHT machine driver */ @@ -153,8 +153,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5645.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -162,8 +162,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5645.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* use CHT driver to Baytrail Chromebooks */ @@ -172,8 +172,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "cht-bsw-max98090", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-max98090.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-max98090.tplg", .asoc_plat_name = "sst-mfld-platform", }, #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index ad1eb2d644be..91bb99b69601 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -44,8 +44,8 @@ static struct snd_soc_acpi_mach cht_surface_mach = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }; @@ -68,8 +68,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5672", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5670.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5670.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -77,8 +77,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5672", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5670.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5670.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -86,8 +86,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -95,8 +95,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -104,8 +104,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -113,8 +113,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-max98090", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-max98090.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-max98090.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -122,8 +122,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-nau8824", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-nau8824.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-nau8824.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -131,8 +131,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-da7213.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -140,8 +140,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-da7213.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -149,8 +149,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcht_es8316", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcht_es8316", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-es8316.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-es8316.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* some CHT-T platforms rely on RT5640, use Baytrail machine driver */ @@ -160,8 +160,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcr_rt5640", .machine_quirk = cht_quirk, - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5640.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -169,8 +169,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcr_rt5640", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcr_rt5640", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5640.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* some CHT-T platforms rely on RT5651, use Baytrail machine driver */ @@ -179,8 +179,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcr_rt5651", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcr_rt5651", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5651.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5651.tplg", .asoc_plat_name = "sst-mfld-platform", }, #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c new file mode 100644 index 000000000000..ec8e28e7b937 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -0,0 +1,32 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-cnl-match.c - tables and support for CNL ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> +#include "../skylake/skl.h" + +static struct skl_machine_pdata cnl_pdata = { + .use_tplg_pcm = true, +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { + { + .id = "INT34C2", + .drv_name = "cnl_rt274", + .fw_filename = "intel/dsp_fw_cnl.bin", + .pdata = &cnl_pdata, + .sof_fw_filename = "intel/sof-cnl.ri", + .sof_tplg_filename = "intel/sof-cnl-rt274.tplg", + .asoc_plat_name = "0000:00:1f.3", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cnl_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c new file mode 100644 index 000000000000..305875af71ca --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -0,0 +1,41 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-glk-match.c - tables and support for GLK ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> + +static struct snd_soc_acpi_codecs glk_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { + { + .id = "INT343A", + .drv_name = "glk_alc298s_i2s", + .fw_filename = "intel/dsp_fw_glk.bin", + .sof_fw_filename = "intel/sof-glk.ri", + .sof_tplg_filename = "intel/sof-glk-alc298.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + { + .id = "DLGS7219", + .drv_name = "glk_da7219_max98357a", + .fw_filename = "intel/dsp_fw_glk.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &glk_codecs, + .sof_fw_filename = "intel/sof-glk.ri", + .sof_tplg_filename = "intel/sof-glk-da7219.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_glk_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index e0e8c8c27528..494a0ea9b029 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -23,8 +23,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_haswell_machines[] = { .id = "INT33CA", .drv_name = "haswell-audio", .fw_filename = "intel/IntcSST1.bin", - .sof_fw_filename = "intel/reef-hsw.ri", - .sof_tplg_filename = "intel/reef-hsw.tplg", + .sof_fw_filename = "intel/sof-hsw.ri", + .sof_tplg_filename = "intel/sof-hsw.tplg", .asoc_plat_name = "haswell-pcm-audio", }, {} @@ -36,24 +36,24 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { .id = "INT343A", .drv_name = "broadwell-audio", .fw_filename = "intel/IntcSST2.bin", - .sof_fw_filename = "intel/reef-bdw.ri", - .sof_tplg_filename = "intel/reef-bdw-rt286.tplg", + .sof_fw_filename = "intel/sof-bdw.ri", + .sof_tplg_filename = "intel/sof-bdw-rt286.tplg", .asoc_plat_name = "haswell-pcm-audio", }, { .id = "RT5677CE", .drv_name = "bdw-rt5677", .fw_filename = "intel/IntcSST2.bin", - .sof_fw_filename = "intel/reef-bdw.ri", - .sof_tplg_filename = "intel/reef-bdw-rt286.tplg", + .sof_fw_filename = "intel/sof-bdw.ri", + .sof_tplg_filename = "intel/sof-bdw-rt5677.tplg", .asoc_plat_name = "haswell-pcm-audio", }, { .id = "INT33CA", .drv_name = "haswell-audio", .fw_filename = "intel/IntcSST2.bin", - .sof_fw_filename = "intel/reef-bdw.ri", - .sof_tplg_filename = "intel/reef-bdw-rt5640.tplg", + .sof_fw_filename = "intel/sof-bdw.ri", + .sof_tplg_filename = "intel/sof-bdw-rt5640.tplg", .asoc_plat_name = "haswell-pcm-audio", }, {} diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c new file mode 100644 index 000000000000..0ee173ca437d --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -0,0 +1,91 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-kbl-match.c - tables and support for KBL ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> +#include "../skylake/skl.h" + +static struct skl_machine_pdata skl_dmic_data; + +static struct snd_soc_acpi_codecs kbl_codecs = { + .num_codecs = 1, + .codecs = {"10508825"} +}; + +static struct snd_soc_acpi_codecs kbl_poppy_codecs = { + .num_codecs = 1, + .codecs = {"10EC5663"} +}; + +static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = { + .num_codecs = 2, + .codecs = {"10EC5663", "10EC5514"} +}; + +static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { + { + .id = "INT343A", + .drv_name = "kbl_alc286s_i2s", + .fw_filename = "intel/dsp_fw_kbl.bin", + }, + { + .id = "INT343B", + .drv_name = "kbl_n88l25_s4567", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98357A", + .drv_name = "kbl_n88l25_m98357a", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98927", + .drv_name = "kbl_r5514_5663_max", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_5663_5514_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98927", + .drv_name = "kbl_rt5663_m98927", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_poppy_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "10EC5663", + .drv_name = "kbl_rt5663", + .fw_filename = "intel/dsp_fw_kbl.bin", + }, + { + .id = "DLGS7219", + .drv_name = "kbl_da7219_max98357a", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_7219_98357_codecs, + .pdata = &skl_dmic_data, + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-skl-match.c b/sound/soc/intel/common/soc-acpi-intel-skl-match.c new file mode 100644 index 000000000000..0c9c0edd35b3 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-skl-match.c @@ -0,0 +1,47 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-skl-match.c - tables and support for SKL ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> +#include "../skylake/skl.h" + +static struct skl_machine_pdata skl_dmic_data; + +static struct snd_soc_acpi_codecs skl_codecs = { + .num_codecs = 1, + .codecs = {"10508825"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_skl_machines[] = { + { + .id = "INT343A", + .drv_name = "skl_alc286s_i2s", + .fw_filename = "intel/dsp_fw_release.bin", + }, + { + .id = "INT343B", + .drv_name = "skl_n88l25_s4567", + .fw_filename = "intel/dsp_fw_release.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &skl_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98357A", + .drv_name = "skl_n88l25_m98357a", + .fw_filename = "intel/dsp_fw_release.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &skl_codecs, + .pdata = &skl_dmic_data, + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_skl_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 657afc02f1c4..11041aedea31 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -270,7 +270,7 @@ void sst_dsp_dma_put_channel(struct sst_dsp *dsp) } EXPORT_SYMBOL_GPL(sst_dsp_dma_put_channel); -int sst_dma_new(struct sst_dsp *sst) +static int sst_dma_new(struct sst_dsp *sst) { struct sst_pdata *sst_pdata = sst->pdata; struct sst_dma *dma; @@ -320,9 +320,8 @@ err_dma_dev: devm_kfree(sst->dev, dma); return ret; } -EXPORT_SYMBOL(sst_dma_new); -void sst_dma_free(struct sst_dma *dma) +static void sst_dma_free(struct sst_dma *dma) { if (dma == NULL) @@ -335,7 +334,6 @@ void sst_dma_free(struct sst_dma *dma) dw_remove(dma->chip); } -EXPORT_SYMBOL(sst_dma_free); /* create new generic firmware object */ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, diff --git a/sound/soc/intel/haswell/sst-haswell-dsp.c b/sound/soc/intel/haswell/sst-haswell-dsp.c index b2bec36d074c..a28220e67cdf 100644 --- a/sound/soc/intel/haswell/sst-haswell-dsp.c +++ b/sound/soc/intel/haswell/sst-haswell-dsp.c @@ -93,29 +93,31 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, struct sst_module_template template; int count, ret; void __iomem *ram; + int type = le16_to_cpu(module->type); + int entry_point = le32_to_cpu(module->entry_point); /* TODO: allowed module types need to be configurable */ - if (module->type != SST_HSW_MODULE_BASE_FW - && module->type != SST_HSW_MODULE_PCM_SYSTEM - && module->type != SST_HSW_MODULE_PCM - && module->type != SST_HSW_MODULE_PCM_REFERENCE - && module->type != SST_HSW_MODULE_PCM_CAPTURE - && module->type != SST_HSW_MODULE_WAVES - && module->type != SST_HSW_MODULE_LPAL) + if (type != SST_HSW_MODULE_BASE_FW && + type != SST_HSW_MODULE_PCM_SYSTEM && + type != SST_HSW_MODULE_PCM && + type != SST_HSW_MODULE_PCM_REFERENCE && + type != SST_HSW_MODULE_PCM_CAPTURE && + type != SST_HSW_MODULE_WAVES && + type != SST_HSW_MODULE_LPAL) return 0; dev_dbg(dsp->dev, "new module sign 0x%s size 0x%x blocks 0x%x type 0x%x\n", module->signature, module->mod_size, - module->blocks, module->type); - dev_dbg(dsp->dev, " entrypoint 0x%x\n", module->entry_point); + module->blocks, type); + dev_dbg(dsp->dev, " entrypoint 0x%x\n", entry_point); dev_dbg(dsp->dev, " persistent 0x%x scratch 0x%x\n", module->info.persistent_size, module->info.scratch_size); memset(&template, 0, sizeof(template)); - template.id = module->type; - template.entry = module->entry_point - 4; - template.persistent_size = module->info.persistent_size; - template.scratch_size = module->info.scratch_size; + template.id = type; + template.entry = entry_point - 4; + template.persistent_size = le32_to_cpu(module->info.persistent_size); + template.scratch_size = le32_to_cpu(module->info.scratch_size); mod = sst_module_new(fw, &template, NULL); if (mod == NULL) @@ -123,26 +125,26 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, block = (void *)module + sizeof(*module); - for (count = 0; count < module->blocks; count++) { + for (count = 0; count < le32_to_cpu(module->blocks); count++) { - if (block->size <= 0) { + if (le32_to_cpu(block->size) <= 0) { dev_err(dsp->dev, "error: block %d size invalid\n", count); sst_module_free(mod); return -EINVAL; } - switch (block->type) { + switch (le32_to_cpu(block->type)) { case SST_HSW_IRAM: ram = dsp->addr.lpe; - mod->offset = - block->ram_offset + dsp->addr.iram_offset; + mod->offset = le32_to_cpu(block->ram_offset) + + dsp->addr.iram_offset; mod->type = SST_MEM_IRAM; break; case SST_HSW_DRAM: case SST_HSW_REGS: ram = dsp->addr.lpe; - mod->offset = block->ram_offset; + mod->offset = le32_to_cpu(block->ram_offset); mod->type = SST_MEM_DRAM; break; default: @@ -152,7 +154,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return -EINVAL; } - mod->size = block->size; + mod->size = le32_to_cpu(block->size); mod->data = (void *)block + sizeof(*block); mod->data_offset = mod->data - fw->dma_buf; @@ -169,7 +171,8 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return ret; } - block = (void *)block + sizeof(*block) + block->size; + block = (void *)block + sizeof(*block) + + le32_to_cpu(block->size); } mod->state = SST_MODULE_STATE_LOADED; @@ -188,7 +191,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) /* verify FW */ if ((strncmp(header->signature, SST_HSW_FW_SIGN, 4) != 0) || - (sst_fw->size != header->file_size + sizeof(*header))) { + (sst_fw->size != + le32_to_cpu(header->file_size) + sizeof(*header))) { dev_err(dsp->dev, "error: invalid fw sign/filesize mismatch\n"); return -EINVAL; } @@ -199,7 +203,7 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) /* parse each module */ module = (void *)sst_fw->dma_buf + sizeof(*header); - for (count = 0; count < header->modules; count++) { + for (count = 0; count < le32_to_cpu(header->modules); count++) { /* module */ ret = hsw_parse_module(dsp, sst_fw, module); @@ -207,7 +211,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) dev_err(dsp->dev, "error: invalid module %d\n", count); return ret; } - module = (void *)module + sizeof(*module) + module->mod_size; + module = (void *)module + sizeof(*module) + + le32_to_cpu(module->mod_size); } return 0; diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index d5f9c30eba32..8bfb8b0fa3d5 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -33,8 +33,7 @@ static int skl_alloc_dma_buf(struct device *dev, struct snd_dma_buffer *dmab, size_t size) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); if (!bus) return -ENODEV; @@ -44,8 +43,7 @@ static int skl_alloc_dma_buf(struct device *dev, static int skl_free_dma_buf(struct device *dev, struct snd_dma_buffer *dmab) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); if (!bus) return -ENODEV; @@ -89,8 +87,7 @@ void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable) static int skl_dsp_setup_spib(struct device *dev, unsigned int size, int stream_tag, int enable) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); struct hdac_stream *stream = snd_hdac_get_stream(bus, SNDRV_PCM_STREAM_PLAYBACK, stream_tag); struct hdac_ext_stream *estream; @@ -100,10 +97,10 @@ static int skl_dsp_setup_spib(struct device *dev, unsigned int size, estream = stream_to_hdac_ext_stream(stream); /* enable/disable SPIB for this hdac stream */ - snd_hdac_ext_stream_spbcap_enable(ebus, enable, stream->index); + snd_hdac_ext_stream_spbcap_enable(bus, enable, stream->index); /* set the spib value */ - snd_hdac_ext_stream_set_spib(ebus, estream, size); + snd_hdac_ext_stream_set_spib(bus, estream, size); return 0; } @@ -111,8 +108,7 @@ static int skl_dsp_setup_spib(struct device *dev, unsigned int size, static int skl_dsp_prepare(struct device *dev, unsigned int format, unsigned int size, struct snd_dma_buffer *dmab) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); struct hdac_ext_stream *estream; struct hdac_stream *stream; struct snd_pcm_substream substream; @@ -124,7 +120,7 @@ static int skl_dsp_prepare(struct device *dev, unsigned int format, memset(&substream, 0, sizeof(substream)); substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - estream = snd_hdac_ext_stream_assign(ebus, &substream, + estream = snd_hdac_ext_stream_assign(bus, &substream, HDAC_EXT_STREAM_TYPE_HOST); if (!estream) return -ENODEV; @@ -143,9 +139,8 @@ static int skl_dsp_prepare(struct device *dev, unsigned int format, static int skl_dsp_trigger(struct device *dev, bool start, int stream_tag) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = dev_get_drvdata(dev); struct hdac_stream *stream; - struct hdac_bus *bus = ebus_to_hbus(ebus); if (!bus) return -ENODEV; @@ -163,10 +158,9 @@ static int skl_dsp_trigger(struct device *dev, bool start, int stream_tag) static int skl_dsp_cleanup(struct device *dev, struct snd_dma_buffer *dmab, int stream_tag) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = dev_get_drvdata(dev); struct hdac_stream *stream; struct hdac_ext_stream *estream; - struct hdac_bus *bus = ebus_to_hbus(ebus); if (!bus) return -ENODEV; @@ -270,8 +264,7 @@ const struct skl_dsp_ops *skl_get_dsp_ops(int pci_id) int skl_init_dsp(struct skl *skl) { void __iomem *mmio_base; - struct hdac_ext_bus *ebus = &skl->ebus; - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct skl_dsp_loader_ops loader_ops; int irq = bus->irq; const struct skl_dsp_ops *ops; @@ -279,8 +272,8 @@ int skl_init_dsp(struct skl *skl) int ret; /* enable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_enable(&skl->ebus, true); - snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, true); + snd_hdac_ext_bus_ppcap_enable(bus, true); + snd_hdac_ext_bus_ppcap_int_enable(bus, true); /* read the BAR of the ADSP MMIO */ mmio_base = pci_ioremap_bar(skl->pci, 4); @@ -335,12 +328,11 @@ unmap_mmio: int skl_free_dsp(struct skl *skl) { - struct hdac_ext_bus *ebus = &skl->ebus; - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct skl_sst *ctx = skl->skl_sst; /* disable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, false); + snd_hdac_ext_bus_ppcap_int_enable(bus, false); ctx->dsp_ops->cleanup(bus->dev, ctx); @@ -383,10 +375,11 @@ int skl_suspend_late_dsp(struct skl *skl) int skl_suspend_dsp(struct skl *skl) { struct skl_sst *ctx = skl->skl_sst; + struct hdac_bus *bus = skl_to_bus(skl); int ret; /* if ppcap is not supported return 0 */ - if (!skl->ebus.bus.ppcap) + if (!bus->ppcap) return 0; ret = skl_dsp_sleep(ctx->dsp); @@ -394,8 +387,8 @@ int skl_suspend_dsp(struct skl *skl) return ret; /* disable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, false); - snd_hdac_ext_bus_ppcap_enable(&skl->ebus, false); + snd_hdac_ext_bus_ppcap_int_enable(bus, false); + snd_hdac_ext_bus_ppcap_enable(bus, false); return 0; } @@ -403,15 +396,16 @@ int skl_suspend_dsp(struct skl *skl) int skl_resume_dsp(struct skl *skl) { struct skl_sst *ctx = skl->skl_sst; + struct hdac_bus *bus = skl_to_bus(skl); int ret; /* if ppcap is not supported return 0 */ - if (!skl->ebus.bus.ppcap) + if (!bus->ppcap) return 0; /* enable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_enable(&skl->ebus, true); - snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, true); + snd_hdac_ext_bus_ppcap_enable(bus, true); + snd_hdac_ext_bus_ppcap_int_enable(bus, true); /* check if DSP 1st boot is done */ if (skl->skl_sst->is_first_boot == true) diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index b9b140275be0..01a050cf8775 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -141,7 +141,7 @@ struct nhlt_specific_cfg { struct nhlt_fmt *fmt; struct nhlt_endpoint *epnt; - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct device *dev = bus->dev; struct nhlt_specific_cfg *sp_config; struct nhlt_acpi_table *nhlt = skl->nhlt; @@ -228,7 +228,7 @@ static void skl_nhlt_trim_space(char *trim) int skl_nhlt_update_topology_bin(struct skl *skl) { struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct device *dev = bus->dev; dev_dbg(dev, "oem_id %.6s, oem_table_id %8s oem_revision %d\n", @@ -248,8 +248,8 @@ static ssize_t skl_nhlt_platform_id_show(struct device *dev, struct device_attribute *attr, char *buf) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); + struct skl *skl = bus_to_skl(bus); struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; char platform_id[32]; diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index afa86b9e4dcf..823e39103edd 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -67,16 +67,15 @@ struct hdac_ext_stream *get_hdac_ext_stream(struct snd_pcm_substream *substream) return substream->runtime->private_data; } -static struct hdac_ext_bus *get_bus_ctx(struct snd_pcm_substream *substream) +static struct hdac_bus *get_bus_ctx(struct snd_pcm_substream *substream) { struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); struct hdac_stream *hstream = hdac_stream(stream); struct hdac_bus *bus = hstream->bus; - - return hbus_to_ebus(bus); + return bus; } -static int skl_substream_alloc_pages(struct hdac_ext_bus *ebus, +static int skl_substream_alloc_pages(struct hdac_bus *bus, struct snd_pcm_substream *substream, size_t size) { @@ -95,7 +94,7 @@ static int skl_substream_free_pages(struct hdac_bus *bus, return snd_pcm_lib_free_pages(substream); } -static void skl_set_pcm_constrains(struct hdac_ext_bus *ebus, +static void skl_set_pcm_constrains(struct hdac_bus *bus, struct snd_pcm_runtime *runtime) { snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -105,9 +104,9 @@ static void skl_set_pcm_constrains(struct hdac_ext_bus *ebus, 20, 178000000); } -static enum hdac_ext_stream_type skl_get_host_stream_type(struct hdac_ext_bus *ebus) +static enum hdac_ext_stream_type skl_get_host_stream_type(struct hdac_bus *bus) { - if ((ebus_to_hbus(ebus))->ppcap) + if (bus->ppcap) return HDAC_EXT_STREAM_TYPE_HOST; else return HDAC_EXT_STREAM_TYPE_COUPLED; @@ -123,9 +122,9 @@ static enum hdac_ext_stream_type skl_get_host_stream_type(struct hdac_ext_bus *e static void skl_set_suspend_active(struct snd_pcm_substream *substream, struct snd_soc_dai *dai, bool enable) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct snd_soc_dapm_widget *w; - struct skl *skl = ebus_to_skl(ebus); + struct skl *skl = bus_to_skl(bus); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) w = dai->playback_widget; @@ -140,8 +139,7 @@ static void skl_set_suspend_active(struct snd_pcm_substream *substream, int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); unsigned int format_val; struct hdac_stream *hstream; struct hdac_ext_stream *stream; @@ -153,7 +151,7 @@ int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params) return -EINVAL; stream = stream_to_hdac_ext_stream(hstream); - snd_hdac_ext_stream_decouple(ebus, stream, true); + snd_hdac_ext_stream_decouple(bus, stream, true); format_val = snd_hdac_calc_stream_format(params->s_freq, params->ch, params->format, params->host_bps, 0); @@ -177,8 +175,7 @@ int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params) int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); unsigned int format_val; struct hdac_stream *hstream; struct hdac_ext_stream *stream; @@ -190,7 +187,7 @@ int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) return -EINVAL; stream = stream_to_hdac_ext_stream(hstream); - snd_hdac_ext_stream_decouple(ebus, stream, true); + snd_hdac_ext_stream_decouple(bus, stream, true); format_val = snd_hdac_calc_stream_format(params->s_freq, params->ch, params->format, params->link_bps, 0); @@ -201,7 +198,7 @@ int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) snd_hdac_ext_link_stream_setup(stream, format_val); - list_for_each_entry(link, &ebus->hlink_list, list) { + list_for_each_entry(link, &bus->hlink_list, list) { if (link->index == params->link_index) snd_hdac_ext_link_set_stream_id(link, hstream->stream_tag); @@ -215,7 +212,7 @@ int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) static int skl_pcm_open(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *stream; struct snd_pcm_runtime *runtime = substream->runtime; struct skl_dma_params *dma_params; @@ -224,12 +221,12 @@ static int skl_pcm_open(struct snd_pcm_substream *substream, dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); - stream = snd_hdac_ext_stream_assign(ebus, substream, - skl_get_host_stream_type(ebus)); + stream = snd_hdac_ext_stream_assign(bus, substream, + skl_get_host_stream_type(bus)); if (stream == NULL) return -EBUSY; - skl_set_pcm_constrains(ebus, runtime); + skl_set_pcm_constrains(bus, runtime); /* * disable WALLCLOCK timestamps for capture streams @@ -301,7 +298,7 @@ static int skl_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct skl_pipe_params p_params = {0}; @@ -309,7 +306,7 @@ static int skl_pcm_hw_params(struct snd_pcm_substream *substream, int ret, dma_id; dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); - ret = skl_substream_alloc_pages(ebus, substream, + ret = skl_substream_alloc_pages(bus, substream, params_buffer_bytes(params)); if (ret < 0) return ret; @@ -343,14 +340,14 @@ static void skl_pcm_close(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct skl_dma_params *dma_params = NULL; - struct skl *skl = ebus_to_skl(ebus); + struct skl *skl = bus_to_skl(bus); struct skl_module_cfg *mconfig; dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); - snd_hdac_ext_stream_release(stream, skl_get_host_stream_type(ebus)); + snd_hdac_ext_stream_release(stream, skl_get_host_stream_type(bus)); dma_params = snd_soc_dai_get_dma_data(dai, substream); /* @@ -380,7 +377,7 @@ static void skl_pcm_close(struct snd_pcm_substream *substream, static int skl_pcm_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); struct skl *skl = get_skl_ctx(dai->dev); struct skl_module_cfg *mconfig; @@ -400,7 +397,7 @@ static int skl_pcm_hw_free(struct snd_pcm_substream *substream, snd_hdac_stream_cleanup(hdac_stream(stream)); hdac_stream(stream)->prepared = 0; - return skl_substream_free_pages(ebus_to_hbus(ebus), substream); + return skl_substream_free_pages(bus, substream); } static int skl_be_hw_params(struct snd_pcm_substream *substream, @@ -420,8 +417,7 @@ static int skl_be_hw_params(struct snd_pcm_substream *substream, static int skl_decoupled_trigger(struct snd_pcm_substream *substream, int cmd) { - struct hdac_ext_bus *ebus = get_bus_ctx(substream); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = get_bus_ctx(substream); struct hdac_ext_stream *stream; int start; unsigned long cookie; @@ -470,7 +466,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct skl *skl = get_skl_ctx(dai->dev); struct skl_sst *ctx = skl->skl_sst; struct skl_module_cfg *mconfig; - struct hdac_ext_bus *ebus = get_bus_ctx(substream); + struct hdac_bus *bus = get_bus_ctx(substream); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); struct snd_soc_dapm_widget *w; int ret; @@ -492,9 +488,9 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, * dpib & lpib position to resume before starting the * DMA */ - snd_hdac_ext_stream_drsm_enable(ebus, true, + snd_hdac_ext_stream_drsm_enable(bus, true, hdac_stream(stream)->index); - snd_hdac_ext_stream_set_dpibr(ebus, stream, + snd_hdac_ext_stream_set_dpibr(bus, stream, stream->lpib); snd_hdac_ext_stream_set_lpib(stream, stream->lpib); } @@ -528,14 +524,14 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, ret = skl_decoupled_trigger(substream, cmd); if ((cmd == SNDRV_PCM_TRIGGER_SUSPEND) && !w->ignore_suspend) { /* save the dpib and lpib positions */ - stream->dpib = readl(ebus->bus.remap_addr + + stream->dpib = readl(bus->remap_addr + AZX_REG_VS_SDXDPIB_XBASE + (AZX_REG_VS_SDXDPIB_XINTERVAL * hdac_stream(stream)->index)); stream->lpib = snd_hdac_stream_get_pos_lpib( hdac_stream(stream)); - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); } break; @@ -546,11 +542,12 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } + static int skl_link_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *link_dev; struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); struct snd_soc_dai *codec_dai = rtd->codec_dai; @@ -558,14 +555,14 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream, struct hdac_ext_link *link; int stream_tag; - link_dev = snd_hdac_ext_stream_assign(ebus, substream, + link_dev = snd_hdac_ext_stream_assign(bus, substream, HDAC_EXT_STREAM_TYPE_LINK); if (!link_dev) return -EBUSY; snd_soc_dai_set_dma_data(dai, substream, (void *)link_dev); - link = snd_hdac_ext_bus_get_link(ebus, codec_dai->component->name); + link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name); if (!link) return -EINVAL; @@ -610,7 +607,7 @@ static int skl_link_pcm_trigger(struct snd_pcm_substream *substream, { struct hdac_ext_stream *link_dev = snd_soc_dai_get_dma_data(dai, substream); - struct hdac_ext_bus *ebus = get_bus_ctx(substream); + struct hdac_bus *bus = get_bus_ctx(substream); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); dev_dbg(dai->dev, "In %s cmd=%d\n", __func__, cmd); @@ -626,7 +623,7 @@ static int skl_link_pcm_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: snd_hdac_ext_link_stream_clear(link_dev); if (cmd == SNDRV_PCM_TRIGGER_SUSPEND) - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); break; default: @@ -638,7 +635,7 @@ static int skl_link_pcm_trigger(struct snd_pcm_substream *substream, static int skl_link_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); struct hdac_ext_stream *link_dev = snd_soc_dai_get_dma_data(dai, substream); @@ -648,7 +645,7 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream, link_dev->link_prepared = 0; - link = snd_hdac_ext_bus_get_link(ebus, rtd->codec_dai->component->name); + link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name); if (!link) return -EINVAL; @@ -1017,10 +1014,11 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, }; -int skl_dai_load(struct snd_soc_component *cmp, - struct snd_soc_dai_driver *pcm_dai) +int skl_dai_load(struct snd_soc_component *cmp, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { - pcm_dai->ops = &skl_pcm_dai_ops; + dai_drv->ops = &skl_pcm_dai_ops; return 0; } @@ -1041,8 +1039,7 @@ static int skl_platform_open(struct snd_pcm_substream *substream) static int skl_coupled_trigger(struct snd_pcm_substream *substream, int cmd) { - struct hdac_ext_bus *ebus = get_bus_ctx(substream); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = get_bus_ctx(substream); struct hdac_ext_stream *stream; struct snd_pcm_substream *s; bool start; @@ -1115,9 +1112,9 @@ static int skl_coupled_trigger(struct snd_pcm_substream *substream, static int skl_platform_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct hdac_ext_bus *ebus = get_bus_ctx(substream); + struct hdac_bus *bus = get_bus_ctx(substream); - if (!(ebus_to_hbus(ebus))->ppcap) + if (!bus->ppcap) return skl_coupled_trigger(substream, cmd); return 0; @@ -1127,7 +1124,7 @@ static snd_pcm_uframes_t skl_platform_pcm_pointer (struct snd_pcm_substream *substream) { struct hdac_ext_stream *hstream = get_hdac_ext_stream(substream); - struct hdac_ext_bus *ebus = get_bus_ctx(substream); + struct hdac_bus *bus = get_bus_ctx(substream); unsigned int pos; /* @@ -1152,12 +1149,12 @@ static snd_pcm_uframes_t skl_platform_pcm_pointer */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - pos = readl(ebus->bus.remap_addr + AZX_REG_VS_SDXDPIB_XBASE + + pos = readl(bus->remap_addr + AZX_REG_VS_SDXDPIB_XBASE + (AZX_REG_VS_SDXDPIB_XINTERVAL * hdac_stream(hstream)->index)); } else { udelay(20); - readl(ebus->bus.remap_addr + + readl(bus->remap_addr + AZX_REG_VS_SDXDPIB_XBASE + (AZX_REG_VS_SDXDPIB_XINTERVAL * hdac_stream(hstream)->index)); @@ -1242,11 +1239,11 @@ static void skl_pcm_free(struct snd_pcm *pcm) static int skl_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; - struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct snd_pcm *pcm = rtd->pcm; unsigned int size; int retval = 0; - struct skl *skl = ebus_to_skl(ebus); + struct skl *skl = bus_to_skl(bus); if (dai->driver->playback.channels_min || dai->driver->capture.channels_min) { @@ -1356,19 +1353,19 @@ static int skl_populate_modules(struct skl *skl) static int skl_platform_soc_probe(struct snd_soc_component *component) { - struct hdac_ext_bus *ebus = dev_get_drvdata(component->dev); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = dev_get_drvdata(component->dev); + struct skl *skl = bus_to_skl(bus); const struct skl_dsp_ops *ops; int ret; pm_runtime_get_sync(component->dev); - if ((ebus_to_hbus(ebus))->ppcap) { + if (bus->ppcap) { skl->component = component; /* init debugfs */ skl->debugfs = skl_debugfs_init(skl); - ret = skl_tplg_init(component, ebus); + ret = skl_tplg_init(component, bus); if (ret < 0) { dev_err(component->dev, "Failed to init topology!\n"); return ret; @@ -1425,10 +1422,10 @@ static const struct snd_soc_component_driver skl_component = { int skl_platform_register(struct device *dev) { int ret; - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct skl *skl = ebus_to_skl(ebus); struct snd_soc_dai_driver *dais; int num_dais = ARRAY_SIZE(skl_platform_dai); + struct hdac_bus *bus = dev_get_drvdata(dev); + struct skl *skl = bus_to_skl(bus); INIT_LIST_HEAD(&skl->ppl_list); INIT_LIST_HEAD(&skl->bind_list); @@ -1464,8 +1461,8 @@ err: int skl_platform_unregister(struct device *dev) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = dev_get_drvdata(dev); + struct skl *skl = bus_to_skl(bus); struct skl_module_deferred_bind *modules, *tmp; if (!list_empty(&skl->bind_list)) { diff --git a/sound/soc/intel/skylake/skl-sst-cldma.c b/sound/soc/intel/skylake/skl-sst-cldma.c index d2b1d60fec02..5bc0d38da7e3 100644 --- a/sound/soc/intel/skylake/skl-sst-cldma.c +++ b/sound/soc/intel/skylake/skl-sst-cldma.c @@ -83,9 +83,9 @@ static void skl_cldma_stream_clear(struct sst_dsp *ctx) /* Code loader helper APIs */ static void skl_cldma_setup_bdle(struct sst_dsp *ctx, struct snd_dma_buffer *dmab_data, - u32 **bdlp, int size, int with_ioc) + __le32 **bdlp, int size, int with_ioc) { - u32 *bdl = *bdlp; + __le32 *bdl = *bdlp; ctx->cl_dev.frags = 0; while (size > 0) { @@ -330,7 +330,7 @@ void skl_cldma_process_intr(struct sst_dsp *ctx) int skl_cldma_prepare(struct sst_dsp *ctx) { int ret; - u32 *bdl; + __le32 *bdl; ctx->cl_dev.bufsize = SKL_MAX_BUFFER_SIZE; @@ -359,7 +359,7 @@ int skl_cldma_prepare(struct sst_dsp *ctx) ctx->dsp_ops.free_dma_buf(ctx->dev, &ctx->cl_dev.dmab_data); return ret; } - bdl = (u32 *)ctx->cl_dev.dmab_bdl.area; + bdl = (__le32 *)ctx->cl_dev.dmab_bdl.area; /* Allocate BDLs */ ctx->cl_dev.ops.cl_setup_bdle(ctx, &ctx->cl_dev.dmab_data, diff --git a/sound/soc/intel/skylake/skl-sst-cldma.h b/sound/soc/intel/skylake/skl-sst-cldma.h index 5b730a1a0ae4..ec736921a083 100644 --- a/sound/soc/intel/skylake/skl-sst-cldma.h +++ b/sound/soc/intel/skylake/skl-sst-cldma.h @@ -203,7 +203,7 @@ struct sst_dsp; struct skl_cl_dev_ops { void (*cl_setup_bdle)(struct sst_dsp *ctx, struct snd_dma_buffer *dmab_data, - u32 **bdlp, int size, int with_ioc); + __le32 **bdlp, int size, int with_ioc); void (*cl_setup_controller)(struct sst_dsp *ctx, struct snd_dma_buffer *dmab_bdl, unsigned int max_size, u32 page_count); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index fcdc716754b6..2620d77729c5 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -108,6 +108,9 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w, case snd_soc_dapm_aif_out: case snd_soc_dapm_dai_out: case snd_soc_dapm_switch: + case snd_soc_dapm_output: + case snd_soc_dapm_mux: + return false; default: return true; @@ -934,7 +937,7 @@ static int skl_tplg_find_moduleid_from_uuid(struct skl *skl, struct soc_bytes_ext *sb = (void *) k->private_value; struct skl_algo_data *bc = (struct skl_algo_data *)sb->dobj.private; struct skl_kpb_params *uuid_params, *params; - struct hdac_bus *bus = ebus_to_hbus(skl_to_ebus(skl)); + struct hdac_bus *bus = skl_to_bus(skl); int i, size, module_id; if (bc->set_params == SKL_PARAM_BIND && bc->max) { @@ -3024,14 +3027,13 @@ void skl_cleanup_resources(struct skl *skl) * information to the driver about module and pipeline parameters which DSP * FW expects like ids, resource values, formats etc */ -static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, +static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, int index, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { int ret; - struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt); - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt); + struct skl *skl = bus_to_skl(bus); struct skl_module_cfg *mconfig; if (!tplg_w->priv.size) @@ -3131,14 +3133,14 @@ static int skl_init_enum_data(struct device *dev, struct soc_enum *se, } static int skl_tplg_control_load(struct snd_soc_component *cmpnt, + int index, struct snd_kcontrol_new *kctl, struct snd_soc_tplg_ctl_hdr *hdr) { struct soc_bytes_ext *sb; struct snd_soc_tplg_bytes_control *tplg_bc; struct snd_soc_tplg_enum_control *tplg_ec; - struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt); struct soc_enum *se; switch (hdr->ops.info) { @@ -3619,12 +3621,11 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, return 0; } -static int skl_manifest_load(struct snd_soc_component *cmpnt, +static int skl_manifest_load(struct snd_soc_component *cmpnt, int index, struct snd_soc_tplg_manifest *manifest) { - struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt); - struct hdac_bus *bus = ebus_to_hbus(ebus); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt); + struct skl *skl = bus_to_skl(bus); /* proceed only if we have private data defined */ if (manifest->priv.size == 0) @@ -3713,12 +3714,11 @@ static void skl_tplg_set_pipe_type(struct skl *skl, struct skl_pipe *pipe) /* * SKL topology init routine */ -int skl_tplg_init(struct snd_soc_component *component, struct hdac_ext_bus *ebus) +int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) { int ret; const struct firmware *fw; - struct hdac_bus *bus = ebus_to_hbus(ebus); - struct skl *skl = ebus_to_skl(ebus); + struct skl *skl = bus_to_skl(bus); struct skl_pipeline *ppl; ret = request_firmware(&fw, skl->tplg_name, bus->dev); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 6d7e0569695f..82282cac9751 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -458,9 +458,9 @@ enum skl_channel { static inline struct skl *get_skl_ctx(struct device *dev) { - struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = dev_get_drvdata(dev); - return ebus_to_skl(ebus); + return bus_to_skl(bus); } int skl_tplg_be_update_params(struct snd_soc_dai *dai, @@ -470,7 +470,7 @@ int skl_dsp_set_dma_control(struct skl_sst *ctx, u32 *caps, void skl_tplg_set_be_dmic_config(struct snd_soc_dai *dai, struct skl_pipe_params *params, int stream); int skl_tplg_init(struct snd_soc_component *component, - struct hdac_ext_bus *ebus); + struct hdac_bus *ebus); struct skl_module_cfg *skl_tplg_fe_get_cpr_module( struct snd_soc_dai *dai, int stream); int skl_tplg_update_pipe_params(struct device *dev, @@ -512,8 +512,9 @@ int skl_pcm_host_dma_prepare(struct device *dev, int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params); -int skl_dai_load(struct snd_soc_component *cmp, - struct snd_soc_dai_driver *pcm_dai); +int skl_dai_load(struct snd_soc_component *cmp, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai); void skl_tplg_add_moduleid_in_bind_params(struct skl *skl, struct snd_soc_dapm_widget *w); #endif diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index f0d9793f872a..dce649485649 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -29,6 +29,7 @@ #include <linux/delay.h> #include <sound/pcm.h> #include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> #include <sound/hda_register.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> @@ -36,8 +37,6 @@ #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" -static struct skl_machine_pdata skl_dmic_data; - /* * initialize the PCI registers */ @@ -54,7 +53,7 @@ static void skl_update_pci_byte(struct pci_dev *pci, unsigned int reg, static void skl_init_pci(struct skl *skl) { - struct hdac_ext_bus *ebus = &skl->ebus; + struct hdac_bus *bus = skl_to_bus(skl); /* * Clear bits 0-2 of PCI register TCSEL (at offset 0x44) @@ -63,7 +62,7 @@ static void skl_init_pci(struct skl *skl) * codecs. * The PCI register TCSEL is defined in the Intel manuals. */ - dev_dbg(ebus_to_hbus(ebus)->dev, "Clearing TCSEL\n"); + dev_dbg(bus->dev, "Clearing TCSEL\n"); skl_update_pci_byte(skl->pci, AZX_PCIREG_TCSEL, 0x07, 0); } @@ -103,8 +102,7 @@ static void skl_enable_miscbdcge(struct device *dev, bool enable) static void skl_clock_power_gating(struct device *dev, bool enable) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); u32 val; /* Update PDCGE bit of CGCTL register */ @@ -127,7 +125,6 @@ static void skl_clock_power_gating(struct device *dev, bool enable) */ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) { - struct hdac_ext_bus *ebus = hbus_to_ebus(bus); struct hdac_ext_link *hlink; int ret; @@ -135,7 +132,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) ret = snd_hdac_bus_init_chip(bus, full_reset); /* Reset stream-to-link mapping */ - list_for_each_entry(hlink, &ebus->hlink_list, list) + list_for_each_entry(hlink, &bus->hlink_list, list) bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); skl_enable_miscbdcge(bus->dev, true); @@ -146,8 +143,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) void skl_update_d0i3c(struct device *dev, bool enable) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); u8 reg; int timeout = 50; @@ -197,8 +193,7 @@ static void skl_stream_update(struct hdac_bus *bus, struct hdac_stream *hstr) static irqreturn_t skl_interrupt(int irq, void *dev_id) { - struct hdac_ext_bus *ebus = dev_id; - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_id; u32 status; if (!pm_runtime_active(bus->dev)) @@ -227,8 +222,7 @@ static irqreturn_t skl_interrupt(int irq, void *dev_id) static irqreturn_t skl_threaded_handler(int irq, void *dev_id) { - struct hdac_ext_bus *ebus = dev_id; - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = dev_id; u32 status; status = snd_hdac_chip_readl(bus, INTSTS); @@ -238,16 +232,15 @@ static irqreturn_t skl_threaded_handler(int irq, void *dev_id) return IRQ_HANDLED; } -static int skl_acquire_irq(struct hdac_ext_bus *ebus, int do_disconnect) +static int skl_acquire_irq(struct hdac_bus *bus, int do_disconnect) { - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = bus_to_skl(bus); int ret; ret = request_threaded_irq(skl->pci->irq, skl_interrupt, skl_threaded_handler, IRQF_SHARED, - KBUILD_MODNAME, ebus); + KBUILD_MODNAME, bus); if (ret) { dev_err(bus->dev, "unable to grab IRQ %d, disabling device\n", @@ -264,21 +257,20 @@ static int skl_acquire_irq(struct hdac_ext_bus *ebus, int do_disconnect) static int skl_suspend_late(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); + struct skl *skl = bus_to_skl(bus); return skl_suspend_late_dsp(skl); } #ifdef CONFIG_PM -static int _skl_suspend(struct hdac_ext_bus *ebus) +static int _skl_suspend(struct hdac_bus *bus) { - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = bus_to_skl(bus); struct pci_dev *pci = to_pci_dev(bus->dev); int ret; - snd_hdac_ext_bus_link_power_down_all(ebus); + snd_hdac_ext_bus_link_power_down_all(bus); ret = skl_suspend_dsp(skl); if (ret < 0) @@ -295,10 +287,9 @@ static int _skl_suspend(struct hdac_ext_bus *ebus) return 0; } -static int _skl_resume(struct hdac_ext_bus *ebus) +static int _skl_resume(struct hdac_bus *bus) { - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = bus_to_skl(bus); skl_init_pci(skl); skl_init_chip(bus, true); @@ -314,9 +305,8 @@ static int _skl_resume(struct hdac_ext_bus *ebus) static int skl_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); + struct skl *skl = bus_to_skl(bus); int ret = 0; /* @@ -325,15 +315,15 @@ static int skl_suspend(struct device *dev) */ if (skl->supend_active) { /* turn off the links and stop the CORB/RIRB DMA if it is On */ - snd_hdac_ext_bus_link_power_down_all(ebus); + snd_hdac_ext_bus_link_power_down_all(bus); - if (ebus->cmd_dma_state) - snd_hdac_bus_stop_cmd_io(&ebus->bus); + if (bus->cmd_dma_state) + snd_hdac_bus_stop_cmd_io(bus); enable_irq_wake(bus->irq); pci_save_state(pci); } else { - ret = _skl_suspend(ebus); + ret = _skl_suspend(bus); if (ret < 0) return ret; skl->skl_sst->fw_loaded = false; @@ -352,9 +342,8 @@ static int skl_suspend(struct device *dev) static int skl_resume(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); + struct skl *skl = bus_to_skl(bus); struct hdac_ext_link *hlink = NULL; int ret; @@ -374,32 +363,32 @@ static int skl_resume(struct device *dev) */ if (skl->supend_active) { pci_restore_state(pci); - snd_hdac_ext_bus_link_power_up_all(ebus); + snd_hdac_ext_bus_link_power_up_all(bus); disable_irq_wake(bus->irq); /* * turn On the links which are On before active suspend * and start the CORB/RIRB DMA if On before * active suspend. */ - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { if (hlink->ref_count) snd_hdac_ext_bus_link_power_up(hlink); } - if (ebus->cmd_dma_state) - snd_hdac_bus_init_cmd_io(&ebus->bus); ret = 0; + if (bus->cmd_dma_state) + snd_hdac_bus_init_cmd_io(bus); } else { - ret = _skl_resume(ebus); + ret = _skl_resume(bus); /* turn off the links which are off before suspend */ - list_for_each_entry(hlink, &ebus->hlink_list, list) { + list_for_each_entry(hlink, &bus->hlink_list, list) { if (!hlink->ref_count) snd_hdac_ext_bus_link_power_down(hlink); } - if (!ebus->cmd_dma_state) - snd_hdac_bus_stop_cmd_io(&ebus->bus); + if (!bus->cmd_dma_state) + snd_hdac_bus_stop_cmd_io(bus); } return ret; @@ -410,23 +399,21 @@ static int skl_resume(struct device *dev) static int skl_runtime_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); dev_dbg(bus->dev, "in %s\n", __func__); - return _skl_suspend(ebus); + return _skl_suspend(bus); } static int skl_runtime_resume(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); dev_dbg(bus->dev, "in %s\n", __func__); - return _skl_resume(ebus); + return _skl_resume(bus); } #endif /* CONFIG_PM */ @@ -439,20 +426,19 @@ static const struct dev_pm_ops skl_pm = { /* * destructor */ -static int skl_free(struct hdac_ext_bus *ebus) +static int skl_free(struct hdac_bus *bus) { - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = bus_to_skl(bus); skl->init_done = 0; /* to be sure */ - snd_hdac_ext_stop_streams(ebus); + snd_hdac_ext_stop_streams(bus); if (bus->irq >= 0) - free_irq(bus->irq, (void *)ebus); + free_irq(bus->irq, (void *)bus); snd_hdac_bus_free_stream_pages(bus); - snd_hdac_stream_free_all(ebus); - snd_hdac_link_free_all(ebus); + snd_hdac_stream_free_all(bus); + snd_hdac_link_free_all(bus); if (bus->remap_addr) iounmap(bus->remap_addr); @@ -460,11 +446,11 @@ static int skl_free(struct hdac_ext_bus *ebus) pci_release_regions(skl->pci); pci_disable_device(skl->pci); - snd_hdac_ext_bus_exit(ebus); + snd_hdac_ext_bus_exit(bus); cancel_work_sync(&skl->probe_work); if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_i915_exit(&ebus->bus); + snd_hdac_i915_exit(bus); return 0; } @@ -488,8 +474,8 @@ static struct skl_ssp_clk skl_ssp_clks[] = { static int skl_find_machine(struct skl *skl, void *driver_data) { + struct hdac_bus *bus = skl_to_bus(skl); struct snd_soc_acpi_mach *mach = driver_data; - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); struct skl_machine_pdata *pdata; mach = snd_soc_acpi_find_machine(mach); @@ -500,17 +486,19 @@ static int skl_find_machine(struct skl *skl, void *driver_data) skl->mach = mach; skl->fw_name = mach->fw_filename; - pdata = skl->mach->pdata; + pdata = mach->pdata; - if (mach->pdata) + if (pdata) { skl->use_tplg_pcm = pdata->use_tplg_pcm; + pdata->dmic_num = skl_get_dmic_geo(skl); + } return 0; } static int skl_machine_device_register(struct skl *skl) { - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct snd_soc_acpi_mach *mach = skl->mach; struct platform_device *pdev; int ret; @@ -544,7 +532,7 @@ static void skl_machine_device_unregister(struct skl *skl) static int skl_dmic_device_register(struct skl *skl) { - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct platform_device *pdev; int ret; @@ -643,12 +631,13 @@ static void skl_clock_device_unregister(struct skl *skl) /* * Probe the given codec address */ -static int probe_codec(struct hdac_ext_bus *ebus, int addr) +static int probe_codec(struct hdac_bus *bus, int addr) { - struct hdac_bus *bus = ebus_to_hbus(ebus); unsigned int cmd = (addr << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; unsigned int res = -1; + struct skl *skl = bus_to_skl(bus); + struct hdac_device *hdev; mutex_lock(&bus->cmd_mutex); snd_hdac_bus_send_cmd(bus, cmd); @@ -658,13 +647,16 @@ static int probe_codec(struct hdac_ext_bus *ebus, int addr) return -EIO; dev_dbg(bus->dev, "codec #%d probed OK\n", addr); - return snd_hdac_ext_bus_device_init(ebus, addr); + hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL); + if (!hdev) + return -ENOMEM; + + return snd_hdac_ext_bus_device_init(bus, addr, hdev); } /* Codec initialization */ -static void skl_codec_create(struct hdac_ext_bus *ebus) +static void skl_codec_create(struct hdac_bus *bus) { - struct hdac_bus *bus = ebus_to_hbus(ebus); int c, max_slots; max_slots = HDA_MAX_CODECS; @@ -672,7 +664,7 @@ static void skl_codec_create(struct hdac_ext_bus *ebus) /* First try to probe all given codec slots */ for (c = 0; c < max_slots; c++) { if ((bus->codec_mask & (1 << c))) { - if (probe_codec(ebus, c) < 0) { + if (probe_codec(bus, c) < 0) { /* * Some BIOSen give you wrong codec addresses * that don't exist @@ -722,8 +714,7 @@ static int skl_i915_init(struct hdac_bus *bus) static void skl_probe_work(struct work_struct *work) { struct skl *skl = container_of(work, struct skl, probe_work); - struct hdac_ext_bus *ebus = &skl->ebus; - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = skl_to_bus(skl); struct hdac_ext_link *hlink = NULL; int err; @@ -744,7 +735,7 @@ static void skl_probe_work(struct work_struct *work) dev_info(bus->dev, "no hda codecs found!\n"); /* create codec instances */ - skl_codec_create(ebus); + skl_codec_create(bus); /* register platform dai and controls */ err = skl_platform_register(bus->dev); @@ -773,8 +764,8 @@ static void skl_probe_work(struct work_struct *work) /* * we are done probing so decrement link counts */ - list_for_each_entry(hlink, &ebus->hlink_list, list) - snd_hdac_ext_bus_link_put(ebus, hlink); + list_for_each_entry(hlink, &bus->hlink_list, list) + snd_hdac_ext_bus_link_put(bus, hlink); /* configure PM */ pm_runtime_put_noidle(bus->dev); @@ -796,7 +787,7 @@ static int skl_create(struct pci_dev *pci, struct skl **rskl) { struct skl *skl; - struct hdac_ext_bus *ebus; + struct hdac_bus *bus; int err; @@ -811,23 +802,22 @@ static int skl_create(struct pci_dev *pci, pci_disable_device(pci); return -ENOMEM; } - ebus = &skl->ebus; - snd_hdac_ext_bus_init(ebus, &pci->dev, &bus_core_ops, io_ops); - ebus->bus.use_posbuf = 1; + + bus = skl_to_bus(skl); + snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, NULL); + bus->use_posbuf = 1; skl->pci = pci; INIT_WORK(&skl->probe_work, skl_probe_work); - - ebus->bus.bdl_pos_adj = 0; + bus->bdl_pos_adj = 0; *rskl = skl; return 0; } -static int skl_first_init(struct hdac_ext_bus *ebus) +static int skl_first_init(struct hdac_bus *bus) { - struct skl *skl = ebus_to_skl(ebus); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *skl = bus_to_skl(bus); struct pci_dev *pci = skl->pci; int err; unsigned short gcap; @@ -848,7 +838,7 @@ static int skl_first_init(struct hdac_ext_bus *ebus) snd_hdac_bus_parse_capabilities(bus); - if (skl_acquire_irq(ebus, 0) < 0) + if (skl_acquire_irq(bus, 0) < 0) return -EBUSY; pci_set_master(pci); @@ -872,14 +862,14 @@ static int skl_first_init(struct hdac_ext_bus *ebus) if (!pb_streams && !cp_streams) return -EIO; - ebus->num_streams = cp_streams + pb_streams; + bus->num_streams = cp_streams + pb_streams; /* initialize streams */ snd_hdac_ext_stream_init_all - (ebus, 0, cp_streams, SNDRV_PCM_STREAM_CAPTURE); + (bus, 0, cp_streams, SNDRV_PCM_STREAM_CAPTURE); start_idx = cp_streams; snd_hdac_ext_stream_init_all - (ebus, start_idx, pb_streams, SNDRV_PCM_STREAM_PLAYBACK); + (bus, start_idx, pb_streams, SNDRV_PCM_STREAM_PLAYBACK); err = snd_hdac_bus_alloc_stream_pages(bus); if (err < 0) @@ -895,7 +885,6 @@ static int skl_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { struct skl *skl; - struct hdac_ext_bus *ebus = NULL; struct hdac_bus *bus = NULL; int err; @@ -904,10 +893,9 @@ static int skl_probe(struct pci_dev *pci, if (err < 0) return err; - ebus = &skl->ebus; - bus = ebus_to_hbus(ebus); + bus = skl_to_bus(skl); - err = skl_first_init(ebus); + err = skl_first_init(bus); if (err < 0) goto out_free; @@ -928,9 +916,7 @@ static int skl_probe(struct pci_dev *pci, skl_nhlt_update_topology_bin(skl); - pci_set_drvdata(skl->pci, ebus); - - skl_dmic_data.dmic_num = skl_get_dmic_geo(skl); + pci_set_drvdata(skl->pci, bus); /* check if dsp is there */ if (bus->ppcap) { @@ -952,7 +938,7 @@ static int skl_probe(struct pci_dev *pci, skl->skl_sst->clock_power_gating = skl_clock_power_gating; } if (bus->mlcap) - snd_hdac_ext_bus_get_ml_capabilities(ebus); + snd_hdac_ext_bus_get_ml_capabilities(bus); snd_hdac_bus_stop_chip(bus); @@ -972,31 +958,30 @@ out_clk_free: out_nhlt_free: skl_nhlt_free(skl->nhlt); out_free: - skl_free(ebus); + skl_free(bus); return err; } static void skl_shutdown(struct pci_dev *pci) { - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); struct hdac_stream *s; struct hdac_ext_stream *stream; struct skl *skl; - if (ebus == NULL) + if (!bus) return; - skl = ebus_to_skl(ebus); + skl = bus_to_skl(bus); if (!skl->init_done) return; - snd_hdac_ext_stop_streams(ebus); + snd_hdac_ext_stop_streams(bus); list_for_each_entry(s, &bus->stream_list, list) { stream = stream_to_hdac_ext_stream(s); - snd_hdac_ext_stream_decouple(ebus, stream, false); + snd_hdac_ext_stream_decouple(bus, stream, false); } snd_hdac_bus_stop_chip(bus); @@ -1004,15 +989,15 @@ static void skl_shutdown(struct pci_dev *pci) static void skl_remove(struct pci_dev *pci) { - struct hdac_ext_bus *ebus = pci_get_drvdata(pci); - struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = pci_get_drvdata(pci); + struct skl *skl = bus_to_skl(bus); release_firmware(skl->tplg); pm_runtime_get_noresume(&pci->dev); /* codec removal, invoke bus_device_remove */ - snd_hdac_ext_bus_device_remove(ebus); + snd_hdac_ext_bus_device_remove(bus); skl->debugfs = NULL; skl_platform_unregister(&pci->dev); @@ -1022,176 +1007,27 @@ static void skl_remove(struct pci_dev *pci) skl_clock_device_unregister(skl); skl_nhlt_remove_sysfs(skl); skl_nhlt_free(skl->nhlt); - skl_free(ebus); + skl_free(bus); dev_set_drvdata(&pci->dev, NULL); } -static struct snd_soc_acpi_codecs skl_codecs = { - .num_codecs = 1, - .codecs = {"10508825"} -}; - -static struct snd_soc_acpi_codecs kbl_codecs = { - .num_codecs = 1, - .codecs = {"10508825"} -}; - -static struct snd_soc_acpi_codecs bxt_codecs = { - .num_codecs = 1, - .codecs = {"MX98357A"} -}; - -static struct snd_soc_acpi_codecs kbl_poppy_codecs = { - .num_codecs = 1, - .codecs = {"10EC5663"} -}; - -static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = { - .num_codecs = 2, - .codecs = {"10EC5663", "10EC5514"} -}; - -static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = { - .num_codecs = 1, - .codecs = {"MX98357A"} -}; - -static struct skl_machine_pdata cnl_pdata = { - .use_tplg_pcm = true, -}; - -static struct snd_soc_acpi_mach sst_skl_devdata[] = { - { - .id = "INT343A", - .drv_name = "skl_alc286s_i2s", - .fw_filename = "intel/dsp_fw_release.bin", - }, - { - .id = "INT343B", - .drv_name = "skl_n88l25_s4567", - .fw_filename = "intel/dsp_fw_release.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &skl_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98357A", - .drv_name = "skl_n88l25_m98357a", - .fw_filename = "intel/dsp_fw_release.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &skl_codecs, - .pdata = &skl_dmic_data - }, - {} -}; - -static struct snd_soc_acpi_mach sst_bxtp_devdata[] = { - { - .id = "INT343A", - .drv_name = "bxt_alc298s_i2s", - .fw_filename = "intel/dsp_fw_bxtn.bin", - }, - { - .id = "DLGS7219", - .drv_name = "bxt_da7219_max98357a_i2s", - .fw_filename = "intel/dsp_fw_bxtn.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &bxt_codecs, - }, - {} -}; - -static struct snd_soc_acpi_mach sst_kbl_devdata[] = { - { - .id = "INT343A", - .drv_name = "kbl_alc286s_i2s", - .fw_filename = "intel/dsp_fw_kbl.bin", - }, - { - .id = "INT343B", - .drv_name = "kbl_n88l25_s4567", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98357A", - .drv_name = "kbl_n88l25_m98357a", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98927", - .drv_name = "kbl_r5514_5663_max", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_5663_5514_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98927", - .drv_name = "kbl_rt5663_m98927", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_poppy_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "10EC5663", - .drv_name = "kbl_rt5663", - .fw_filename = "intel/dsp_fw_kbl.bin", - }, - { - .id = "DLGS7219", - .drv_name = "kbl_da7219_max98357a", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_7219_98357_codecs, - .pdata = &skl_dmic_data - }, - - {} -}; - -static struct snd_soc_acpi_mach sst_glk_devdata[] = { - { - .id = "INT343A", - .drv_name = "glk_alc298s_i2s", - .fw_filename = "intel/dsp_fw_glk.bin", - }, - {} -}; - -static const struct snd_soc_acpi_mach sst_cnl_devdata[] = { - { - .id = "INT34C2", - .drv_name = "cnl_rt274", - .fw_filename = "intel/dsp_fw_cnl.bin", - .pdata = &cnl_pdata, - }, - {} -}; - /* PCI IDs */ static const struct pci_device_id skl_ids[] = { /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), - .driver_data = (unsigned long)&sst_skl_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_skl_machines}, /* BXT-P */ { PCI_DEVICE(0x8086, 0x5a98), - .driver_data = (unsigned long)&sst_bxtp_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_bxt_machines}, /* KBL */ { PCI_DEVICE(0x8086, 0x9D71), - .driver_data = (unsigned long)&sst_kbl_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_kbl_machines}, /* GLK */ { PCI_DEVICE(0x8086, 0x3198), - .driver_data = (unsigned long)&sst_glk_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_glk_machines}, /* CNL */ { PCI_DEVICE(0x8086, 0x9dc8), - .driver_data = (unsigned long)&sst_cnl_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_cnl_machines}, { 0, } }; MODULE_DEVICE_TABLE(pci, skl_ids); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 0d5375cbcf6e..78aa8bdcb619 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -71,7 +71,7 @@ struct skl_fw_config { }; struct skl { - struct hdac_ext_bus ebus; + struct hdac_bus hbus; struct pci_dev *pci; unsigned int init_done:1; /* delayed init status */ @@ -105,9 +105,8 @@ struct skl { struct snd_soc_acpi_mach *mach; }; -#define skl_to_ebus(s) (&(s)->ebus) -#define ebus_to_skl(sbus) \ - container_of(sbus, struct skl, sbus) +#define skl_to_bus(s) (&(s)->hbus) +#define bus_to_skl(bus) container_of(bus, struct skl, hbus) /* to pass dai dma data */ struct skl_dma_params { diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 51ec4ff6ed95..697aa50aff9a 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -15,20 +15,12 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) { - struct snd_soc_dai_driver *sub_dai_drivers; + struct mtk_base_afe_dai *dai; size_t num_dai_drivers = 0, dai_idx = 0; - int i; - - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } /* calcualte total dai driver size */ - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) - num_dai_drivers += afe->sub_dais[i].num_dai_drivers; + list_for_each_entry(dai, &afe->sub_dais, list) { + num_dai_drivers += dai->num_dai_drivers; } dev_info(afe->dev, "%s(), num of dai %zd\n", __func__, num_dai_drivers); @@ -42,19 +34,14 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) if (!afe->dai_drivers) return -ENOMEM; - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) { - sub_dai_drivers = afe->sub_dais[i].dai_drivers; - /* dai driver */ - memcpy(&afe->dai_drivers[dai_idx], - sub_dai_drivers, - afe->sub_dais[i].num_dai_drivers * - sizeof(struct snd_soc_dai_driver)); - dai_idx += afe->sub_dais[i].num_dai_drivers; - } + list_for_each_entry(dai, &afe->sub_dais, list) { + /* dai driver */ + memcpy(&afe->dai_drivers[dai_idx], + dai->dai_drivers, + dai->num_dai_drivers * + sizeof(struct snd_soc_dai_driver)); + dai_idx += dai->num_dai_drivers; } - return 0; } EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai); @@ -62,28 +49,25 @@ EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai); int mtk_afe_add_sub_dai_control(struct snd_soc_component *component) { struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - int i; + struct mtk_base_afe_dai *dai; - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } - - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].controls) + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->controls) snd_soc_add_component_controls(component, - afe->sub_dais[i].controls, - afe->sub_dais[i].num_controls); + dai->controls, + dai->num_controls); - if (afe->sub_dais[i].dapm_widgets) + if (dai->dapm_widgets) snd_soc_dapm_new_controls(&component->dapm, - afe->sub_dais[i].dapm_widgets, - afe->sub_dais[i].num_dapm_widgets); - - if (afe->sub_dais[i].dapm_routes) + dai->dapm_widgets, + dai->num_dapm_widgets); + } + /* add routes after all widgets are added */ + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->dapm_routes) snd_soc_dapm_add_routes(&component->dapm, - afe->sub_dais[i].dapm_routes, - afe->sub_dais[i].num_dapm_routes); + dai->dapm_routes, + dai->num_dapm_routes); } snd_soc_dapm_new_widgets(component->dapm.card); diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h index bcf562f029b6..bd8d5e0c6843 100644 --- a/sound/soc/mediatek/common/mtk-base-afe.h +++ b/sound/soc/mediatek/common/mtk-base-afe.h @@ -46,6 +46,7 @@ struct mtk_base_irq_data { }; struct device; +struct list_head; struct mtk_base_afe_memif; struct mtk_base_afe_irq; struct mtk_base_afe_dai; @@ -72,8 +73,7 @@ struct mtk_base_afe { struct mtk_base_afe_irq *irqs; int irqs_size; - struct mtk_base_afe_dai *sub_dais; - int num_sub_dais; + struct list_head sub_dais; struct snd_soc_dai_driver *dai_drivers; unsigned int num_dai_drivers; @@ -110,6 +110,8 @@ struct mtk_base_afe_dai { unsigned int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; unsigned int num_dapm_routes; + + struct list_head list; }; #endif diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-common.h b/sound/soc/mediatek/mt6797/mt6797-afe-common.h index 22eb7b455cf1..4eac9977b2b0 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-common.h +++ b/sound/soc/mediatek/mt6797/mt6797-afe-common.h @@ -10,6 +10,7 @@ #define _MT_6797_AFE_COMMON_H_ #include <sound/soc.h> +#include <linux/list.h> #include <linux/regmap.h> #include "../common/mtk-base-afe.h" diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 6c5dd9fc9976..192f4d7b37b6 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -733,6 +733,34 @@ static const struct snd_soc_component_driver mt6797_afe_component = { .probe = mt6797_afe_component_probe, }; +static int mt6797_dai_memif_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mt6797_memif_dai_driver; + dai->num_dai_drivers = ARRAY_SIZE(mt6797_memif_dai_driver); + + dai->dapm_widgets = mt6797_memif_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mt6797_memif_widgets); + dai->dapm_routes = mt6797_memif_routes; + dai->num_dapm_routes = ARRAY_SIZE(mt6797_memif_routes); + return 0; +} + +typedef int (*dai_register_cb)(struct mtk_base_afe *); +static const dai_register_cb dai_register_cbs[] = { + mt6797_dai_adda_register, + mt6797_dai_pcm_register, + mt6797_dai_hostless_register, + mt6797_dai_memif_register, +}; + static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) { struct mtk_base_afe *afe; @@ -811,29 +839,24 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) } /* init sub_dais */ - afe->num_sub_dais = MT6797_DAI_NUM; - afe->sub_dais = devm_kcalloc(dev, afe->num_sub_dais, - sizeof(*afe->sub_dais), - GFP_KERNEL); - if (!afe->sub_dais) - return -ENOMEM; - - mt6797_dai_adda_register(afe); - mt6797_dai_pcm_register(afe); - mt6797_dai_hostless_register(afe); - - afe->sub_dais[MT6797_MEMIF_DL1].dai_drivers = mt6797_memif_dai_driver; - afe->sub_dais[MT6797_MEMIF_DL1].num_dai_drivers = - ARRAY_SIZE(mt6797_memif_dai_driver); - afe->sub_dais[MT6797_MEMIF_DL1].dapm_widgets = mt6797_memif_widgets; - afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_widgets = - ARRAY_SIZE(mt6797_memif_widgets); - afe->sub_dais[MT6797_MEMIF_DL1].dapm_routes = mt6797_memif_routes; - afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_routes = - ARRAY_SIZE(mt6797_memif_routes); + INIT_LIST_HEAD(&afe->sub_dais); + + for (i = 0; i < ARRAY_SIZE(dai_register_cbs); i++) { + ret = dai_register_cbs[i](afe); + if (ret) { + dev_warn(afe->dev, "dai register i %d fail, ret %d\n", + i, ret); + return ret; + } + } /* init dai_driver and component_driver */ - mtk_afe_combine_sub_dai(afe); + ret = mtk_afe_combine_sub_dai(afe); + if (ret) { + dev_warn(afe->dev, "mtk_afe_combine_sub_dai fail, ret %d\n", + ret); + return ret; + } afe->mtk_afe_hardware = &mt6797_afe_hardware; afe->memif_fs = mt6797_memif_fs; diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c index ad083265ce94..0ac6409c6d61 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c @@ -383,14 +383,20 @@ static struct snd_soc_dai_driver mtk_dai_adda_driver[] = { int mt6797_dai_adda_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_ADDA; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_adda_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_widgets = mtk_dai_adda_widgets; - afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets); - afe->sub_dais[id].dapm_routes = mtk_dai_adda_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes); + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_adda_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver); + + dai->dapm_widgets = mtk_dai_adda_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets); + dai->dapm_routes = mtk_dai_adda_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes); return 0; } diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c index 4cf985b15a11..ed23e6a53b08 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c @@ -100,13 +100,19 @@ static struct snd_soc_dai_driver mtk_dai_hostless_driver[] = { int mt6797_dai_hostless_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_HOSTLESS_LPBK; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_hostless_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_routes = mtk_dai_hostless_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes); + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_hostless_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver); + + dai->dapm_routes = mtk_dai_hostless_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes); return 0; } diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c index 16d5b5067204..3136f0bc7827 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c @@ -298,15 +298,20 @@ static struct snd_soc_dai_driver mtk_dai_pcm_driver[] = { int mt6797_dai_pcm_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_PCM_1; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_pcm_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_widgets = mtk_dai_pcm_widgets; - afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets); - afe->sub_dais[id].dapm_routes = mtk_dai_pcm_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes); + list_add(&dai->list, &afe->sub_dais); + dai->dai_drivers = mtk_dai_pcm_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver); + + dai->dapm_widgets = mtk_dai_pcm_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets); + dai->dapm_routes = mtk_dai_pcm_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes); return 0; } diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig new file mode 100644 index 000000000000..8af8bc358a90 --- /dev/null +++ b/sound/soc/meson/Kconfig @@ -0,0 +1,65 @@ +menu "ASoC support for Amlogic platforms" + depends on ARCH_MESON || COMPILE_TEST + +config SND_MESON_AXG_FIFO + tristate + select REGMAP_MMIO + +config SND_MESON_AXG_FRDDR + tristate "Amlogic AXG Playback FIFO support" + select SND_MESON_AXG_FIFO + help + Select Y or M to add support for the frontend playback interfaces + embedded in the Amlogic AXG SoC family + +config SND_MESON_AXG_TODDR + tristate "Amlogic AXG Capture FIFO support" + select SND_MESON_AXG_FIFO + help + Select Y or M to add support for the frontend capture interfaces + embedded in the Amlogic AXG SoC family + +config SND_MESON_AXG_TDM_FORMATTER + tristate + select REGMAP_MMIO + +config SND_MESON_AXG_TDM_INTERFACE + tristate + select SND_MESON_AXG_TDM_FORMATTER + +config SND_MESON_AXG_TDMIN + tristate "Amlogic AXG TDM Input Support" + select SND_MESON_AXG_TDM_FORMATTER + select SND_MESON_AXG_TDM_INTERFACE + help + Select Y or M to add support for TDM input formatter embedded + in the Amlogic AXG SoC family + +config SND_MESON_AXG_TDMOUT + tristate "Amlogic AXG TDM Output Support" + select SND_MESON_AXG_TDM_FORMATTER + select SND_MESON_AXG_TDM_INTERFACE + help + Select Y or M to add support for TDM output formatter embedded + in the Amlogic AXG SoC family + +config SND_MESON_AXG_SOUND_CARD + tristate "Amlogic AXG Sound Card Support" + select SND_MESON_AXG_TDM_INTERFACE + imply SND_MESON_AXG_FRDDR + imply SND_MESON_AXG_TODDR + imply SND_MESON_AXG_TDMIN + imply SND_MESON_AXG_TDMOUT + imply SND_MESON_AXG_SPDIFOUT + help + Select Y or M to add support for the AXG SoC sound card + +config SND_MESON_AXG_SPDIFOUT + tristate "Amlogic AXG SPDIF Output Support" + select SND_PCM_IEC958 + imply SND_SOC_SPDIF + help + Select Y or M to add support for SPDIF output serializer embedded + in the Amlogic AXG SoC family + +endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile new file mode 100644 index 000000000000..c5e003b093db --- /dev/null +++ b/sound/soc/meson/Makefile @@ -0,0 +1,21 @@ +# SPDX-License-Identifier: (GPL-2.0 OR MIT) + +snd-soc-meson-axg-fifo-objs := axg-fifo.o +snd-soc-meson-axg-frddr-objs := axg-frddr.o +snd-soc-meson-axg-toddr-objs := axg-toddr.o +snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o +snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o +snd-soc-meson-axg-tdmin-objs := axg-tdmin.o +snd-soc-meson-axg-tdmout-objs := axg-tdmout.o +snd-soc-meson-axg-sound-card-objs := axg-card.o +snd-soc-meson-axg-spdifout-objs := axg-spdifout.o + +obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o +obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o +obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o +obj-$(CONFIG_SND_MESON_AXG_TDM_FORMATTER) += snd-soc-meson-axg-tdm-formatter.o +obj-$(CONFIG_SND_MESON_AXG_TDM_INTERFACE) += snd-soc-meson-axg-tdm-interface.o +obj-$(CONFIG_SND_MESON_AXG_TDMIN) += snd-soc-meson-axg-tdmin.o +obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o +obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o +obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c new file mode 100644 index 000000000000..2914ba0d965b --- /dev/null +++ b/sound/soc/meson/axg-card.c @@ -0,0 +1,671 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "axg-tdm.h" + +struct axg_card { + struct snd_soc_card card; + void **link_data; +}; + +struct axg_dai_link_tdm_mask { + u32 tx; + u32 rx; +}; + +struct axg_dai_link_tdm_data { + unsigned int mclk_fs; + unsigned int slots; + unsigned int slot_width; + u32 *tx_mask; + u32 *rx_mask; + struct axg_dai_link_tdm_mask *codec_masks; +}; + +#define PREFIX "amlogic," + +static int axg_card_reallocate_links(struct axg_card *priv, + unsigned int num_links) +{ + struct snd_soc_dai_link *links; + void **ldata; + + links = krealloc(priv->card.dai_link, + num_links * sizeof(*priv->card.dai_link), + GFP_KERNEL | __GFP_ZERO); + ldata = krealloc(priv->link_data, + num_links * sizeof(*priv->link_data), + GFP_KERNEL | __GFP_ZERO); + + if (!links || !ldata) { + dev_err(priv->card.dev, "failed to allocate links\n"); + return -ENOMEM; + } + + priv->card.dai_link = links; + priv->link_data = ldata; + priv->card.num_links = num_links; + return 0; +} + +static int axg_card_parse_dai(struct snd_soc_card *card, + struct device_node *node, + struct device_node **dai_of_node, + const char **dai_name) +{ + struct of_phandle_args args; + int ret; + + if (!dai_name || !dai_of_node || !node) + return -EINVAL; + + ret = of_parse_phandle_with_args(node, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(card->dev, "can't parse dai %d\n", ret); + return ret; + } + *dai_of_node = args.np; + + return snd_soc_get_dai_name(&args, dai_name); +} + +static int axg_card_set_link_name(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + const char *prefix) +{ + char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s", + prefix, link->cpu_of_node->full_name); + if (!name) + return -ENOMEM; + + link->name = name; + link->stream_name = name; + + return 0; +} + +static void axg_card_clean_references(struct axg_card *priv) +{ + struct snd_soc_card *card = &priv->card; + struct snd_soc_dai_link *link; + int i, j; + + if (card->dai_link) { + for (i = 0; i < card->num_links; i++) { + link = &card->dai_link[i]; + of_node_put(link->cpu_of_node); + for (j = 0; j < link->num_codecs; j++) + of_node_put(link->codecs[j].of_node); + } + } + + if (card->aux_dev) { + for (i = 0; i < card->num_aux_devs; i++) + of_node_put(card->aux_dev[i].codec_of_node); + } + + kfree(card->dai_link); + kfree(priv->link_data); +} + +static int axg_card_add_aux_devices(struct snd_soc_card *card) +{ + struct device_node *node = card->dev->of_node; + struct snd_soc_aux_dev *aux; + int num, i; + + num = of_count_phandle_with_args(node, "audio-aux-devs", NULL); + if (num == -ENOENT) { + /* + * It is ok to have no auxiliary devices but for this card it + * is a strange situtation. Let's warn the about it. + */ + dev_warn(card->dev, "card has no auxiliary devices\n"); + return 0; + } else if (num < 0) { + dev_err(card->dev, "error getting auxiliary devices: %d\n", + num); + return num; + } + + aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL); + if (!aux) + return -ENOMEM; + card->aux_dev = aux; + card->num_aux_devs = num; + + for (i = 0; i < card->num_aux_devs; i++, aux++) { + aux->codec_of_node = + of_parse_phandle(node, "audio-aux-devs", i); + if (!aux->codec_of_node) + return -EINVAL; + } + + return 0; +} + +static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct axg_dai_link_tdm_data *be = + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + struct snd_soc_dai *codec_dai; + unsigned int mclk; + int ret, i; + + if (be->mclk_fs) { + mclk = params_rate(params) * be->mclk_fs; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret && ret != -ENOTSUPP) + return ret; + } + + return 0; +} + +static const struct snd_soc_ops axg_card_tdm_be_ops = { + .hw_params = axg_card_tdm_be_hw_params, +}; + +static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct axg_dai_link_tdm_data *be = + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + struct snd_soc_dai *codec_dai; + int ret, i; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + ret = snd_soc_dai_set_tdm_slot(codec_dai, + be->codec_masks[i].tx, + be->codec_masks[i].rx, + be->slots, be->slot_width); + if (ret && ret != -ENOTSUPP) { + dev_err(codec_dai->dev, + "setting tdm link slots failed\n"); + return ret; + } + } + + ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, be->tx_mask, be->rx_mask, + be->slots, be->slot_width); + if (ret) { + dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n"); + return ret; + } + + return 0; +} + +static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct axg_dai_link_tdm_data *be = + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + int ret; + + /* The loopback rx_mask is the pad tx_mask */ + ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, NULL, be->tx_mask, + be->slots, be->slot_width); + if (ret) { + dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n"); + return ret; + } + + return 0; +} + +static int axg_card_add_tdm_loopback(struct snd_soc_card *card, + int *index) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai_link *pad = &card->dai_link[*index]; + struct snd_soc_dai_link *lb; + int ret; + + /* extend links */ + ret = axg_card_reallocate_links(priv, card->num_links + 1); + if (ret) + return ret; + + lb = &card->dai_link[*index + 1]; + + lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name); + if (!lb->name) + return -ENOMEM; + + lb->stream_name = lb->name; + lb->cpu_of_node = pad->cpu_of_node; + lb->cpu_dai_name = "TDM Loopback"; + lb->codec_name = "snd-soc-dummy"; + lb->codec_dai_name = "snd-soc-dummy-dai"; + lb->dpcm_capture = 1; + lb->no_pcm = 1; + lb->ops = &axg_card_tdm_be_ops; + lb->init = axg_card_tdm_dai_lb_init; + + /* Provide the same link data to the loopback */ + priv->link_data[*index + 1] = priv->link_data[*index]; + + /* + * axg_card_clean_references() will iterate over this link, + * make sure the node count is balanced + */ + of_node_get(lb->cpu_of_node); + + /* Let add_links continue where it should */ + *index += 1; + + return 0; +} + +static unsigned int axg_card_parse_daifmt(struct device_node *node, + struct device_node *cpu_node) +{ + struct device_node *bitclkmaster = NULL; + struct device_node *framemaster = NULL; + unsigned int daifmt; + + daifmt = snd_soc_of_parse_daifmt(node, PREFIX, + &bitclkmaster, &framemaster); + daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + + /* If no master is provided, default to cpu master */ + if (!bitclkmaster || bitclkmaster == cpu_node) { + daifmt |= (!framemaster || framemaster == cpu_node) ? + SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBS_CFM; + } else { + daifmt |= (!framemaster || framemaster == cpu_node) ? + SND_SOC_DAIFMT_CBM_CFS : SND_SOC_DAIFMT_CBM_CFM; + } + + of_node_put(bitclkmaster); + of_node_put(framemaster); + + return daifmt; +} + +static int axg_card_parse_cpu_tdm_slots(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node, + struct axg_dai_link_tdm_data *be) +{ + char propname[32]; + u32 tx, rx; + int i; + + be->tx_mask = devm_kcalloc(card->dev, AXG_TDM_NUM_LANES, + sizeof(*be->tx_mask), GFP_KERNEL); + be->rx_mask = devm_kcalloc(card->dev, AXG_TDM_NUM_LANES, + sizeof(*be->rx_mask), GFP_KERNEL); + if (!be->tx_mask || !be->rx_mask) + return -ENOMEM; + + for (i = 0, tx = 0; i < AXG_TDM_NUM_LANES; i++) { + snprintf(propname, 32, "dai-tdm-slot-tx-mask-%d", i); + snd_soc_of_get_slot_mask(node, propname, &be->tx_mask[i]); + tx = max(tx, be->tx_mask[i]); + } + + /* Disable playback is the interface has no tx slots */ + if (!tx) + link->dpcm_playback = 0; + + for (i = 0, rx = 0; i < AXG_TDM_NUM_LANES; i++) { + snprintf(propname, 32, "dai-tdm-slot-rx-mask-%d", i); + snd_soc_of_get_slot_mask(node, propname, &be->rx_mask[i]); + rx = max(rx, be->rx_mask[i]); + } + + /* Disable capture is the interface has no rx slots */ + if (!rx) + link->dpcm_capture = 0; + + /* ... but the interface should at least have one of them */ + if (!tx && !rx) { + dev_err(card->dev, "tdm link has no cpu slots\n"); + return -EINVAL; + } + + of_property_read_u32(node, "dai-tdm-slot-num", &be->slots); + if (!be->slots) { + /* + * If the slot number is not provided, set it such as it + * accommodates the largest mask + */ + be->slots = fls(max(tx, rx)); + } else if (be->slots < fls(max(tx, rx)) || be->slots > 32) { + /* + * Error if the slots can't accommodate the largest mask or + * if it is just too big + */ + dev_err(card->dev, "bad slot number\n"); + return -EINVAL; + } + + of_property_read_u32(node, "dai-tdm-slot-width", &be->slot_width); + + return 0; +} + +static int axg_card_parse_codecs_masks(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node, + struct axg_dai_link_tdm_data *be) +{ + struct axg_dai_link_tdm_mask *codec_mask; + struct device_node *np; + + codec_mask = devm_kcalloc(card->dev, link->num_codecs, + sizeof(*codec_mask), GFP_KERNEL); + if (!codec_mask) + return -ENOMEM; + + be->codec_masks = codec_mask; + + for_each_child_of_node(node, np) { + snd_soc_of_get_slot_mask(np, "dai-tdm-slot-rx-mask", + &codec_mask->rx); + snd_soc_of_get_slot_mask(np, "dai-tdm-slot-tx-mask", + &codec_mask->tx); + + codec_mask++; + } + + return 0; +} + +static int axg_card_parse_tdm(struct snd_soc_card *card, + struct device_node *node, + int *index) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai_link *link = &card->dai_link[*index]; + struct axg_dai_link_tdm_data *be; + int ret; + + /* Allocate tdm link parameters */ + be = devm_kzalloc(card->dev, sizeof(*be), GFP_KERNEL); + if (!be) + return -ENOMEM; + priv->link_data[*index] = be; + + /* Setup tdm link */ + link->ops = &axg_card_tdm_be_ops; + link->init = axg_card_tdm_dai_init; + link->dai_fmt = axg_card_parse_daifmt(node, link->cpu_of_node); + + of_property_read_u32(node, "mclk-fs", &be->mclk_fs); + + ret = axg_card_parse_cpu_tdm_slots(card, link, node, be); + if (ret) { + dev_err(card->dev, "error parsing tdm link slots\n"); + return ret; + } + + ret = axg_card_parse_codecs_masks(card, link, node, be); + if (ret) + return ret; + + /* Add loopback if the pad dai has playback */ + if (link->dpcm_playback) { + ret = axg_card_add_tdm_loopback(card, index); + if (ret) + return ret; + } + + return 0; +} + +static int axg_card_set_be_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node) +{ + struct snd_soc_dai_link_component *codec; + struct device_node *np; + int ret, num_codecs; + + link->no_pcm = 1; + link->dpcm_playback = 1; + link->dpcm_capture = 1; + + num_codecs = of_get_child_count(node); + if (!num_codecs) { + dev_err(card->dev, "be link %s has no codec\n", + node->full_name); + return -EINVAL; + } + + codec = devm_kcalloc(card->dev, num_codecs, sizeof(*codec), GFP_KERNEL); + if (!codec) + return -ENOMEM; + + link->codecs = codec; + link->num_codecs = num_codecs; + + for_each_child_of_node(node, np) { + ret = axg_card_parse_dai(card, np, &codec->of_node, + &codec->dai_name); + if (ret) { + of_node_put(np); + return ret; + } + + codec++; + } + + ret = axg_card_set_link_name(card, link, "be"); + if (ret) + dev_err(card->dev, "error setting %s link name\n", np->name); + + return ret; +} + +static int axg_card_set_fe_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + bool is_playback) +{ + link->dynamic = 1; + link->dpcm_merged_format = 1; + link->dpcm_merged_chan = 1; + link->dpcm_merged_rate = 1; + link->codec_dai_name = "snd-soc-dummy-dai"; + link->codec_name = "snd-soc-dummy"; + + if (is_playback) + link->dpcm_playback = 1; + else + link->dpcm_capture = 1; + + return axg_card_set_link_name(card, link, "fe"); +} + +static int axg_card_cpu_is_capture_fe(struct device_node *np) +{ + return of_device_is_compatible(np, PREFIX "axg-toddr"); +} + +static int axg_card_cpu_is_playback_fe(struct device_node *np) +{ + return of_device_is_compatible(np, PREFIX "axg-frddr"); +} + +static int axg_card_cpu_is_tdm_iface(struct device_node *np) +{ + return of_device_is_compatible(np, PREFIX "axg-tdm-iface"); +} + +static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, + int *index) +{ + struct snd_soc_dai_link *dai_link = &card->dai_link[*index]; + int ret; + + ret = axg_card_parse_dai(card, np, &dai_link->cpu_of_node, + &dai_link->cpu_dai_name); + if (ret) + return ret; + + if (axg_card_cpu_is_playback_fe(dai_link->cpu_of_node)) + ret = axg_card_set_fe_link(card, dai_link, true); + else if (axg_card_cpu_is_capture_fe(dai_link->cpu_of_node)) + ret = axg_card_set_fe_link(card, dai_link, false); + else + ret = axg_card_set_be_link(card, dai_link, np); + + if (ret) + return ret; + + if (axg_card_cpu_is_tdm_iface(dai_link->cpu_of_node)) + ret = axg_card_parse_tdm(card, np, index); + + return ret; +} + +static int axg_card_add_links(struct snd_soc_card *card) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct device_node *node = card->dev->of_node; + struct device_node *np; + int num, i, ret; + + num = of_get_child_count(node); + if (!num) { + dev_err(card->dev, "card has no links\n"); + return -EINVAL; + } + + ret = axg_card_reallocate_links(priv, num); + if (ret) + return ret; + + i = 0; + for_each_child_of_node(node, np) { + ret = axg_card_add_link(card, np, &i); + if (ret) { + of_node_put(np); + return ret; + } + + i++; + } + + return 0; +} + +static int axg_card_parse_of_optional(struct snd_soc_card *card, + const char *propname, + int (*func)(struct snd_soc_card *c, + const char *p)) +{ + /* If property is not provided, don't fail ... */ + if (!of_property_read_bool(card->dev->of_node, propname)) + return 0; + + /* ... but do fail if it is provided and the parsing fails */ + return func(card, propname); +} + +static const struct of_device_id axg_card_of_match[] = { + { .compatible = "amlogic,axg-sound-card", }, + {} +}; +MODULE_DEVICE_TABLE(of, axg_card_of_match); + +static int axg_card_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct axg_card *priv; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + platform_set_drvdata(pdev, priv); + snd_soc_card_set_drvdata(&priv->card, priv); + + priv->card.owner = THIS_MODULE; + priv->card.dev = dev; + + ret = snd_soc_of_parse_card_name(&priv->card, "model"); + if (ret < 0) + return ret; + + ret = axg_card_parse_of_optional(&priv->card, "audio-routing", + snd_soc_of_parse_audio_routing); + if (ret) { + dev_err(dev, "error while parsing routing\n"); + return ret; + } + + ret = axg_card_parse_of_optional(&priv->card, "audio-widgets", + snd_soc_of_parse_audio_simple_widgets); + if (ret) { + dev_err(dev, "error while parsing widgets\n"); + return ret; + } + + ret = axg_card_add_links(&priv->card); + if (ret) + goto out_err; + + ret = axg_card_add_aux_devices(&priv->card); + if (ret) + goto out_err; + + ret = devm_snd_soc_register_card(dev, &priv->card); + if (ret) + goto out_err; + + return 0; + +out_err: + axg_card_clean_references(priv); + return ret; +} + +static int axg_card_remove(struct platform_device *pdev) +{ + struct axg_card *priv = platform_get_drvdata(pdev); + + axg_card_clean_references(priv); + + return 0; +} + +static struct platform_driver axg_card_pdrv = { + .probe = axg_card_probe, + .remove = axg_card_remove, + .driver = { + .name = "axg-sound-card", + .of_match_table = axg_card_of_match, + }, +}; +module_platform_driver(axg_card_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG ALSA machine driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c new file mode 100644 index 000000000000..30262550e37b --- /dev/null +++ b/sound/soc/meson/axg-fifo.c @@ -0,0 +1,341 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/clk.h> +#include <linux/of_irq.h> +#include <linux/of_platform.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/reset.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "axg-fifo.h" + +/* + * This file implements the platform operations common to the playback and + * capture frontend DAI. The logic behind this two types of fifo is very + * similar but some difference exist. + * These differences the respective DAI drivers + */ + +static struct snd_pcm_hardware axg_fifo_hw = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE), + + .formats = AXG_FIFO_FORMATS, + .rate_min = 5512, + .rate_max = 192000, + .channels_min = 1, + .channels_max = AXG_FIFO_CH_MAX, + .period_bytes_min = AXG_FIFO_MIN_DEPTH, + .period_bytes_max = UINT_MAX, + .periods_min = 2, + .periods_max = UINT_MAX, + + /* No real justification for this */ + .buffer_bytes_max = 1 * 1024 * 1024, +}; + +static struct snd_soc_dai *axg_fifo_dai(struct snd_pcm_substream *ss) +{ + struct snd_soc_pcm_runtime *rtd = ss->private_data; + + return rtd->cpu_dai; +} + +static struct axg_fifo *axg_fifo_data(struct snd_pcm_substream *ss) +{ + struct snd_soc_dai *dai = axg_fifo_dai(ss); + + return snd_soc_dai_get_drvdata(dai); +} + +static struct device *axg_fifo_dev(struct snd_pcm_substream *ss) +{ + struct snd_soc_dai *dai = axg_fifo_dai(ss); + + return dai->dev; +} + +static void __dma_enable(struct axg_fifo *fifo, bool enable) +{ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_DMA_EN, + enable ? CTRL0_DMA_EN : 0); +} + +static int axg_fifo_pcm_trigger(struct snd_pcm_substream *ss, int cmd) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + __dma_enable(fifo, true); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_STOP: + __dma_enable(fifo, false); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t axg_fifo_pcm_pointer(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + struct snd_pcm_runtime *runtime = ss->runtime; + unsigned int addr; + + regmap_read(fifo->map, FIFO_STATUS2, &addr); + + return bytes_to_frames(runtime, addr - (unsigned int)runtime->dma_addr); +} + +static int axg_fifo_pcm_hw_params(struct snd_pcm_substream *ss, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = ss->runtime; + struct axg_fifo *fifo = axg_fifo_data(ss); + dma_addr_t end_ptr; + unsigned int burst_num; + int ret; + + ret = snd_pcm_lib_malloc_pages(ss, params_buffer_bytes(params)); + if (ret < 0) + return ret; + + /* Setup dma memory pointers */ + end_ptr = runtime->dma_addr + runtime->dma_bytes - AXG_FIFO_BURST; + regmap_write(fifo->map, FIFO_START_ADDR, runtime->dma_addr); + regmap_write(fifo->map, FIFO_FINISH_ADDR, end_ptr); + + /* Setup interrupt periodicity */ + burst_num = params_period_bytes(params) / AXG_FIFO_BURST; + regmap_write(fifo->map, FIFO_INT_ADDR, burst_num); + + /* Enable block count irq */ + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), + CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT)); + + return 0; +} + +static int axg_fifo_pcm_hw_free(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + + /* Disable the block count irq */ + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), 0); + + return snd_pcm_lib_free_pages(ss); +} + +static void axg_fifo_ack_irq(struct axg_fifo *fifo, u8 mask) +{ + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_INT_CLR(FIFO_INT_MASK), + CTRL1_INT_CLR(mask)); + + /* Clear must also be cleared */ + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_INT_CLR(FIFO_INT_MASK), + 0); +} + +static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) +{ + struct snd_pcm_substream *ss = dev_id; + struct axg_fifo *fifo = axg_fifo_data(ss); + unsigned int status; + + regmap_read(fifo->map, FIFO_STATUS1, &status); + + status = STATUS1_INT_STS(status) & FIFO_INT_MASK; + if (status & FIFO_INT_COUNT_REPEAT) + snd_pcm_period_elapsed(ss); + else + dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", + status); + + /* Ack irqs */ + axg_fifo_ack_irq(fifo, status); + + return IRQ_RETVAL(status); +} + +static int axg_fifo_pcm_open(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + struct device *dev = axg_fifo_dev(ss); + int ret; + + snd_soc_set_runtime_hwparams(ss, &axg_fifo_hw); + + /* + * Make sure the buffer and period size are multiple of the FIFO + * minimum depth size + */ + ret = snd_pcm_hw_constraint_step(ss->runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + AXG_FIFO_MIN_DEPTH); + if (ret) + return ret; + + ret = snd_pcm_hw_constraint_step(ss->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + AXG_FIFO_MIN_DEPTH); + if (ret) + return ret; + + ret = request_irq(fifo->irq, axg_fifo_pcm_irq_block, 0, + dev_name(dev), ss); + + /* Enable pclk to access registers and clock the fifo ip */ + ret = clk_prepare_enable(fifo->pclk); + if (ret) + return ret; + + /* Setup status2 so it reports the memory pointer */ + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_STATUS2_SEL_MASK, + CTRL1_STATUS2_SEL(STATUS2_SEL_DDR_READ)); + + /* Make sure the dma is initially disabled */ + __dma_enable(fifo, false); + + /* Disable irqs until params are ready */ + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_INT_EN(FIFO_INT_MASK), 0); + + /* Clear any pending interrupt */ + axg_fifo_ack_irq(fifo, FIFO_INT_MASK); + + /* Take memory arbitror out of reset */ + ret = reset_control_deassert(fifo->arb); + if (ret) + clk_disable_unprepare(fifo->pclk); + + return ret; +} + +static int axg_fifo_pcm_close(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + int ret; + + /* Put the memory arbitror back in reset */ + ret = reset_control_assert(fifo->arb); + + /* Disable fifo ip and register access */ + clk_disable_unprepare(fifo->pclk); + + /* remove IRQ */ + free_irq(fifo->irq, ss); + + return ret; +} + +const struct snd_pcm_ops axg_fifo_pcm_ops = { + .open = axg_fifo_pcm_open, + .close = axg_fifo_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = axg_fifo_pcm_hw_params, + .hw_free = axg_fifo_pcm_hw_free, + .pointer = axg_fifo_pcm_pointer, + .trigger = axg_fifo_pcm_trigger, +}; +EXPORT_SYMBOL_GPL(axg_fifo_pcm_ops); + +int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type) +{ + struct snd_card *card = rtd->card->snd_card; + size_t size = axg_fifo_hw.buffer_bytes_max; + + return snd_pcm_lib_preallocate_pages(rtd->pcm->streams[type].substream, + SNDRV_DMA_TYPE_DEV, card->dev, + size, size); +} +EXPORT_SYMBOL_GPL(axg_fifo_pcm_new); + +static const struct regmap_config axg_fifo_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = FIFO_STATUS2, +}; + +int axg_fifo_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + const struct axg_fifo_match_data *data; + struct axg_fifo *fifo; + struct resource *res; + void __iomem *regs; + + data = of_device_get_match_data(dev); + if (!data) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + fifo = devm_kzalloc(dev, sizeof(*fifo), GFP_KERNEL); + if (!fifo) + return -ENOMEM; + platform_set_drvdata(pdev, fifo); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + fifo->map = devm_regmap_init_mmio(dev, regs, &axg_fifo_regmap_cfg); + if (IS_ERR(fifo->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(fifo->map)); + return PTR_ERR(fifo->map); + } + + fifo->pclk = devm_clk_get(dev, NULL); + if (IS_ERR(fifo->pclk)) { + if (PTR_ERR(fifo->pclk) != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %ld\n", + PTR_ERR(fifo->pclk)); + return PTR_ERR(fifo->pclk); + } + + fifo->arb = devm_reset_control_get_exclusive(dev, NULL); + if (IS_ERR(fifo->arb)) { + if (PTR_ERR(fifo->arb) != -EPROBE_DEFER) + dev_err(dev, "failed to get arb reset: %ld\n", + PTR_ERR(fifo->arb)); + return PTR_ERR(fifo->arb); + } + + fifo->irq = of_irq_get(dev->of_node, 0); + if (fifo->irq <= 0) { + dev_err(dev, "failed to get irq: %d\n", fifo->irq); + return fifo->irq; + } + + return devm_snd_soc_register_component(dev, data->component_drv, + data->dai_drv, 1); +} +EXPORT_SYMBOL_GPL(axg_fifo_probe); + +MODULE_DESCRIPTION("Amlogic AXG fifo driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h new file mode 100644 index 000000000000..cb6c4013ca33 --- /dev/null +++ b/sound/soc/meson/axg-fifo.h @@ -0,0 +1,80 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */ +/* + * Copyright (c) 2018 BayLibre, SAS. + * Author: Jerome Brunet <jbrunet@baylibre.com> + */ + +#ifndef _MESON_AXG_FIFO_H +#define _MESON_AXG_FIFO_H + +struct clk; +struct platform_device; +struct regmap; +struct reset_control; + +struct snd_soc_component_driver; +struct snd_soc_dai; +struct snd_soc_dai_driver; +struct snd_pcm_ops; +struct snd_soc_pcm_runtime; + +#define AXG_FIFO_CH_MAX 128 +#define AXG_FIFO_RATES (SNDRV_PCM_RATE_5512 | \ + SNDRV_PCM_RATE_8000_192000) +#define AXG_FIFO_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define AXG_FIFO_BURST 8 +#define AXG_FIFO_MIN_CNT 64 +#define AXG_FIFO_MIN_DEPTH (AXG_FIFO_BURST * AXG_FIFO_MIN_CNT) + +#define FIFO_INT_ADDR_FINISH BIT(0) +#define FIFO_INT_ADDR_INT BIT(1) +#define FIFO_INT_COUNT_REPEAT BIT(2) +#define FIFO_INT_COUNT_ONCE BIT(3) +#define FIFO_INT_FIFO_ZERO BIT(4) +#define FIFO_INT_FIFO_DEPTH BIT(5) +#define FIFO_INT_MASK GENMASK(7, 0) + +#define FIFO_CTRL0 0x00 +#define CTRL0_DMA_EN BIT(31) +#define CTRL0_INT_EN(x) ((x) << 16) +#define CTRL0_SEL_MASK GENMASK(2, 0) +#define CTRL0_SEL_SHIFT 0 +#define FIFO_CTRL1 0x04 +#define CTRL1_INT_CLR(x) ((x) << 0) +#define CTRL1_STATUS2_SEL_MASK GENMASK(11, 8) +#define CTRL1_STATUS2_SEL(x) ((x) << 8) +#define STATUS2_SEL_DDR_READ 0 +#define CTRL1_THRESHOLD_MASK GENMASK(23, 16) +#define CTRL1_THRESHOLD(x) ((x) << 16) +#define CTRL1_FRDDR_DEPTH_MASK GENMASK(31, 24) +#define CTRL1_FRDDR_DEPTH(x) ((x) << 24) +#define FIFO_START_ADDR 0x08 +#define FIFO_FINISH_ADDR 0x0c +#define FIFO_INT_ADDR 0x10 +#define FIFO_STATUS1 0x14 +#define STATUS1_INT_STS(x) ((x) << 0) +#define FIFO_STATUS2 0x18 + +struct axg_fifo { + struct regmap *map; + struct clk *pclk; + struct reset_control *arb; + int irq; +}; + +struct axg_fifo_match_data { + const struct snd_soc_component_driver *component_drv; + struct snd_soc_dai_driver *dai_drv; +}; + +extern const struct snd_pcm_ops axg_fifo_pcm_ops; + +int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type); +int axg_fifo_probe(struct platform_device *pdev); + +#endif /* _MESON_AXG_FIFO_H */ diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c new file mode 100644 index 000000000000..a6f6f6a2eca8 --- /dev/null +++ b/sound/soc/meson/axg-frddr.c @@ -0,0 +1,141 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +/* This driver implements the frontend playback DAI of AXG based SoCs */ + +#include <linux/clk.h> +#include <linux/regmap.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "axg-fifo.h" + +#define CTRL0_FRDDR_PP_MODE BIT(30) + +static int axg_frddr_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + unsigned int fifo_depth, fifo_threshold; + int ret; + + /* Enable pclk to access registers and clock the fifo ip */ + ret = clk_prepare_enable(fifo->pclk); + if (ret) + return ret; + + /* Apply single buffer mode to the interface */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_FRDDR_PP_MODE, 0); + + /* + * TODO: We could adapt the fifo depth and the fifo threshold + * depending on the expected memory throughput and lantencies + * For now, we'll just use the same values as the vendor kernel + * Depth and threshold are zero based. + */ + fifo_depth = AXG_FIFO_MIN_CNT - 1; + fifo_threshold = (AXG_FIFO_MIN_CNT / 2) - 1; + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_FRDDR_DEPTH_MASK | CTRL1_THRESHOLD_MASK, + CTRL1_FRDDR_DEPTH(fifo_depth) | + CTRL1_THRESHOLD(fifo_threshold)); + + return 0; +} + +static void axg_frddr_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(fifo->pclk); +} + +static int axg_frddr_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + return axg_fifo_pcm_new(rtd, SNDRV_PCM_STREAM_PLAYBACK); +} + +static const struct snd_soc_dai_ops axg_frddr_ops = { + .startup = axg_frddr_dai_startup, + .shutdown = axg_frddr_dai_shutdown, +}; + +static struct snd_soc_dai_driver axg_frddr_dai_drv = { + .name = "FRDDR", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = AXG_FIFO_CH_MAX, + .rates = AXG_FIFO_RATES, + .formats = AXG_FIFO_FORMATS, + }, + .ops = &axg_frddr_ops, + .pcm_new = axg_frddr_pcm_new, +}; + +static const char * const axg_frddr_sel_texts[] = { + "OUT 0", "OUT 1", "OUT 2", "OUT 3" +}; + +static SOC_ENUM_SINGLE_DECL(axg_frddr_sel_enum, FIFO_CTRL0, CTRL0_SEL_SHIFT, + axg_frddr_sel_texts); + +static const struct snd_kcontrol_new axg_frddr_out_demux = + SOC_DAPM_ENUM("Output Sink", axg_frddr_sel_enum); + +static const struct snd_soc_dapm_widget axg_frddr_dapm_widgets[] = { + SND_SOC_DAPM_DEMUX("SINK SEL", SND_SOC_NOPM, 0, 0, + &axg_frddr_out_demux), + SND_SOC_DAPM_AIF_OUT("OUT 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 3", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_frddr_dapm_routes[] = { + { "SINK SEL", NULL, "Playback" }, + { "OUT 0", "OUT 0", "SINK SEL" }, + { "OUT 1", "OUT 1", "SINK SEL" }, + { "OUT 2", "OUT 2", "SINK SEL" }, + { "OUT 3", "OUT 3", "SINK SEL" }, +}; + +static const struct snd_soc_component_driver axg_frddr_component_drv = { + .dapm_widgets = axg_frddr_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_frddr_dapm_widgets), + .dapm_routes = axg_frddr_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_frddr_dapm_routes), + .ops = &axg_fifo_pcm_ops +}; + +static const struct axg_fifo_match_data axg_frddr_match_data = { + .component_drv = &axg_frddr_component_drv, + .dai_drv = &axg_frddr_dai_drv +}; + +static const struct of_device_id axg_frddr_of_match[] = { + { + .compatible = "amlogic,axg-frddr", + .data = &axg_frddr_match_data, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_frddr_of_match); + +static struct platform_driver axg_frddr_pdrv = { + .probe = axg_fifo_probe, + .driver = { + .name = "axg-frddr", + .of_match_table = axg_frddr_of_match, + }, +}; +module_platform_driver(axg_frddr_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG playback fifo driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c new file mode 100644 index 000000000000..9dea528053ad --- /dev/null +++ b/sound/soc/meson/axg-spdifout.c @@ -0,0 +1,456 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/pcm_params.h> +#include <sound/pcm_iec958.h> + +/* + * NOTE: + * The meaning of bits SPDIFOUT_CTRL0_XXX_SEL is actually the opposite + * of what the documentation says. Manual control on V, U and C bits is + * applied when the related sel bits are cleared + */ + +#define SPDIFOUT_STAT 0x00 +#define SPDIFOUT_GAIN0 0x04 +#define SPDIFOUT_GAIN1 0x08 +#define SPDIFOUT_CTRL0 0x0c +#define SPDIFOUT_CTRL0_EN BIT(31) +#define SPDIFOUT_CTRL0_RST_OUT BIT(29) +#define SPDIFOUT_CTRL0_RST_IN BIT(28) +#define SPDIFOUT_CTRL0_USEL BIT(26) +#define SPDIFOUT_CTRL0_USET BIT(25) +#define SPDIFOUT_CTRL0_CHSTS_SEL BIT(24) +#define SPDIFOUT_CTRL0_DATA_SEL BIT(20) +#define SPDIFOUT_CTRL0_MSB_FIRST BIT(19) +#define SPDIFOUT_CTRL0_VSEL BIT(18) +#define SPDIFOUT_CTRL0_VSET BIT(17) +#define SPDIFOUT_CTRL0_MASK_MASK GENMASK(11, 4) +#define SPDIFOUT_CTRL0_MASK(x) ((x) << 4) +#define SPDIFOUT_CTRL1 0x10 +#define SPDIFOUT_CTRL1_MSB_POS_MASK GENMASK(12, 8) +#define SPDIFOUT_CTRL1_MSB_POS(x) ((x) << 8) +#define SPDIFOUT_CTRL1_TYPE_MASK GENMASK(6, 4) +#define SPDIFOUT_CTRL1_TYPE(x) ((x) << 4) +#define SPDIFOUT_PREAMB 0x14 +#define SPDIFOUT_SWAP 0x18 +#define SPDIFOUT_CHSTS0 0x1c +#define SPDIFOUT_CHSTS1 0x20 +#define SPDIFOUT_CHSTS2 0x24 +#define SPDIFOUT_CHSTS3 0x28 +#define SPDIFOUT_CHSTS4 0x2c +#define SPDIFOUT_CHSTS5 0x30 +#define SPDIFOUT_CHSTS6 0x34 +#define SPDIFOUT_CHSTS7 0x38 +#define SPDIFOUT_CHSTS8 0x3c +#define SPDIFOUT_CHSTS9 0x40 +#define SPDIFOUT_CHSTSA 0x44 +#define SPDIFOUT_CHSTSB 0x48 +#define SPDIFOUT_MUTE_VAL 0x4c + +struct axg_spdifout { + struct regmap *map; + struct clk *mclk; + struct clk *pclk; +}; + +static void axg_spdifout_enable(struct regmap *map) +{ + /* Apply both reset */ + regmap_update_bits(map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_RST_OUT | SPDIFOUT_CTRL0_RST_IN, + 0); + + /* Clear out reset before in reset */ + regmap_update_bits(map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_RST_OUT, SPDIFOUT_CTRL0_RST_OUT); + regmap_update_bits(map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_RST_IN, SPDIFOUT_CTRL0_RST_IN); + + /* Enable spdifout */ + regmap_update_bits(map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_EN, + SPDIFOUT_CTRL0_EN); +} + +static void axg_spdifout_disable(struct regmap *map) +{ + regmap_update_bits(map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_EN, 0); +} + +static int axg_spdifout_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + axg_spdifout_enable(priv->map); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + axg_spdifout_disable(priv->map); + return 0; + + default: + return -EINVAL; + } +} + +static int axg_spdifout_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + + /* Use spdif valid bit to perform digital mute */ + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_VSET, + mute ? SPDIFOUT_CTRL0_VSET : 0); + + return 0; +} + +static int axg_spdifout_sample_fmt(struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + unsigned int val; + + /* Set the samples spdifout will pull from the FIFO */ + switch (params_channels(params)) { + case 1: + val = SPDIFOUT_CTRL0_MASK(0x1); + break; + case 2: + val = SPDIFOUT_CTRL0_MASK(0x3); + break; + default: + dev_err(dai->dev, "too many channels for spdif dai: %u\n", + params_channels(params)); + return -EINVAL; + } + + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_MASK_MASK, val); + + /* FIFO data are arranged in chunks of 64bits */ + switch (params_physical_width(params)) { + case 8: + /* 8 samples of 8 bits */ + val = SPDIFOUT_CTRL1_TYPE(0); + break; + case 16: + /* 4 samples of 16 bits - right justified */ + val = SPDIFOUT_CTRL1_TYPE(2); + break; + case 32: + /* 2 samples of 32 bits - right justified */ + val = SPDIFOUT_CTRL1_TYPE(4); + break; + default: + dev_err(dai->dev, "Unsupported physical width: %u\n", + params_physical_width(params)); + return -EINVAL; + } + + /* Position of the MSB in FIFO samples */ + val |= SPDIFOUT_CTRL1_MSB_POS(params_width(params) - 1); + + regmap_update_bits(priv->map, SPDIFOUT_CTRL1, + SPDIFOUT_CTRL1_MSB_POS_MASK | + SPDIFOUT_CTRL1_TYPE_MASK, val); + + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_MSB_FIRST | SPDIFOUT_CTRL0_DATA_SEL, + 0); + + return 0; +} + +static int axg_spdifout_set_chsts(struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + unsigned int offset; + int ret; + u8 cs[4]; + u32 val; + + ret = snd_pcm_create_iec958_consumer_hw_params(params, cs, 4); + if (ret < 0) { + dev_err(dai->dev, "Creating IEC958 channel status failed %d\n", + ret); + return ret; + } + val = cs[0] | cs[1] << 8 | cs[2] << 16 | cs[3] << 24; + + /* Setup channel status A bits [31 - 0]*/ + regmap_write(priv->map, SPDIFOUT_CHSTS0, val); + + /* Clear channel status A bits [191 - 32] */ + for (offset = SPDIFOUT_CHSTS1; offset <= SPDIFOUT_CHSTS5; + offset += regmap_get_reg_stride(priv->map)) + regmap_write(priv->map, offset, 0); + + /* Setup channel status B bits [31 - 0]*/ + regmap_write(priv->map, SPDIFOUT_CHSTS6, val); + + /* Clear channel status B bits [191 - 32] */ + for (offset = SPDIFOUT_CHSTS7; offset <= SPDIFOUT_CHSTSB; + offset += regmap_get_reg_stride(priv->map)) + regmap_write(priv->map, offset, 0); + + return 0; +} + +static int axg_spdifout_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + unsigned int rate = params_rate(params); + int ret; + + /* 2 * 32bits per subframe * 2 channels = 128 */ + ret = clk_set_rate(priv->mclk, rate * 128); + if (ret) { + dev_err(dai->dev, "failed to set spdif clock\n"); + return ret; + } + + ret = axg_spdifout_sample_fmt(params, dai); + if (ret) { + dev_err(dai->dev, "failed to setup sample format\n"); + return ret; + } + + ret = axg_spdifout_set_chsts(params, dai); + if (ret) { + dev_err(dai->dev, "failed to setup channel status words\n"); + return ret; + } + + return 0; +} + +static int axg_spdifout_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + int ret; + + /* Clock the spdif output block */ + ret = clk_prepare_enable(priv->pclk); + if (ret) { + dev_err(dai->dev, "failed to enable pclk\n"); + return ret; + } + + /* Make sure the block is initially stopped */ + axg_spdifout_disable(priv->map); + + /* Insert data from bit 27 lsb first */ + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_MSB_FIRST | SPDIFOUT_CTRL0_DATA_SEL, + 0); + + /* Manual control of V, C and U, U = 0 */ + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_CHSTS_SEL | SPDIFOUT_CTRL0_VSEL | + SPDIFOUT_CTRL0_USEL | SPDIFOUT_CTRL0_USET, + 0); + + /* Static SWAP configuration ATM */ + regmap_write(priv->map, SPDIFOUT_SWAP, 0x10); + + return 0; +} + +static void axg_spdifout_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(priv->pclk); +} + +static const struct snd_soc_dai_ops axg_spdifout_ops = { + .trigger = axg_spdifout_trigger, + .digital_mute = axg_spdifout_digital_mute, + .hw_params = axg_spdifout_hw_params, + .startup = axg_spdifout_startup, + .shutdown = axg_spdifout_shutdown, +}; + +static struct snd_soc_dai_driver axg_spdifout_dai_drv[] = { + { + .name = "SPDIF Output", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000), + .formats = (SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &axg_spdifout_ops, + }, +}; + +static const char * const spdifout_sel_texts[] = { + "IN 0", "IN 1", "IN 2", +}; + +static SOC_ENUM_SINGLE_DECL(axg_spdifout_sel_enum, SPDIFOUT_CTRL1, 24, + spdifout_sel_texts); + +static const struct snd_kcontrol_new axg_spdifout_in_mux = + SOC_DAPM_ENUM("Input Source", axg_spdifout_sel_enum); + +static const struct snd_soc_dapm_widget axg_spdifout_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_spdifout_in_mux), +}; + +static const struct snd_soc_dapm_route axg_spdifout_dapm_routes[] = { + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "Playback", NULL, "SRC SEL" }, +}; + +static const struct snd_kcontrol_new axg_spdifout_controls[] = { + SOC_DOUBLE("Playback Volume", SPDIFOUT_GAIN0, 0, 8, 255, 0), + SOC_DOUBLE("Playback Switch", SPDIFOUT_CTRL0, 22, 21, 1, 1), + SOC_SINGLE("Playback Gain Enable Switch", + SPDIFOUT_CTRL1, 26, 1, 0), + SOC_SINGLE("Playback Channels Mix Switch", + SPDIFOUT_CTRL0, 23, 1, 0), +}; + +static int axg_spdifout_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct axg_spdifout *priv = snd_soc_component_get_drvdata(component); + enum snd_soc_bias_level now = + snd_soc_component_get_bias_level(component); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (now == SND_SOC_BIAS_STANDBY) + ret = clk_prepare_enable(priv->mclk); + break; + + case SND_SOC_BIAS_STANDBY: + if (now == SND_SOC_BIAS_PREPARE) + clk_disable_unprepare(priv->mclk); + break; + + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_ON: + break; + } + + return ret; +} + +static const struct snd_soc_component_driver axg_spdifout_component_drv = { + .controls = axg_spdifout_controls, + .num_controls = ARRAY_SIZE(axg_spdifout_controls), + .dapm_widgets = axg_spdifout_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_spdifout_dapm_widgets), + .dapm_routes = axg_spdifout_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_spdifout_dapm_routes), + .set_bias_level = axg_spdifout_set_bias_level, +}; + +static const struct regmap_config axg_spdifout_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = SPDIFOUT_MUTE_VAL, +}; + +static const struct of_device_id axg_spdifout_of_match[] = { + { .compatible = "amlogic,axg-spdifout", }, + {} +}; +MODULE_DEVICE_TABLE(of, axg_spdifout_of_match); + +static int axg_spdifout_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct axg_spdifout *priv; + struct resource *res; + void __iomem *regs; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + priv->map = devm_regmap_init_mmio(dev, regs, &axg_spdifout_regmap_cfg); + if (IS_ERR(priv->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(priv->map)); + return PTR_ERR(priv->map); + } + + priv->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(priv->pclk)) { + ret = PTR_ERR(priv->pclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %d\n", ret); + return ret; + } + + priv->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(priv->mclk)) { + ret = PTR_ERR(priv->mclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get mclk: %d\n", ret); + return ret; + } + + return devm_snd_soc_register_component(dev, &axg_spdifout_component_drv, + axg_spdifout_dai_drv, ARRAY_SIZE(axg_spdifout_dai_drv)); +} + +static struct platform_driver axg_spdifout_pdrv = { + .probe = axg_spdifout_probe, + .driver = { + .name = "axg-spdifout", + .of_match_table = axg_spdifout_of_match, + }, +}; +module_platform_driver(axg_spdifout_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG SPDIF Output driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c new file mode 100644 index 000000000000..43e390f9358a --- /dev/null +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -0,0 +1,381 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/regmap.h> +#include <sound/soc.h> + +#include "axg-tdm-formatter.h" + +struct axg_tdm_formatter { + struct list_head list; + struct axg_tdm_stream *stream; + const struct axg_tdm_formatter_driver *drv; + struct clk *pclk; + struct clk *sclk; + struct clk *lrclk; + struct clk *sclk_sel; + struct clk *lrclk_sel; + bool enabled; + struct regmap *map; +}; + +int axg_tdm_formatter_set_channel_masks(struct regmap *map, + struct axg_tdm_stream *ts, + unsigned int offset) +{ + unsigned int val, ch = ts->channels; + unsigned long mask; + int i, j; + + /* + * Distribute the channels of the stream over the available slots + * of each TDM lane + */ + for (i = 0; i < AXG_TDM_NUM_LANES; i++) { + val = 0; + mask = ts->mask[i]; + + for (j = find_first_bit(&mask, 32); + (j < 32) && ch; + j = find_next_bit(&mask, 32, j + 1)) { + val |= 1 << j; + ch -= 1; + } + + regmap_write(map, offset, val); + offset += regmap_get_reg_stride(map); + } + + /* + * If we still have channel left at the end of the process, it means + * the stream has more channels than we can accommodate and we should + * have caught this earlier. + */ + if (WARN_ON(ch != 0)) { + pr_err("channel mask error\n"); + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks); + +static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter) +{ + struct axg_tdm_stream *ts = formatter->stream; + bool invert = formatter->drv->invert_sclk; + int ret; + + /* Do nothing if the formatter is already enabled */ + if (formatter->enabled) + return 0; + + /* + * If sclk is inverted, invert it back and provide the inversion + * required by the formatter + */ + invert ^= axg_tdm_sclk_invert(ts->iface->fmt); + ret = clk_set_phase(formatter->sclk, invert ? 180 : 0); + if (ret) + return ret; + + /* Setup the stream parameter in the formatter */ + ret = formatter->drv->ops->prepare(formatter->map, formatter->stream); + if (ret) + return ret; + + /* Enable the signal clocks feeding the formatter */ + ret = clk_prepare_enable(formatter->sclk); + if (ret) + return ret; + + ret = clk_prepare_enable(formatter->lrclk); + if (ret) { + clk_disable_unprepare(formatter->sclk); + return ret; + } + + /* Finally, actually enable the formatter */ + formatter->drv->ops->enable(formatter->map); + formatter->enabled = true; + + return 0; +} + +static void axg_tdm_formatter_disable(struct axg_tdm_formatter *formatter) +{ + /* Do nothing if the formatter is already disabled */ + if (!formatter->enabled) + return; + + formatter->drv->ops->disable(formatter->map); + clk_disable_unprepare(formatter->lrclk); + clk_disable_unprepare(formatter->sclk); + formatter->enabled = false; +} + +static int axg_tdm_formatter_attach(struct axg_tdm_formatter *formatter) +{ + struct axg_tdm_stream *ts = formatter->stream; + int ret = 0; + + mutex_lock(&ts->lock); + + /* Catch up if the stream is already running when we attach */ + if (ts->ready) { + ret = axg_tdm_formatter_enable(formatter); + if (ret) { + pr_err("failed to enable formatter\n"); + goto out; + } + } + + list_add_tail(&formatter->list, &ts->formatter_list); +out: + mutex_unlock(&ts->lock); + return ret; +} + +static void axg_tdm_formatter_dettach(struct axg_tdm_formatter *formatter) +{ + struct axg_tdm_stream *ts = formatter->stream; + + mutex_lock(&ts->lock); + list_del(&formatter->list); + mutex_unlock(&ts->lock); + + axg_tdm_formatter_disable(formatter); +} + +static int axg_tdm_formatter_power_up(struct axg_tdm_formatter *formatter, + struct snd_soc_dapm_widget *w) +{ + struct axg_tdm_stream *ts = formatter->drv->ops->get_stream(w); + int ret; + + /* + * If we don't get a stream at this stage, it would mean that the + * widget is powering up but is not attached to any backend DAI. + * It should not happen, ever ! + */ + if (WARN_ON(!ts)) + return -ENODEV; + + /* Clock our device */ + ret = clk_prepare_enable(formatter->pclk); + if (ret) + return ret; + + /* Reparent the bit clock to the TDM interface */ + ret = clk_set_parent(formatter->sclk_sel, ts->iface->sclk); + if (ret) + goto disable_pclk; + + /* Reparent the sample clock to the TDM interface */ + ret = clk_set_parent(formatter->lrclk_sel, ts->iface->lrclk); + if (ret) + goto disable_pclk; + + formatter->stream = ts; + ret = axg_tdm_formatter_attach(formatter); + if (ret) + goto disable_pclk; + + return 0; + +disable_pclk: + clk_disable_unprepare(formatter->pclk); + return ret; +} + +static void axg_tdm_formatter_power_down(struct axg_tdm_formatter *formatter) +{ + axg_tdm_formatter_dettach(formatter); + clk_disable_unprepare(formatter->pclk); + formatter->stream = NULL; +} + +int axg_tdm_formatter_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *control, + int event) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct axg_tdm_formatter *formatter = snd_soc_component_get_drvdata(c); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = axg_tdm_formatter_power_up(formatter, w); + break; + + case SND_SOC_DAPM_PRE_PMD: + axg_tdm_formatter_power_down(formatter); + break; + + default: + dev_err(c->dev, "Unexpected event %d\n", event); + return -EINVAL; + } + + return ret; +} +EXPORT_SYMBOL_GPL(axg_tdm_formatter_event); + +int axg_tdm_formatter_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + const struct axg_tdm_formatter_driver *drv; + struct axg_tdm_formatter *formatter; + struct resource *res; + void __iomem *regs; + int ret; + + drv = of_device_get_match_data(dev); + if (!drv) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + formatter = devm_kzalloc(dev, sizeof(*formatter), GFP_KERNEL); + if (!formatter) + return -ENOMEM; + platform_set_drvdata(pdev, formatter); + formatter->drv = drv; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + formatter->map = devm_regmap_init_mmio(dev, regs, drv->regmap_cfg); + if (IS_ERR(formatter->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(formatter->map)); + return PTR_ERR(formatter->map); + } + + /* Peripharal clock */ + formatter->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(formatter->pclk)) { + ret = PTR_ERR(formatter->pclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %d\n", ret); + return ret; + } + + /* Formatter bit clock */ + formatter->sclk = devm_clk_get(dev, "sclk"); + if (IS_ERR(formatter->sclk)) { + ret = PTR_ERR(formatter->sclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get sclk: %d\n", ret); + return ret; + } + + /* Formatter sample clock */ + formatter->lrclk = devm_clk_get(dev, "lrclk"); + if (IS_ERR(formatter->lrclk)) { + ret = PTR_ERR(formatter->lrclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get lrclk: %d\n", ret); + return ret; + } + + /* Formatter bit clock input multiplexer */ + formatter->sclk_sel = devm_clk_get(dev, "sclk_sel"); + if (IS_ERR(formatter->sclk_sel)) { + ret = PTR_ERR(formatter->sclk_sel); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get sclk_sel: %d\n", ret); + return ret; + } + + /* Formatter sample clock input multiplexer */ + formatter->lrclk_sel = devm_clk_get(dev, "lrclk_sel"); + if (IS_ERR(formatter->lrclk_sel)) { + ret = PTR_ERR(formatter->lrclk_sel); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get lrclk_sel: %d\n", ret); + return ret; + } + + return devm_snd_soc_register_component(dev, drv->component_drv, + NULL, 0); +} +EXPORT_SYMBOL_GPL(axg_tdm_formatter_probe); + +int axg_tdm_stream_start(struct axg_tdm_stream *ts) +{ + struct axg_tdm_formatter *formatter; + int ret = 0; + + mutex_lock(&ts->lock); + ts->ready = true; + + /* Start all the formatters attached to the stream */ + list_for_each_entry(formatter, &ts->formatter_list, list) { + ret = axg_tdm_formatter_enable(formatter); + if (ret) { + pr_err("failed to start tdm stream\n"); + goto out; + } + } + +out: + mutex_unlock(&ts->lock); + return ret; +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_start); + +void axg_tdm_stream_stop(struct axg_tdm_stream *ts) +{ + struct axg_tdm_formatter *formatter; + + mutex_lock(&ts->lock); + ts->ready = false; + + /* Stop all the formatters attached to the stream */ + list_for_each_entry(formatter, &ts->formatter_list, list) { + axg_tdm_formatter_disable(formatter); + } + + mutex_unlock(&ts->lock); +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_stop); + +struct axg_tdm_stream *axg_tdm_stream_alloc(struct axg_tdm_iface *iface) +{ + struct axg_tdm_stream *ts; + + ts = kzalloc(sizeof(*ts), GFP_KERNEL); + if (ts) { + INIT_LIST_HEAD(&ts->formatter_list); + mutex_init(&ts->lock); + ts->iface = iface; + } + + return ts; +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_alloc); + +void axg_tdm_stream_free(struct axg_tdm_stream *ts) +{ + /* + * If the list is not empty, it would mean that one of the formatter + * widget is still powered and attached to the interface while we + * we are removing the TDM DAI. It should not be possible + */ + WARN_ON(!list_empty(&ts->formatter_list)); + mutex_destroy(&ts->lock); + kfree(ts); +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_free); + +MODULE_DESCRIPTION("Amlogic AXG TDM formatter driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-tdm-formatter.h b/sound/soc/meson/axg-tdm-formatter.h new file mode 100644 index 000000000000..cf947caf3cb1 --- /dev/null +++ b/sound/soc/meson/axg-tdm-formatter.h @@ -0,0 +1,39 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) + * + * Copyright (c) 2018 Baylibre SAS. + * Author: Jerome Brunet <jbrunet@baylibre.com> + */ + +#ifndef _MESON_AXG_TDM_FORMATTER_H +#define _MESON_AXG_TDM_FORMATTER_H + +#include "axg-tdm.h" + +struct platform_device; +struct regmap; +struct snd_soc_dapm_widget; +struct snd_kcontrol; + +struct axg_tdm_formatter_ops { + struct axg_tdm_stream *(*get_stream)(struct snd_soc_dapm_widget *w); + void (*enable)(struct regmap *map); + void (*disable)(struct regmap *map); + int (*prepare)(struct regmap *map, struct axg_tdm_stream *ts); +}; + +struct axg_tdm_formatter_driver { + const struct snd_soc_component_driver *component_drv; + const struct regmap_config *regmap_cfg; + const struct axg_tdm_formatter_ops *ops; + bool invert_sclk; +}; + +int axg_tdm_formatter_set_channel_masks(struct regmap *map, + struct axg_tdm_stream *ts, + unsigned int offset); +int axg_tdm_formatter_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *control, + int event); +int axg_tdm_formatter_probe(struct platform_device *pdev); + +#endif /* _MESON_AXG_TDM_FORMATTER_H */ diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c new file mode 100644 index 000000000000..7b8baf46d968 --- /dev/null +++ b/sound/soc/meson/axg-tdm-interface.c @@ -0,0 +1,542 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "axg-tdm.h" + +enum { + TDM_IFACE_PAD, + TDM_IFACE_LOOPBACK, +}; + +static unsigned int axg_tdm_slots_total(u32 *mask) +{ + unsigned int slots = 0; + int i; + + if (!mask) + return 0; + + /* Count the total number of slots provided by all 4 lanes */ + for (i = 0; i < AXG_TDM_NUM_LANES; i++) + slots += hweight32(mask[i]); + + return slots; +} + +int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, + u32 *rx_mask, unsigned int slots, + unsigned int slot_width) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + struct axg_tdm_stream *tx = (struct axg_tdm_stream *) + dai->playback_dma_data; + struct axg_tdm_stream *rx = (struct axg_tdm_stream *) + dai->capture_dma_data; + unsigned int tx_slots, rx_slots; + + tx_slots = axg_tdm_slots_total(tx_mask); + rx_slots = axg_tdm_slots_total(rx_mask); + + /* We should at least have a slot for a valid interface */ + if (!tx_slots && !rx_slots) { + dev_err(dai->dev, "interface has no slot\n"); + return -EINVAL; + } + + /* + * Amend the dai driver channel number and let dpcm channel merge do + * its job + */ + if (tx) { + tx->mask = tx_mask; + dai->driver->playback.channels_max = tx_slots; + } + + if (rx) { + rx->mask = rx_mask; + dai->driver->capture.channels_max = rx_slots; + } + + iface->slots = slots; + + switch (slot_width) { + case 0: + /* defaults width to 32 if not provided */ + iface->slot_width = 32; + break; + case 8: + case 16: + case 24: + case 32: + iface->slot_width = slot_width; + break; + default: + dev_err(dai->dev, "unsupported slot width: %d\n", slot_width); + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(axg_tdm_set_tdm_slots); + +static int axg_tdm_iface_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + int ret = -ENOTSUPP; + + if (dir == SND_SOC_CLOCK_OUT && clk_id == 0) { + if (!iface->mclk) { + dev_warn(dai->dev, "master clock not provided\n"); + } else { + ret = clk_set_rate(iface->mclk, freq); + if (!ret) + iface->mclk_rate = freq; + } + } + + return ret; +} + +static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + + /* These modes are not supported */ + if (fmt & (SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_CBM_CFS)) { + dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n"); + return -EINVAL; + } + + /* If the TDM interface is the clock master, it requires mclk */ + if (!iface->mclk && (fmt & SND_SOC_DAIFMT_CBS_CFS)) { + dev_err(dai->dev, "cpu clock master: mclk missing\n"); + return -ENODEV; + } + + iface->fmt = fmt; + return 0; +} + +static int axg_tdm_iface_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + struct axg_tdm_stream *ts = + snd_soc_dai_get_dma_data(dai, substream); + int ret; + + if (!axg_tdm_slots_total(ts->mask)) { + dev_err(dai->dev, "interface has not slots\n"); + return -EINVAL; + } + + /* Apply component wide rate symmetry */ + if (dai->component->active) { + ret = snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + iface->rate); + if (ret < 0) { + dev_err(dai->dev, + "can't set iface rate constraint\n"); + return ret; + } + } + + return 0; +} + +static int axg_tdm_iface_set_stream(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); + unsigned int channels = params_channels(params); + unsigned int width = params_width(params); + + /* Save rate and sample_bits for component symmetry */ + iface->rate = params_rate(params); + + /* Make sure this interface can cope with the stream */ + if (axg_tdm_slots_total(ts->mask) < channels) { + dev_err(dai->dev, "not enough slots for channels\n"); + return -EINVAL; + } + + if (iface->slot_width < width) { + dev_err(dai->dev, "incompatible slots width for stream\n"); + return -EINVAL; + } + + /* Save the parameter for tdmout/tdmin widgets */ + ts->physical_width = params_physical_width(params); + ts->width = params_width(params); + ts->channels = params_channels(params); + + return 0; +} + +static int axg_tdm_iface_set_lrclk(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + unsigned int ratio_num; + int ret; + + ret = clk_set_rate(iface->lrclk, params_rate(params)); + if (ret) { + dev_err(dai->dev, "setting sample clock failed: %d\n", ret); + return ret; + } + + switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + /* 50% duty cycle ratio */ + ratio_num = 1; + break; + + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + /* + * A zero duty cycle ratio will result in setting the mininum + * ratio possible which, for this clock, is 1 cycle of the + * parent bclk clock high and the rest low, This is exactly + * what we want here. + */ + ratio_num = 0; + break; + + default: + return -EINVAL; + } + + ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); + if (ret) { + dev_err(dai->dev, + "setting sample clock duty cycle failed: %d\n", ret); + return ret; + } + + /* Set sample clock inversion */ + ret = clk_set_phase(iface->lrclk, + axg_tdm_lrclk_invert(iface->fmt) ? 180 : 0); + if (ret) { + dev_err(dai->dev, + "setting sample clock phase failed: %d\n", ret); + return ret; + } + + return 0; +} + +static int axg_tdm_iface_set_sclk(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + unsigned long srate; + int ret; + + srate = iface->slots * iface->slot_width * params_rate(params); + + if (!iface->mclk_rate) { + /* If no specific mclk is requested, default to bit clock * 4 */ + clk_set_rate(iface->mclk, 4 * srate); + } else { + /* Check if we can actually get the bit clock from mclk */ + if (iface->mclk_rate % srate) { + dev_err(dai->dev, + "can't derive sclk %lu from mclk %lu\n", + srate, iface->mclk_rate); + return -EINVAL; + } + } + + ret = clk_set_rate(iface->sclk, srate); + if (ret) { + dev_err(dai->dev, "setting bit clock failed: %d\n", ret); + return ret; + } + + /* Set the bit clock inversion */ + ret = clk_set_phase(iface->sclk, + axg_tdm_sclk_invert(iface->fmt) ? 0 : 180); + if (ret) { + dev_err(dai->dev, "setting bit clock phase failed: %d\n", ret); + return ret; + } + + return ret; +} + +static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + int ret; + + switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + if (iface->slots > 2) { + dev_err(dai->dev, "bad slot number for format: %d\n", + iface->slots); + return -EINVAL; + } + break; + + case SND_SOC_DAI_FORMAT_DSP_A: + case SND_SOC_DAI_FORMAT_DSP_B: + break; + + default: + dev_err(dai->dev, "unsupported dai format\n"); + return -EINVAL; + } + + ret = axg_tdm_iface_set_stream(substream, params, dai); + if (ret) + return ret; + + if (iface->fmt & SND_SOC_DAIFMT_CBS_CFS) { + ret = axg_tdm_iface_set_sclk(dai, params); + if (ret) + return ret; + + ret = axg_tdm_iface_set_lrclk(dai, params); + if (ret) + return ret; + } + + return 0; +} + +static int axg_tdm_iface_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); + + /* Stop all attached formatters */ + axg_tdm_stream_stop(ts); + + return 0; +} + +static int axg_tdm_iface_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); + + /* Force all attached formatters to update */ + return axg_tdm_stream_reset(ts); +} + +static int axg_tdm_iface_remove_dai(struct snd_soc_dai *dai) +{ + if (dai->capture_dma_data) + axg_tdm_stream_free(dai->capture_dma_data); + + if (dai->playback_dma_data) + axg_tdm_stream_free(dai->playback_dma_data); + + return 0; +} + +static int axg_tdm_iface_probe_dai(struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + + if (dai->capture_widget) { + dai->capture_dma_data = axg_tdm_stream_alloc(iface); + if (!dai->capture_dma_data) + return -ENOMEM; + } + + if (dai->playback_widget) { + dai->playback_dma_data = axg_tdm_stream_alloc(iface); + if (!dai->playback_dma_data) { + axg_tdm_iface_remove_dai(dai); + return -ENOMEM; + } + } + + return 0; +} + +static const struct snd_soc_dai_ops axg_tdm_iface_ops = { + .set_sysclk = axg_tdm_iface_set_sysclk, + .set_fmt = axg_tdm_iface_set_fmt, + .startup = axg_tdm_iface_startup, + .hw_params = axg_tdm_iface_hw_params, + .prepare = axg_tdm_iface_prepare, + .hw_free = axg_tdm_iface_hw_free, +}; + +/* TDM Backend DAIs */ +static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = { + [TDM_IFACE_PAD] = { + .name = "TDM Pad", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = AXG_TDM_CHANNEL_MAX, + .rates = AXG_TDM_RATES, + .formats = AXG_TDM_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = AXG_TDM_CHANNEL_MAX, + .rates = AXG_TDM_RATES, + .formats = AXG_TDM_FORMATS, + }, + .id = TDM_IFACE_PAD, + .ops = &axg_tdm_iface_ops, + .probe = axg_tdm_iface_probe_dai, + .remove = axg_tdm_iface_remove_dai, + }, + [TDM_IFACE_LOOPBACK] = { + .name = "TDM Loopback", + .capture = { + .stream_name = "Loopback", + .channels_min = 1, + .channels_max = AXG_TDM_CHANNEL_MAX, + .rates = AXG_TDM_RATES, + .formats = AXG_TDM_FORMATS, + }, + .id = TDM_IFACE_LOOPBACK, + .ops = &axg_tdm_iface_ops, + .probe = axg_tdm_iface_probe_dai, + .remove = axg_tdm_iface_remove_dai, + }, +}; + +static int axg_tdm_iface_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct axg_tdm_iface *iface = snd_soc_component_get_drvdata(component); + enum snd_soc_bias_level now = + snd_soc_component_get_bias_level(component); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (now == SND_SOC_BIAS_STANDBY) + ret = clk_prepare_enable(iface->mclk); + break; + + case SND_SOC_BIAS_STANDBY: + if (now == SND_SOC_BIAS_PREPARE) + clk_disable_unprepare(iface->mclk); + break; + + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_ON: + break; + } + + return ret; +} + +static const struct snd_soc_component_driver axg_tdm_iface_component_drv = { + .set_bias_level = axg_tdm_iface_set_bias_level, +}; + +static const struct of_device_id axg_tdm_iface_of_match[] = { + { .compatible = "amlogic,axg-tdm-iface", }, + {} +}; +MODULE_DEVICE_TABLE(of, axg_tdm_iface_of_match); + +static int axg_tdm_iface_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct snd_soc_dai_driver *dai_drv; + struct axg_tdm_iface *iface; + int ret, i; + + iface = devm_kzalloc(dev, sizeof(*iface), GFP_KERNEL); + if (!iface) + return -ENOMEM; + platform_set_drvdata(pdev, iface); + + /* + * Duplicate dai driver: depending on the slot masks configuration + * We'll change the number of channel provided by DAI stream, so dpcm + * channel merge can be done properly + */ + dai_drv = devm_kcalloc(dev, ARRAY_SIZE(axg_tdm_iface_dai_drv), + sizeof(*dai_drv), GFP_KERNEL); + if (!dai_drv) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(axg_tdm_iface_dai_drv); i++) + memcpy(&dai_drv[i], &axg_tdm_iface_dai_drv[i], + sizeof(*dai_drv)); + + /* Bit clock provided on the pad */ + iface->sclk = devm_clk_get(dev, "sclk"); + if (IS_ERR(iface->sclk)) { + ret = PTR_ERR(iface->sclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get sclk: %d\n", ret); + return ret; + } + + /* Sample clock provided on the pad */ + iface->lrclk = devm_clk_get(dev, "lrclk"); + if (IS_ERR(iface->lrclk)) { + ret = PTR_ERR(iface->lrclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get lrclk: %d\n", ret); + return ret; + } + + /* + * mclk maybe be missing when the cpu dai is in slave mode and + * the codec does not require it to provide a master clock. + * At this point, ignore the error if mclk is missing. We'll + * throw an error if the cpu dai is master and mclk is missing + */ + iface->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(iface->mclk)) { + ret = PTR_ERR(iface->mclk); + if (ret == -ENOENT) { + iface->mclk = NULL; + } else { + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get mclk: %d\n", ret); + return ret; + } + } + + return devm_snd_soc_register_component(dev, + &axg_tdm_iface_component_drv, dai_drv, + ARRAY_SIZE(axg_tdm_iface_dai_drv)); +} + +static struct platform_driver axg_tdm_iface_pdrv = { + .probe = axg_tdm_iface_probe, + .driver = { + .name = "axg-tdm-iface", + .of_match_table = axg_tdm_iface_of_match, + }, +}; +module_platform_driver(axg_tdm_iface_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG TDM interface driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h new file mode 100644 index 000000000000..e578b6f40a07 --- /dev/null +++ b/sound/soc/meson/axg-tdm.h @@ -0,0 +1,78 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) + * + * Copyright (c) 2018 Baylibre SAS. + * Author: Jerome Brunet <jbrunet@baylibre.com> + */ + +#ifndef _MESON_AXG_TDM_H +#define _MESON_AXG_TDM_H + +#include <linux/clk.h> +#include <linux/regmap.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#define AXG_TDM_NUM_LANES 4 +#define AXG_TDM_CHANNEL_MAX 128 +#define AXG_TDM_RATES (SNDRV_PCM_RATE_5512 | \ + SNDRV_PCM_RATE_8000_192000) +#define AXG_TDM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +struct axg_tdm_iface { + struct clk *sclk; + struct clk *lrclk; + struct clk *mclk; + unsigned long mclk_rate; + + /* format is common to all the DAIs of the iface */ + unsigned int fmt; + unsigned int slots; + unsigned int slot_width; + + /* For component wide symmetry */ + int rate; +}; + +static inline bool axg_tdm_lrclk_invert(unsigned int fmt) +{ + return (fmt & SND_SOC_DAIFMT_I2S) ^ + !!(fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_NB_IF)); +} + +static inline bool axg_tdm_sclk_invert(unsigned int fmt) +{ + return fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_IB_NF); +} + +struct axg_tdm_stream { + struct axg_tdm_iface *iface; + struct list_head formatter_list; + struct mutex lock; + unsigned int channels; + unsigned int width; + unsigned int physical_width; + u32 *mask; + bool ready; +}; + +struct axg_tdm_stream *axg_tdm_stream_alloc(struct axg_tdm_iface *iface); +void axg_tdm_stream_free(struct axg_tdm_stream *ts); +int axg_tdm_stream_start(struct axg_tdm_stream *ts); +void axg_tdm_stream_stop(struct axg_tdm_stream *ts); + +static inline int axg_tdm_stream_reset(struct axg_tdm_stream *ts) +{ + axg_tdm_stream_stop(ts); + return axg_tdm_stream_start(ts); +} + +int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, + u32 *rx_mask, unsigned int slots, + unsigned int slot_width); + +#endif /* _MESON_AXG_TDM_H */ diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c new file mode 100644 index 000000000000..bbac44c81688 --- /dev/null +++ b/sound/soc/meson/axg-tdmin.c @@ -0,0 +1,229 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "axg-tdm-formatter.h" + +#define TDMIN_CTRL 0x00 +#define TDMIN_CTRL_ENABLE BIT(31) +#define TDMIN_CTRL_I2S_MODE BIT(30) +#define TDMIN_CTRL_RST_OUT BIT(29) +#define TDMIN_CTRL_RST_IN BIT(28) +#define TDMIN_CTRL_WS_INV BIT(25) +#define TDMIN_CTRL_SEL_SHIFT 20 +#define TDMIN_CTRL_IN_BIT_SKEW_MASK GENMASK(18, 16) +#define TDMIN_CTRL_IN_BIT_SKEW(x) ((x) << 16) +#define TDMIN_CTRL_LSB_FIRST BIT(5) +#define TDMIN_CTRL_BITNUM_MASK GENMASK(4, 0) +#define TDMIN_CTRL_BITNUM(x) ((x) << 0) +#define TDMIN_SWAP 0x04 +#define TDMIN_MASK0 0x08 +#define TDMIN_MASK1 0x0c +#define TDMIN_MASK2 0x10 +#define TDMIN_MASK3 0x14 +#define TDMIN_STAT 0x18 +#define TDMIN_MUTE_VAL 0x1c +#define TDMIN_MUTE0 0x20 +#define TDMIN_MUTE1 0x24 +#define TDMIN_MUTE2 0x28 +#define TDMIN_MUTE3 0x2c + +static const struct regmap_config axg_tdmin_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = TDMIN_MUTE3, +}; + +static const char * const axg_tdmin_sel_texts[] = { + "IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 5", +}; + +/* Change to special mux control to reset dapm */ +static SOC_ENUM_SINGLE_DECL(axg_tdmin_sel_enum, TDMIN_CTRL, + TDMIN_CTRL_SEL_SHIFT, axg_tdmin_sel_texts); + +static const struct snd_kcontrol_new axg_tdmin_in_mux = + SOC_DAPM_ENUM("Input Source", axg_tdmin_sel_enum); + +static struct snd_soc_dai * +axg_tdmin_get_be(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dai *be; + + snd_soc_dapm_widget_for_each_source_path(w, p) { + if (!p->connect) + continue; + + if (p->source->id == snd_soc_dapm_dai_out) + return (struct snd_soc_dai *)p->source->priv; + + be = axg_tdmin_get_be(p->source); + if (be) + return be; + } + + return NULL; +} + +static struct axg_tdm_stream * +axg_tdmin_get_tdm_stream(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dai *be = axg_tdmin_get_be(w); + + if (!be) + return NULL; + + return be->capture_dma_data; +} + +static void axg_tdmin_enable(struct regmap *map) +{ + /* Apply both reset */ + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_RST_OUT | TDMIN_CTRL_RST_IN, 0); + + /* Clear out reset before in reset */ + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_RST_OUT, TDMIN_CTRL_RST_OUT); + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_RST_IN, TDMIN_CTRL_RST_IN); + + /* Actually enable tdmin */ + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_ENABLE, TDMIN_CTRL_ENABLE); +} + +static void axg_tdmin_disable(struct regmap *map) +{ + regmap_update_bits(map, TDMIN_CTRL, TDMIN_CTRL_ENABLE, 0); +} + +static int axg_tdmin_prepare(struct regmap *map, struct axg_tdm_stream *ts) +{ + unsigned int val = 0; + + /* Set stream skew */ + switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + val |= TDMIN_CTRL_IN_BIT_SKEW(3); + break; + + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_DSP_B: + val = TDMIN_CTRL_IN_BIT_SKEW(2); + break; + + default: + pr_err("Unsupported format: %u\n", + ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + /* Set stream format mode */ + switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + val |= TDMIN_CTRL_I2S_MODE; + break; + } + + /* If the sample clock is inverted, invert it back for the formatter */ + if (axg_tdm_lrclk_invert(ts->iface->fmt)) + val |= TDMIN_CTRL_WS_INV; + + /* Set the slot width */ + val |= TDMIN_CTRL_BITNUM(ts->iface->slot_width - 1); + + /* + * The following also reset LSB_FIRST which result in the formatter + * placing the first bit received at bit 31 + */ + regmap_update_bits(map, TDMIN_CTRL, + (TDMIN_CTRL_IN_BIT_SKEW_MASK | TDMIN_CTRL_WS_INV | + TDMIN_CTRL_I2S_MODE | TDMIN_CTRL_LSB_FIRST | + TDMIN_CTRL_BITNUM_MASK), val); + + /* Set static swap mask configuration */ + regmap_write(map, TDMIN_SWAP, 0x76543210); + + return axg_tdm_formatter_set_channel_masks(map, ts, TDMIN_MASK0); +} + +static const struct snd_soc_dapm_widget axg_tdmin_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 5", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_tdmin_in_mux), + SND_SOC_DAPM_PGA_E("DEC", SND_SOC_NOPM, 0, 0, NULL, 0, + axg_tdm_formatter_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD)), + SND_SOC_DAPM_AIF_OUT("OUT", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_tdmin_dapm_routes[] = { + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "SRC SEL", "IN 3", "IN 3" }, + { "SRC SEL", "IN 4", "IN 4" }, + { "SRC SEL", "IN 5", "IN 5" }, + { "DEC", NULL, "SRC SEL" }, + { "OUT", NULL, "DEC" }, +}; + +static const struct snd_soc_component_driver axg_tdmin_component_drv = { + .dapm_widgets = axg_tdmin_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_tdmin_dapm_widgets), + .dapm_routes = axg_tdmin_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_tdmin_dapm_routes), +}; + +static const struct axg_tdm_formatter_ops axg_tdmin_ops = { + .get_stream = axg_tdmin_get_tdm_stream, + .prepare = axg_tdmin_prepare, + .enable = axg_tdmin_enable, + .disable = axg_tdmin_disable, +}; + +static const struct axg_tdm_formatter_driver axg_tdmin_drv = { + .component_drv = &axg_tdmin_component_drv, + .regmap_cfg = &axg_tdmin_regmap_cfg, + .ops = &axg_tdmin_ops, + .invert_sclk = false, +}; + +static const struct of_device_id axg_tdmin_of_match[] = { + { + .compatible = "amlogic,axg-tdmin", + .data = &axg_tdmin_drv, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_tdmin_of_match); + +static struct platform_driver axg_tdmin_pdrv = { + .probe = axg_tdm_formatter_probe, + .driver = { + .name = "axg-tdmin", + .of_match_table = axg_tdmin_of_match, + }, +}; +module_platform_driver(axg_tdmin_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG TDM input formatter driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c new file mode 100644 index 000000000000..f73368ee1088 --- /dev/null +++ b/sound/soc/meson/axg-tdmout.c @@ -0,0 +1,259 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "axg-tdm-formatter.h" + +#define TDMOUT_CTRL0 0x00 +#define TDMOUT_CTRL0_BITNUM_MASK GENMASK(4, 0) +#define TDMOUT_CTRL0_BITNUM(x) ((x) << 0) +#define TDMOUT_CTRL0_SLOTNUM_MASK GENMASK(9, 5) +#define TDMOUT_CTRL0_SLOTNUM(x) ((x) << 5) +#define TDMOUT_CTRL0_INIT_BITNUM_MASK GENMASK(19, 15) +#define TDMOUT_CTRL0_INIT_BITNUM(x) ((x) << 15) +#define TDMOUT_CTRL0_ENABLE BIT(31) +#define TDMOUT_CTRL0_RST_OUT BIT(29) +#define TDMOUT_CTRL0_RST_IN BIT(28) +#define TDMOUT_CTRL1 0x04 +#define TDMOUT_CTRL1_TYPE_MASK GENMASK(6, 4) +#define TDMOUT_CTRL1_TYPE(x) ((x) << 4) +#define TDMOUT_CTRL1_MSB_POS_MASK GENMASK(12, 8) +#define TDMOUT_CTRL1_MSB_POS(x) ((x) << 8) +#define TDMOUT_CTRL1_SEL_SHIFT 24 +#define TDMOUT_CTRL1_GAIN_EN 26 +#define TDMOUT_CTRL1_WS_INV BIT(28) +#define TDMOUT_SWAP 0x08 +#define TDMOUT_MASK0 0x0c +#define TDMOUT_MASK1 0x10 +#define TDMOUT_MASK2 0x14 +#define TDMOUT_MASK3 0x18 +#define TDMOUT_STAT 0x1c +#define TDMOUT_GAIN0 0x20 +#define TDMOUT_GAIN1 0x24 +#define TDMOUT_MUTE_VAL 0x28 +#define TDMOUT_MUTE0 0x2c +#define TDMOUT_MUTE1 0x30 +#define TDMOUT_MUTE2 0x34 +#define TDMOUT_MUTE3 0x38 +#define TDMOUT_MASK_VAL 0x3c + +static const struct regmap_config axg_tdmout_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = TDMOUT_MASK_VAL, +}; + +static const struct snd_kcontrol_new axg_tdmout_controls[] = { + SOC_DOUBLE("Lane 0 Volume", TDMOUT_GAIN0, 0, 8, 255, 0), + SOC_DOUBLE("Lane 1 Volume", TDMOUT_GAIN0, 16, 24, 255, 0), + SOC_DOUBLE("Lane 2 Volume", TDMOUT_GAIN1, 0, 8, 255, 0), + SOC_DOUBLE("Lane 3 Volume", TDMOUT_GAIN1, 16, 24, 255, 0), + SOC_SINGLE("Gain Enable Switch", TDMOUT_CTRL1, + TDMOUT_CTRL1_GAIN_EN, 1, 0), +}; + +static const char * const tdmout_sel_texts[] = { + "IN 0", "IN 1", "IN 2", +}; + +static SOC_ENUM_SINGLE_DECL(axg_tdmout_sel_enum, TDMOUT_CTRL1, + TDMOUT_CTRL1_SEL_SHIFT, tdmout_sel_texts); + +static const struct snd_kcontrol_new axg_tdmout_in_mux = + SOC_DAPM_ENUM("Input Source", axg_tdmout_sel_enum); + +static struct snd_soc_dai * +axg_tdmout_get_be(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dai *be; + + snd_soc_dapm_widget_for_each_sink_path(w, p) { + if (!p->connect) + continue; + + if (p->sink->id == snd_soc_dapm_dai_in) + return (struct snd_soc_dai *)p->sink->priv; + + be = axg_tdmout_get_be(p->sink); + if (be) + return be; + } + + return NULL; +} + +static struct axg_tdm_stream * +axg_tdmout_get_tdm_stream(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dai *be = axg_tdmout_get_be(w); + + if (!be) + return NULL; + + return be->playback_dma_data; +} + +static void axg_tdmout_enable(struct regmap *map) +{ + /* Apply both reset */ + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_RST_OUT | TDMOUT_CTRL0_RST_IN, 0); + + /* Clear out reset before in reset */ + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_RST_OUT, TDMOUT_CTRL0_RST_OUT); + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_RST_IN, TDMOUT_CTRL0_RST_IN); + + /* Actually enable tdmout */ + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_ENABLE, TDMOUT_CTRL0_ENABLE); +} + +static void axg_tdmout_disable(struct regmap *map) +{ + regmap_update_bits(map, TDMOUT_CTRL0, TDMOUT_CTRL0_ENABLE, 0); +} + +static int axg_tdmout_prepare(struct regmap *map, struct axg_tdm_stream *ts) +{ + unsigned int val = 0; + + /* Set the stream skew */ + switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + val |= TDMOUT_CTRL0_INIT_BITNUM(1); + break; + + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_DSP_B: + val |= TDMOUT_CTRL0_INIT_BITNUM(2); + break; + + default: + pr_err("Unsupported format: %u\n", + ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + /* Set the slot width */ + val |= TDMOUT_CTRL0_BITNUM(ts->iface->slot_width - 1); + + /* Set the slot number */ + val |= TDMOUT_CTRL0_SLOTNUM(ts->iface->slots - 1); + + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_INIT_BITNUM_MASK | + TDMOUT_CTRL0_BITNUM_MASK | + TDMOUT_CTRL0_SLOTNUM_MASK, val); + + /* Set the sample width */ + val = TDMOUT_CTRL1_MSB_POS(ts->width - 1); + + /* FIFO data are arranged in chunks of 64bits */ + switch (ts->physical_width) { + case 8: + /* 8 samples of 8 bits */ + val |= TDMOUT_CTRL1_TYPE(0); + break; + case 16: + /* 4 samples of 16 bits - right justified */ + val |= TDMOUT_CTRL1_TYPE(2); + break; + case 32: + /* 2 samples of 32 bits - right justified */ + val |= TDMOUT_CTRL1_TYPE(4); + break; + default: + pr_err("Unsupported physical width: %u\n", + ts->physical_width); + return -EINVAL; + } + + /* If the sample clock is inverted, invert it back for the formatter */ + if (axg_tdm_lrclk_invert(ts->iface->fmt)) + val |= TDMOUT_CTRL1_WS_INV; + + regmap_update_bits(map, TDMOUT_CTRL1, + (TDMOUT_CTRL1_TYPE_MASK | TDMOUT_CTRL1_MSB_POS_MASK | + TDMOUT_CTRL1_WS_INV), val); + + /* Set static swap mask configuration */ + regmap_write(map, TDMOUT_SWAP, 0x76543210); + + return axg_tdm_formatter_set_channel_masks(map, ts, TDMOUT_MASK0); +} + +static const struct snd_soc_dapm_widget axg_tdmout_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_tdmout_in_mux), + SND_SOC_DAPM_PGA_E("ENC", SND_SOC_NOPM, 0, 0, NULL, 0, + axg_tdm_formatter_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD)), + SND_SOC_DAPM_AIF_OUT("OUT", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_tdmout_dapm_routes[] = { + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "ENC", NULL, "SRC SEL" }, + { "OUT", NULL, "ENC" }, +}; + +static const struct snd_soc_component_driver axg_tdmout_component_drv = { + .controls = axg_tdmout_controls, + .num_controls = ARRAY_SIZE(axg_tdmout_controls), + .dapm_widgets = axg_tdmout_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_tdmout_dapm_widgets), + .dapm_routes = axg_tdmout_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_tdmout_dapm_routes), +}; + +static const struct axg_tdm_formatter_ops axg_tdmout_ops = { + .get_stream = axg_tdmout_get_tdm_stream, + .prepare = axg_tdmout_prepare, + .enable = axg_tdmout_enable, + .disable = axg_tdmout_disable, +}; + +static const struct axg_tdm_formatter_driver axg_tdmout_drv = { + .component_drv = &axg_tdmout_component_drv, + .regmap_cfg = &axg_tdmout_regmap_cfg, + .ops = &axg_tdmout_ops, + .invert_sclk = true, +}; + +static const struct of_device_id axg_tdmout_of_match[] = { + { + .compatible = "amlogic,axg-tdmout", + .data = &axg_tdmout_drv, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_tdmout_of_match); + +static struct platform_driver axg_tdmout_pdrv = { + .probe = axg_tdm_formatter_probe, + .driver = { + .name = "axg-tdmout", + .of_match_table = axg_tdmout_of_match, + }, +}; +module_platform_driver(axg_tdmout_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG TDM output formatter driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c new file mode 100644 index 000000000000..c2c9bb312586 --- /dev/null +++ b/sound/soc/meson/axg-toddr.c @@ -0,0 +1,199 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +/* This driver implements the frontend capture DAI of AXG based SoCs */ + +#include <linux/clk.h> +#include <linux/regmap.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "axg-fifo.h" + +#define CTRL0_TODDR_SEL_RESAMPLE BIT(30) +#define CTRL0_TODDR_EXT_SIGNED BIT(29) +#define CTRL0_TODDR_PP_MODE BIT(28) +#define CTRL0_TODDR_TYPE_MASK GENMASK(15, 13) +#define CTRL0_TODDR_TYPE(x) ((x) << 13) +#define CTRL0_TODDR_MSB_POS_MASK GENMASK(12, 8) +#define CTRL0_TODDR_MSB_POS(x) ((x) << 8) +#define CTRL0_TODDR_LSB_POS_MASK GENMASK(7, 3) +#define CTRL0_TODDR_LSB_POS(x) ((x) << 3) + +static int axg_toddr_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + return axg_fifo_pcm_new(rtd, SNDRV_PCM_STREAM_CAPTURE); +} + +static int axg_toddr_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + unsigned int type, width, msb = 31; + + /* + * NOTE: + * Almost all backend will place the MSB at bit 31, except SPDIF Input + * which will put it at index 28. When adding support for the SPDIF + * Input, we'll need to find which type of backend we are connected to. + */ + + switch (params_physical_width(params)) { + case 8: + type = 0; /* 8 samples of 8 bits */ + break; + case 16: + type = 2; /* 4 samples of 16 bits - right justified */ + break; + case 32: + type = 4; /* 2 samples of 32 bits - right justified */ + break; + default: + return -EINVAL; + } + + width = params_width(params); + + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_TODDR_TYPE_MASK | + CTRL0_TODDR_MSB_POS_MASK | + CTRL0_TODDR_LSB_POS_MASK, + CTRL0_TODDR_TYPE(type) | + CTRL0_TODDR_MSB_POS(msb) | + CTRL0_TODDR_LSB_POS(msb - (width - 1))); + + return 0; +} + +static int axg_toddr_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + unsigned int fifo_threshold; + int ret; + + /* Enable pclk to access registers and clock the fifo ip */ + ret = clk_prepare_enable(fifo->pclk); + if (ret) + return ret; + + /* Select orginal data - resampling not supported ATM */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_SEL_RESAMPLE, 0); + + /* Only signed format are supported ATM */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_EXT_SIGNED, + CTRL0_TODDR_EXT_SIGNED); + + /* Apply single buffer mode to the interface */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_PP_MODE, 0); + + /* TODDR does not have a configurable fifo depth */ + fifo_threshold = AXG_FIFO_MIN_CNT - 1; + regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_THRESHOLD_MASK, + CTRL1_THRESHOLD(fifo_threshold)); + + return 0; +} + +static void axg_toddr_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(fifo->pclk); +} + +static const struct snd_soc_dai_ops axg_toddr_ops = { + .hw_params = axg_toddr_dai_hw_params, + .startup = axg_toddr_dai_startup, + .shutdown = axg_toddr_dai_shutdown, +}; + +static struct snd_soc_dai_driver axg_toddr_dai_drv = { + .name = "TODDR", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = AXG_FIFO_CH_MAX, + .rates = AXG_FIFO_RATES, + .formats = AXG_FIFO_FORMATS, + }, + .ops = &axg_toddr_ops, + .pcm_new = axg_toddr_pcm_new, +}; + +static const char * const axg_toddr_sel_texts[] = { + "IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 6" +}; + +static const unsigned int axg_toddr_sel_values[] = { + 0, 1, 2, 3, 4, 6 +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(axg_toddr_sel_enum, FIFO_CTRL0, + CTRL0_SEL_SHIFT, CTRL0_SEL_MASK, + axg_toddr_sel_texts, axg_toddr_sel_values); + +static const struct snd_kcontrol_new axg_toddr_in_mux = + SOC_DAPM_ENUM("Input Source", axg_toddr_sel_enum); + +static const struct snd_soc_dapm_widget axg_toddr_dapm_widgets[] = { + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_toddr_in_mux), + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 6", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_toddr_dapm_routes[] = { + { "Capture", NULL, "SRC SEL" }, + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "SRC SEL", "IN 3", "IN 3" }, + { "SRC SEL", "IN 4", "IN 4" }, + { "SRC SEL", "IN 6", "IN 6" }, +}; + +static const struct snd_soc_component_driver axg_toddr_component_drv = { + .dapm_widgets = axg_toddr_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_toddr_dapm_widgets), + .dapm_routes = axg_toddr_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_toddr_dapm_routes), + .ops = &axg_fifo_pcm_ops +}; + +static const struct axg_fifo_match_data axg_toddr_match_data = { + .component_drv = &axg_toddr_component_drv, + .dai_drv = &axg_toddr_dai_drv +}; + +static const struct of_device_id axg_toddr_of_match[] = { + { + .compatible = "amlogic,axg-toddr", + .data = &axg_toddr_match_data, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_toddr_of_match); + +static struct platform_driver axg_toddr_pdrv = { + .probe = axg_fifo_probe, + .driver = { + .name = "axg-toddr", + .of_match_table = axg_toddr_of_match, + }, +}; +module_platform_driver(axg_toddr_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG capture fifo driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 15ccbf479c96..d5ae9eb8c756 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -40,7 +40,7 @@ struct abe_twl6040 { int mclk_freq; /* MCLK frequency speed for twl6040 */ }; -struct platform_device *dmic_codec_dev; +static struct platform_device *dmic_codec_dev; static int omap_abe_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 51dd7c65096b..fe966272bd0c 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -213,8 +213,10 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream, switch (channels) { case 6: dmic->ch_enabled |= OMAP_DMIC_UP3_ENABLE; + /* fall through */ case 4: dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE; + /* fall through */ case 2: dmic->ch_enabled |= OMAP_DMIC_UP1_ENABLE; break; diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 0e97360f9890..4c1be36c2207 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -310,15 +310,19 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, /* up to 3 channels for capture */ return -EINVAL; link_mask |= 1 << 4; + /* fall through */ case 4: if (stream == SNDRV_PCM_STREAM_CAPTURE) /* up to 3 channels for capture */ return -EINVAL; link_mask |= 1 << 3; + /* fall through */ case 3: link_mask |= 1 << 2; + /* fall through */ case 2: link_mask |= 1 << 1; + /* fall through */ case 1: link_mask |= 1 << 0; break; diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 960744e46edc..776e148b0aa2 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -24,15 +24,19 @@ config SND_PXA2XX_AC97 config SND_PXA2XX_SOC_AC97 tristate select AC97_BUS + select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 select SND_SOC_AC97_BUS config SND_PXA2XX_SOC_I2S + select SND_PXA2XX_LIB tristate config SND_PXA_SOC_SSP - tristate + tristate "Soc Audio via PXA2xx/PXA3xx SSP ports" + depends on PLAT_PXA select PXA_SSP + select SND_PXA2XX_LIB config SND_MMP_SOC_SSPA tristate diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 2fc012b06c43..935a248e5bf6 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -90,95 +90,9 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int acps, acds, width; - unsigned int div4 = PXA_SSP_CLK_SCDB_4; + unsigned int width; int ret = 0; - width = snd_pcm_format_physical_width(params_format(params)); - - /* - * rate = SSPSCLK / (2 * width(16 or 32)) - * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1) - */ - switch (params_rate(params)) { - case 8000: - /* off by a factor of 2: bug in the PXA27x audio clock? */ - acps = 32842000; - switch (width) { - case 16: - /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_16; - break; - default: /* 32 */ - /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_8; - } - break; - case 11025: - acps = 5622000; - switch (width) { - case 16: - /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_4; - break; - default: /* 32 */ - /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - } - break; - case 22050: - acps = 5622000; - switch (width) { - case 16: - /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 44100: - acps = 5622000; - switch (width) { - case 16: - /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 48000: - acps = 12235000; - switch (width) { - case 16: - /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 96000: - default: - acps = 12235000; - switch (width) { - case 16: - /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - break; - default: /* 32 */ - /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - div4 = PXA_SSP_CLK_SCDB_1; - break; - } - break; - } - /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -191,6 +105,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + width = snd_pcm_format_physical_width(params_format(params)); ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; @@ -201,23 +116,6 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* set the SSP audio system clock ACDS divider */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_ACDS, acds); - if (ret < 0) - return ret; - - /* set the SSP audio system clock SCDB divider4 */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_SCDB, div4); - if (ret < 0) - return ret; - - /* set SSP audio pll clock */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps); - if (ret < 0) - return ret; - return 0; } diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 6fc986080130..69033e1a84e6 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -34,7 +34,6 @@ #include <sound/pxa2xx-lib.h> #include <sound/dmaengine_pcm.h> -#include "../../arm/pxa2xx-pcm.h" #include "pxa-ssp.h" /* @@ -42,6 +41,8 @@ */ struct ssp_priv { struct ssp_device *ssp; + struct clk *extclk; + unsigned long ssp_clk; unsigned int sysclk; unsigned int dai_fmt; unsigned int configured_dai_fmt; @@ -105,9 +106,8 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (!dma) return -ENOMEM; - - dma->filter_data = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - &ssp->drcmr_tx : &ssp->drcmr_rx; + dma->chan_name = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "tx" : "rx"; snd_soc_dai_set_dma_data(cpu_dai, substream, dma); @@ -194,21 +194,6 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) pxa_ssp_write_reg(ssp, SSCR0, sscr0); } -/** - * pxa_ssp_get_clkdiv - get SSP clock divider - */ -static u32 pxa_ssp_get_scr(struct ssp_device *ssp) -{ - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - u32 div; - - if (ssp->type == PXA25x_SSP) - div = ((sscr0 >> 8) & 0xff) * 2 + 2; - else - div = ((sscr0 >> 8) & 0xfff) + 1; - return div; -} - /* * Set the SSP ports SYSCLK. */ @@ -221,6 +206,21 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + if (priv->extclk) { + int ret; + + /* + * For DT based boards, if an extclk is given, use it + * here and configure PXA_SSP_CLK_EXT. + */ + + ret = clk_set_rate(priv->extclk, freq); + if (ret < 0) + return ret; + + clk_id = PXA_SSP_CLK_EXT; + } + dev_dbg(&ssp->pdev->dev, "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", cpu_dai->id, clk_id, freq); @@ -265,66 +265,17 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, } /* - * Set the SSP clock dividers. - */ -static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); - struct ssp_device *ssp = priv->ssp; - int val; - - switch (div_id) { - case PXA_SSP_AUDIO_DIV_ACDS: - val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); - pxa_ssp_write_reg(ssp, SSACD, val); - break; - case PXA_SSP_AUDIO_DIV_SCDB: - val = pxa_ssp_read_reg(ssp, SSACD); - val &= ~SSACD_SCDB; - if (ssp->type == PXA3xx_SSP) - val &= ~SSACD_SCDX8; - switch (div) { - case PXA_SSP_CLK_SCDB_1: - val |= SSACD_SCDB; - break; - case PXA_SSP_CLK_SCDB_4: - break; - case PXA_SSP_CLK_SCDB_8: - if (ssp->type == PXA3xx_SSP) - val |= SSACD_SCDX8; - else - return -EINVAL; - break; - default: - return -EINVAL; - } - pxa_ssp_write_reg(ssp, SSACD, val); - break; - case PXA_SSP_DIV_SCR: - pxa_ssp_set_scr(ssp, div); - break; - default: - return -ENODEV; - } - - return 0; -} - -/* * Configure the PLL frequency pxa27x and (afaik - pxa320 only) */ -static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) +static int pxa_ssp_set_pll(struct ssp_priv *priv, unsigned int freq) { - struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; if (ssp->type == PXA3xx_SSP) pxa_ssp_write_reg(ssp, SSACDD, 0); - switch (freq_out) { + switch (freq) { case 5622000: break; case 11345000: @@ -355,7 +306,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, u64 tmp = 19968; tmp *= 1000000; - do_div(tmp, freq_out); + do_div(tmp, freq); val = tmp; val = (val << 16) | 64; @@ -365,7 +316,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, dev_dbg(&ssp->pdev->dev, "Using SSACDD %x to supply %uHz\n", - val, freq_out); + val, freq); break; } @@ -535,6 +486,7 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; + /* fall through */ case SND_SOC_DAIFMT_DSP_B: sscr0 |= SSCR0_MOD | SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; @@ -570,6 +522,24 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) return 0; } +struct pxa_ssp_clock_mode { + int rate; + int pll; + u8 acds; + u8 scdb; +}; + +static const struct pxa_ssp_clock_mode pxa_ssp_clock_modes[] = { + { .rate = 8000, .pll = 32842000, .acds = SSACD_ACDS_32, .scdb = SSACD_SCDB_4X }, + { .rate = 11025, .pll = 5622000, .acds = SSACD_ACDS_4, .scdb = SSACD_SCDB_4X }, + { .rate = 16000, .pll = 32842000, .acds = SSACD_ACDS_16, .scdb = SSACD_SCDB_4X }, + { .rate = 22050, .pll = 5622000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 44100, .pll = 11345000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 48000, .pll = 12235000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 96000, .pll = 12235000, .acds = SSACD_ACDS_4, .scdb = SSACD_SCDB_1X }, + {} +}; + /* * Set the SSP audio DMA parameters and sample size. * Can be called multiple times by oss emulation. @@ -581,11 +551,12 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; int chn = params_channels(params); - u32 sscr0; - u32 sspsp; + u32 sscr0, sspsp; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; struct snd_dmaengine_dai_dma_data *dma_data; + int rate = params_rate(params); + int bclk = rate * chn * (width / 8); int ret; dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); @@ -625,11 +596,57 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, } pxa_ssp_write_reg(ssp, SSCR0, sscr0); + if (sscr0 & SSCR0_ACS) { + ret = pxa_ssp_set_pll(priv, bclk); + + /* + * If we were able to generate the bclk directly, + * all is fine. Otherwise, look up the closest rate + * from the table and also set the dividers. + */ + + if (ret < 0) { + const struct pxa_ssp_clock_mode *m; + int ssacd, acds; + + for (m = pxa_ssp_clock_modes; m->rate; m++) { + if (m->rate == rate) + break; + } + + if (!m->rate) + return -EINVAL; + + acds = m->acds; + + /* The values in the table are for 16 bits */ + if (width == 32) + acds--; + + ret = pxa_ssp_set_pll(priv, bclk); + if (ret < 0) + return ret; + + ssacd = pxa_ssp_read_reg(ssp, SSACD); + ssacd &= ~(SSACD_ACDS(7) | SSACD_SCDB_1X); + ssacd |= SSACD_ACDS(m->acds); + ssacd |= m->scdb; + pxa_ssp_write_reg(ssp, SSACD, ssacd); + } + } else if (sscr0 & SSCR0_ECS) { + /* + * For setups with external clocking, the PLL and its diviers + * are not active. Instead, the SCR bits in SSCR0 can be used + * to divide the clock. + */ + pxa_ssp_set_scr(ssp, bclk / rate); + } + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: sspsp = pxa_ssp_read_reg(ssp, SSPSP); - if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { + if (((priv->sysclk / bclk) == 64) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. @@ -773,6 +790,15 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) ret = -ENODEV; goto err_priv; } + + priv->extclk = devm_clk_get(dev, "extclk"); + if (IS_ERR(priv->extclk)) { + ret = PTR_ERR(priv->extclk); + if (ret == -EPROBE_DEFER) + return ret; + + priv->extclk = NULL; + } } else { priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); if (priv->ssp == NULL) { @@ -814,8 +840,6 @@ static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { .trigger = pxa_ssp_trigger, .hw_params = pxa_ssp_hw_params, .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, .set_fmt = pxa_ssp_set_dai_fmt, .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, .set_tristate = pxa_ssp_set_dai_tristate, @@ -843,6 +867,9 @@ static struct snd_soc_dai_driver pxa_ssp_dai = { static const struct snd_soc_component_driver pxa_ssp_component = { .name = "pxa-ssp", + .ops = &pxa2xx_pcm_ops, + .pcm_new = pxa2xx_soc_pcm_new, + .pcm_free = pxa2xx_pcm_free_dma_buffers, }; #ifdef CONFIG_OF diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 803818aabee9..9f779657bc86 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -68,61 +68,39 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, -}; - static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_in", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_out", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 10, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_out", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 9, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_in", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 8, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .addr = __PREG(MCDR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mic_mono", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream, @@ -236,7 +214,21 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { static const struct snd_soc_component_driver pxa_ac97_component = { .name = "pxa-ac97", + .ops = &pxa2xx_pcm_ops, + .pcm_new = pxa2xx_soc_pcm_new, + .pcm_free = pxa2xx_pcm_free_dma_buffers, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pxa2xx_ac97_dt_ids[] = { + { .compatible = "marvell,pxa250-ac97", }, + { .compatible = "marvell,pxa270-ac97", }, + { .compatible = "marvell,pxa300-ac97", }, + { } }; +MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids); + +#endif static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { @@ -296,6 +288,7 @@ static struct platform_driver pxa2xx_ac97_driver = { #ifdef CONFIG_PM_SLEEP .pm = &pxa2xx_ac97_pm_ops, #endif + .of_match_table = of_match_ptr(pxa2xx_ac97_dt_ids), }, }; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 3fb60baf6eab..42820121e5b9 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -82,20 +82,18 @@ static struct pxa_i2s_port pxa_i2s; static struct clk *clk_i2s; static int clk_ena = 0; -static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "tx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_out_req, }; -static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "rx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_in_req, }; static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, @@ -366,6 +364,9 @@ static struct snd_soc_dai_driver pxa_i2s_dai = { static const struct snd_soc_component_driver pxa_i2s_component = { .name = "pxa-i2s", + .ops = &pxa2xx_pcm_ops, + .pcm_new = pxa2xx_soc_pcm_new, + .pcm_free = pxa2xx_pcm_free_dma_buffers, }; static int pxa2xx_i2s_drv_probe(struct platform_device *pdev) diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 8b6a70e94c01..72eaaef1b426 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -20,70 +20,6 @@ #include <sound/pxa2xx-lib.h> #include <sound/dmaengine_pcm.h> -#include "../../arm/pxa2xx-pcm.h" - -static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_dmaengine_dai_dma_data *dma; - - dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; - - return __pxa2xx_pcm_hw_params(substream, params); -} - -static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) -{ - __pxa2xx_pcm_hw_free(substream); - - return 0; -} - -static const struct snd_pcm_ops pxa2xx_pcm_ops = { - .open = __pxa2xx_pcm_open, - .close = __pxa2xx_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = pxa2xx_pcm_hw_params, - .hw_free = pxa2xx_pcm_hw_free, - .prepare = __pxa2xx_pcm_prepare, - .trigger = pxa2xx_pcm_trigger, - .pointer = pxa2xx_pcm_pointer, - .mmap = pxa2xx_pcm_mmap, -}; - -static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - static const struct snd_soc_component_driver pxa2xx_soc_platform = { .ops = &pxa2xx_pcm_ops, .pcm_new = pxa2xx_soc_pcm_new, @@ -96,18 +32,9 @@ static int pxa2xx_soc_platform_probe(struct platform_device *pdev) NULL, 0); } -#ifdef CONFIG_OF -static const struct of_device_id snd_soc_pxa_audio_match[] = { - { .compatible = "mrvl,pxa-pcm-audio" }, - { } -}; -MODULE_DEVICE_TABLE(of, snd_soc_pxa_audio_match); -#endif - static struct platform_driver pxa_pcm_driver = { .driver = { .name = "pxa-pcm-audio", - .of_match_table = of_match_ptr(snd_soc_pxa_audio_match), }, .probe = pxa2xx_soc_platform_probe, diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ba468e560dd2..230eee450f45 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -83,11 +83,9 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int pll_out = 0; unsigned int wm9713_div = 0; int ret = 0; int rate = params_rate(params); - int width = snd_pcm_format_physical_width(params_format(params)); /* Only support ratios that we can generate neatly from the AC97 * based master clock - in particular, this excludes 44.1kHz. @@ -109,17 +107,10 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Add 1 to the width for the leading clock cycle */ - pll_out = rate * (width + 1) * 8; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out); - if (ret < 0) - return ret; - if (clk_pout) ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV, WM9713_PCMDIV(wm9713_div)); diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 87838fa27997..2a4c912d1e48 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -41,6 +41,9 @@ config SND_SOC_APQ8016_SBC APQ8016 SOC-based systems. Say Y if you want to use audio devices on MI2S. +config SND_SOC_QCOM_COMMON + tristate + config SND_SOC_QDSP6_COMMON tristate @@ -86,7 +89,18 @@ config SND_SOC_MSM8996 tristate "SoC Machine driver for MSM8996 and APQ8096 boards" depends on QCOM_APR select SND_SOC_QDSP6 + select SND_SOC_QCOM_COMMON help Support for Qualcomm Technologies LPASS audio block in APQ8096 SoC-based systems. Say Y if you want to use audio device on this SoCs + +config SND_SOC_SDM845 + tristate "SoC Machine driver for SDM845 boards" + depends on QCOM_APR + select SND_SOC_QDSP6 + select SND_SOC_QCOM_COMMON + help + To add support for audio on Qualcomm Technologies Inc. + SDM845 SoC-based systems. + Say Y if you want to use audio device on this SoCs. diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index 206945bb9ba1..41b2c7a23a4d 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -14,10 +14,14 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o snd-soc-storm-objs := storm.o snd-soc-apq8016-sbc-objs := apq8016_sbc.o snd-soc-apq8096-objs := apq8096.o +snd-soc-sdm845-objs := sdm845.o +snd-soc-qcom-common-objs := common.o obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-apq8096.o +obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o +obj-$(CONFIG_SND_SOC_QCOM_COMMON) += snd-soc-qcom-common.o #DSP lib obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/ diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 561cd429e6f2..1543e85629f8 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -1,14 +1,13 @@ // SPDX-License-Identifier: GPL-2.0 // Copyright (c) 2018, Linaro Limited -#include <linux/soc/qcom/apr.h> #include <linux/module.h> -#include <linux/component.h> #include <linux/platform_device.h> #include <linux/of_device.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/pcm.h> +#include "common.h" static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) @@ -24,211 +23,57 @@ static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int apq8096_sbc_parse_of(struct snd_soc_card *card) +static void apq8096_add_be_ops(struct snd_soc_card *card) { - struct device_node *np; - struct device_node *codec = NULL; - struct device_node *platform = NULL; - struct device_node *cpu = NULL; - struct device *dev = card->dev; - struct snd_soc_dai_link *link; - int ret, num_links; - - ret = snd_soc_of_parse_card_name(card, "qcom,model"); - if (ret) { - dev_err(dev, "Error parsing card name: %d\n", ret); - return ret; - } - - /* DAPM routes */ - if (of_property_read_bool(dev->of_node, "qcom,audio-routing")) { - ret = snd_soc_of_parse_audio_routing(card, - "qcom,audio-routing"); - if (ret) - return ret; - } - - /* Populate links */ - num_links = of_get_child_count(dev->of_node); + struct snd_soc_dai_link *link = card->dai_link; + int i, num_links = card->num_links; - /* Allocate the DAI link array */ - card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL); - if (!card->dai_link) - return -ENOMEM; - - card->num_links = num_links; - link = card->dai_link; - - for_each_child_of_node(dev->of_node, np) { - cpu = of_get_child_by_name(np, "cpu"); - if (!cpu) { - dev_err(dev, "Can't find cpu DT node\n"); - ret = -EINVAL; - goto err; - } - - link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); - if (!link->cpu_of_node) { - dev_err(card->dev, "error getting cpu phandle\n"); - ret = -EINVAL; - goto err; - } - - ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); - if (ret) { - dev_err(card->dev, "error getting cpu dai name\n"); - goto err; - } - - platform = of_get_child_by_name(np, "platform"); - codec = of_get_child_by_name(np, "codec"); - if (codec && platform) { - link->platform_of_node = of_parse_phandle(platform, - "sound-dai", - 0); - if (!link->platform_of_node) { - dev_err(card->dev, "platform dai not found\n"); - ret = -EINVAL; - goto err; - } - - ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); - if (ret < 0) { - dev_err(card->dev, "codec dai not found\n"); - goto err; - } - link->no_pcm = 1; - link->ignore_pmdown_time = 1; + for (i = 0; i < num_links; i++) { + if (link->no_pcm == 1) link->be_hw_params_fixup = apq8096_be_hw_params_fixup; - } else { - link->platform_of_node = link->cpu_of_node; - link->codec_dai_name = "snd-soc-dummy-dai"; - link->codec_name = "snd-soc-dummy"; - link->dynamic = 1; - } - - link->ignore_suspend = 1; - ret = of_property_read_string(np, "link-name", &link->name); - if (ret) { - dev_err(card->dev, "error getting codec dai_link name\n"); - goto err; - } - - link->dpcm_playback = 1; - link->dpcm_capture = 1; - link->stream_name = link->name; link++; } - - return 0; -err: - of_node_put(cpu); - of_node_put(codec); - of_node_put(platform); - kfree(card->dai_link); - return ret; } -static int apq8096_bind(struct device *dev) +static int apq8096_platform_probe(struct platform_device *pdev) { struct snd_soc_card *card; + struct device *dev = &pdev->dev; int ret; card = kzalloc(sizeof(*card), GFP_KERNEL); if (!card) return -ENOMEM; - component_bind_all(dev, card); card->dev = dev; - ret = apq8096_sbc_parse_of(card); + dev_set_drvdata(dev, card); + ret = qcom_snd_parse_of(card); if (ret) { dev_err(dev, "Error parsing OF data\n"); goto err; } + apq8096_add_be_ops(card); ret = snd_soc_register_card(card); if (ret) - goto err; + goto err_card_register; return 0; +err_card_register: + kfree(card->dai_link); err: - component_unbind_all(dev, card); kfree(card); return ret; } -static void apq8096_unbind(struct device *dev) +static int apq8096_platform_remove(struct platform_device *pdev) { - struct snd_soc_card *card = dev_get_drvdata(dev); + struct snd_soc_card *card = dev_get_drvdata(&pdev->dev); snd_soc_unregister_card(card); - component_unbind_all(dev, card); kfree(card->dai_link); kfree(card); -} - -static const struct component_master_ops apq8096_ops = { - .bind = apq8096_bind, - .unbind = apq8096_unbind, -}; - -static int apq8016_compare_of(struct device *dev, void *data) -{ - return dev->of_node == data; -} - -static void apq8016_release_of(struct device *dev, void *data) -{ - of_node_put(data); -} - -static int add_audio_components(struct device *dev, - struct component_match **matchptr) -{ - struct device_node *np, *platform, *cpu, *node, *dai_node; - - node = dev->of_node; - - for_each_child_of_node(node, np) { - cpu = of_get_child_by_name(np, "cpu"); - if (cpu) { - dai_node = of_parse_phandle(cpu, "sound-dai", 0); - of_node_get(dai_node); - component_match_add_release(dev, matchptr, - apq8016_release_of, - apq8016_compare_of, - dai_node); - } - - platform = of_get_child_by_name(np, "platform"); - if (platform) { - dai_node = of_parse_phandle(platform, "sound-dai", 0); - component_match_add_release(dev, matchptr, - apq8016_release_of, - apq8016_compare_of, - dai_node); - } - } - - return 0; -} - -static int apq8096_platform_probe(struct platform_device *pdev) -{ - struct component_match *match = NULL; - int ret; - - ret = add_audio_components(&pdev->dev, &match); - if (ret) - return ret; - - return component_master_add_with_match(&pdev->dev, &apq8096_ops, match); -} - -static int apq8096_platform_remove(struct platform_device *pdev) -{ - component_master_del(&pdev->dev, &apq8096_ops); return 0; } @@ -245,7 +90,6 @@ static struct platform_driver msm_snd_apq8096_driver = { .remove = apq8096_platform_remove, .driver = { .name = "msm-snd-apq8096", - .owner = THIS_MODULE, .of_match_table = msm_snd_apq8096_dt_match, }, }; diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c new file mode 100644 index 000000000000..eb1b9da05dd4 --- /dev/null +++ b/sound/soc/qcom/common.c @@ -0,0 +1,112 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2018, Linaro Limited. +// Copyright (c) 2018, The Linux Foundation. All rights reserved. + +#include <linux/module.h> +#include "common.h" + +int qcom_snd_parse_of(struct snd_soc_card *card) +{ + struct device_node *np; + struct device_node *codec = NULL; + struct device_node *platform = NULL; + struct device_node *cpu = NULL; + struct device *dev = card->dev; + struct snd_soc_dai_link *link; + int ret, num_links; + + ret = snd_soc_of_parse_card_name(card, "model"); + if (ret) { + dev_err(dev, "Error parsing card name: %d\n", ret); + return ret; + } + + /* DAPM routes */ + if (of_property_read_bool(dev->of_node, "audio-routing")) { + ret = snd_soc_of_parse_audio_routing(card, + "audio-routing"); + if (ret) + return ret; + } + + /* Populate links */ + num_links = of_get_child_count(dev->of_node); + + /* Allocate the DAI link array */ + card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL); + if (!card->dai_link) + return -ENOMEM; + + card->num_links = num_links; + link = card->dai_link; + for_each_child_of_node(dev->of_node, np) { + cpu = of_get_child_by_name(np, "cpu"); + if (!cpu) { + dev_err(dev, "Can't find cpu DT node\n"); + ret = -EINVAL; + goto err; + } + + link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); + if (!link->cpu_of_node) { + dev_err(card->dev, "error getting cpu phandle\n"); + ret = -EINVAL; + goto err; + } + + ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); + if (ret) { + dev_err(card->dev, "error getting cpu dai name\n"); + goto err; + } + + platform = of_get_child_by_name(np, "platform"); + codec = of_get_child_by_name(np, "codec"); + if (codec && platform) { + link->platform_of_node = of_parse_phandle(platform, + "sound-dai", + 0); + if (!link->platform_of_node) { + dev_err(card->dev, "platform dai not found\n"); + ret = -EINVAL; + goto err; + } + + ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); + if (ret < 0) { + dev_err(card->dev, "codec dai not found\n"); + goto err; + } + link->no_pcm = 1; + link->ignore_pmdown_time = 1; + } else { + link->platform_of_node = link->cpu_of_node; + link->codec_dai_name = "snd-soc-dummy-dai"; + link->codec_name = "snd-soc-dummy"; + link->dynamic = 1; + } + + link->ignore_suspend = 1; + ret = of_property_read_string(np, "link-name", &link->name); + if (ret) { + dev_err(card->dev, "error getting codec dai_link name\n"); + goto err; + } + + link->dpcm_playback = 1; + link->dpcm_capture = 1; + link->stream_name = link->name; + link++; + } + + return 0; +err: + of_node_put(cpu); + of_node_put(codec); + of_node_put(platform); + kfree(card->dai_link); + return ret; +} +EXPORT_SYMBOL(qcom_snd_parse_of); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/qcom/common.h b/sound/soc/qcom/common.h new file mode 100644 index 000000000000..f05c05b12bd7 --- /dev/null +++ b/sound/soc/qcom/common.h @@ -0,0 +1,11 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// Copyright (c) 2018, The Linux Foundation. All rights reserved. + +#ifndef __QCOM_SND_COMMON_H__ +#define __QCOM_SND_COMMON_H__ + +#include <sound/soc.h> + +int qcom_snd_parse_of(struct snd_soc_card *card); + +#endif diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 31fe78aa207f..d07271ea4c45 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -458,7 +458,7 @@ static irqreturn_t lpass_dma_interrupt_handler( return IRQ_NONE; } dev_warn(soc_runtime->dev, "xrun warning\n"); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(substream); ret = IRQ_HANDLED; } diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 9983c665a941..932c3ebfd252 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -64,7 +64,6 @@ struct q6adm { struct aprv2_ibasic_rsp_result_t result; struct mutex lock; wait_queue_head_t matrix_map_wait; - struct platform_device *pdev_routing; }; struct q6adm_cmd_device_open_v5 { @@ -588,7 +587,6 @@ EXPORT_SYMBOL_GPL(q6adm_close); static int q6adm_probe(struct apr_device *adev) { struct device *dev = &adev->dev; - struct device_node *dais_np; struct q6adm *adm; adm = devm_kzalloc(&adev->dev, sizeof(*adm), GFP_KERNEL); @@ -605,22 +603,12 @@ static int q6adm_probe(struct apr_device *adev) INIT_LIST_HEAD(&adm->copps_list); spin_lock_init(&adm->copps_list_lock); - dais_np = of_get_child_by_name(dev->of_node, "routing"); - if (dais_np) { - adm->pdev_routing = of_platform_device_create(dais_np, - "q6routing", dev); - of_node_put(dais_np); - } - - return 0; + return of_platform_populate(dev->of_node, NULL, NULL, dev); } static int q6adm_remove(struct apr_device *adev) { - struct q6adm *adm = dev_get_drvdata(&adev->dev); - - if (adm->pdev_routing) - of_platform_device_destroy(&adm->pdev_routing->dev, NULL); + of_platform_depopulate(&adev->dev); return 0; } diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 5002dd05bf27..60ff4a2d3577 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -4,7 +4,6 @@ #include <linux/err.h> #include <linux/init.h> -#include <linux/component.h> #include <linux/module.h> #include <linux/device.h> #include <linux/platform_device.h> @@ -81,7 +80,6 @@ static int q6slim_hw_params(struct snd_pcm_substream *substream, struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); struct q6afe_slim_cfg *slim = &dai_data->port_config[dai->id].slim; - slim->num_channels = params_channels(params); slim->sample_rate = params_rate(params); switch (params_format(params)) { @@ -315,6 +313,9 @@ static void q6afe_dai_shutdown(struct snd_pcm_substream *substream, struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); int rc; + if (!dai_data->is_port_started[dai->id]) + return; + rc = q6afe_port_stop(dai_data->port[dai->id]); if (rc < 0) dev_err(dai->dev, "fail to close AFE port (%d)\n", rc); @@ -382,23 +383,31 @@ static int q6slim_set_channel_map(struct snd_soc_dai *dai, struct q6afe_port_config *pcfg = &dai_data->port_config[dai->id]; int i; - if (!rx_slot) { - pr_err("%s: rx slot not found\n", __func__); - return -EINVAL; - } + if (dai->id & 0x1) { + /* TX */ + if (!tx_slot) { + pr_err("%s: tx slot not found\n", __func__); + return -EINVAL; + } - for (i = 0; i < rx_num; i++) { - pcfg->slim.ch_mapping[i] = rx_slot[i]; - pr_debug("%s: find number of channels[%d] ch[%d]\n", - __func__, i, rx_slot[i]); - } + for (i = 0; i < tx_num; i++) + pcfg->slim.ch_mapping[i] = tx_slot[i]; + + pcfg->slim.num_channels = tx_num; + + + } else { + if (!rx_slot) { + pr_err("%s: rx slot not found\n", __func__); + return -EINVAL; + } - pcfg->slim.num_channels = rx_num; + for (i = 0; i < rx_num; i++) + pcfg->slim.ch_mapping[i] = rx_slot[i]; - pr_debug("%s: SLIMBUS_%d_RX cnt[%d] ch[%d %d]\n", __func__, - (dai->id - SLIMBUS_0_RX) / 2, rx_num, - pcfg->slim.ch_mapping[0], - pcfg->slim.ch_mapping[1]); + pcfg->slim.num_channels = rx_num; + + } return 0; } @@ -443,6 +452,14 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"Slimbus5 Playback", NULL, "SLIMBUS_5_RX"}, {"Slimbus6 Playback", NULL, "SLIMBUS_6_RX"}, + {"SLIMBUS_0_TX", NULL, "Slimbus Capture"}, + {"SLIMBUS_1_TX", NULL, "Slimbus1 Capture"}, + {"SLIMBUS_2_TX", NULL, "Slimbus2 Capture"}, + {"SLIMBUS_3_TX", NULL, "Slimbus3 Capture"}, + {"SLIMBUS_4_TX", NULL, "Slimbus4 Capture"}, + {"SLIMBUS_5_TX", NULL, "Slimbus5 Capture"}, + {"SLIMBUS_6_TX", NULL, "Slimbus6 Capture"}, + {"Primary MI2S Playback", NULL, "PRI_MI2S_RX"}, {"Secondary MI2S Playback", NULL, "SEC_MI2S_RX"}, {"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"}, @@ -637,6 +654,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rate_max = 192000, }, }, { + .name = "SLIMBUS_0_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_0_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, + }, { .playback = { .stream_name = "Slimbus1 Playback", .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | @@ -655,6 +690,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, }, { + .name = "SLIMBUS_1_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_1_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus1 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, + }, { .playback = { .stream_name = "Slimbus2 Playback", .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | @@ -672,6 +725,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_2_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_2_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_2_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus2 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus3 Playback", @@ -690,6 +762,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_3_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_3_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_3_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus3 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus4 Playback", @@ -708,6 +799,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_4_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_4_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_4_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus4 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus5 Playback", @@ -726,6 +836,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_5_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_5_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_5_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus5 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus6 Playback", @@ -744,6 +873,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_6_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_6_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_6_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus6 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Primary MI2S Playback", @@ -972,6 +1120,13 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_RX", "Slimbus4 Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_RX", "Slimbus5 Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_RX", "Slimbus6 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_0_TX", "Slimbus Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_1_TX", "Slimbus1 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_2_TX", "Slimbus2 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_3_TX", "Slimbus3 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_4_TX", "Slimbus4 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_5_TX", "Slimbus5 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_6_TX", "Slimbus6 Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_RX", "Quaternary MI2S Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_MI2S_TX", "Quaternary MI2S Capture", @@ -1180,7 +1335,7 @@ static void of_q6afe_parse_dai_data(struct device *dev, int id, i, num_lines; ret = of_property_read_u32(node, "reg", &id); - if (ret || id > AFE_PORT_MAX) { + if (ret || id < 0 || id >= AFE_PORT_MAX) { dev_err(dev, "valid dai id not found:%d\n", ret); continue; } @@ -1249,11 +1404,12 @@ static void of_q6afe_parse_dai_data(struct device *dev, } } -static int q6afe_dai_bind(struct device *dev, struct device *master, void *data) +static int q6afe_dai_dev_probe(struct platform_device *pdev) { struct q6afe_dai_data *dai_data; + struct device *dev = &pdev->dev; - dai_data = kzalloc(sizeof(*dai_data), GFP_KERNEL); + dai_data = devm_kzalloc(dev, sizeof(*dai_data), GFP_KERNEL); if (!dai_data) return -ENOMEM; @@ -1261,41 +1417,22 @@ static int q6afe_dai_bind(struct device *dev, struct device *master, void *data) of_q6afe_parse_dai_data(dev, dai_data); - return snd_soc_register_component(dev, &q6afe_dai_component, + return devm_snd_soc_register_component(dev, &q6afe_dai_component, q6afe_dais, ARRAY_SIZE(q6afe_dais)); } -static void q6afe_dai_unbind(struct device *dev, struct device *master, - void *data) -{ - struct q6afe_dai_data *dai_data = dev_get_drvdata(dev); - - snd_soc_unregister_component(dev); - kfree(dai_data); -} - -static const struct component_ops q6afe_dai_comp_ops = { - .bind = q6afe_dai_bind, - .unbind = q6afe_dai_unbind, +static const struct of_device_id q6afe_dai_device_id[] = { + { .compatible = "qcom,q6afe-dais" }, + {}, }; - -static int q6afe_dai_dev_probe(struct platform_device *pdev) -{ - return component_add(&pdev->dev, &q6afe_dai_comp_ops); -} - -static int q6afe_dai_dev_remove(struct platform_device *pdev) -{ - component_del(&pdev->dev, &q6afe_dai_comp_ops); - return 0; -} +MODULE_DEVICE_TABLE(of, q6afe_dai_device_id); static struct platform_driver q6afe_dai_platform_driver = { .driver = { .name = "q6afe-dai", + .of_match_table = of_match_ptr(q6afe_dai_device_id), }, .probe = q6afe_dai_dev_probe, - .remove = q6afe_dai_dev_remove, }; module_platform_driver(q6afe_dai_platform_driver); diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 01f43218984b..000775b4bba8 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -316,7 +316,6 @@ struct q6afe { struct mutex lock; struct list_head port_list; spinlock_t port_list_lock; - struct platform_device *pdev_dais; }; struct afe_port_cmd_device_start { @@ -515,6 +514,20 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = { SLIMBUS_5_RX, 1, 1}, [SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX, SLIMBUS_6_RX, 1, 1}, + [SLIMBUS_0_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX, + SLIMBUS_0_TX, 0, 1}, + [SLIMBUS_1_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX, + SLIMBUS_1_TX, 0, 1}, + [SLIMBUS_2_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX, + SLIMBUS_2_TX, 0, 1}, + [SLIMBUS_3_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX, + SLIMBUS_3_TX, 0, 1}, + [SLIMBUS_4_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX, + SLIMBUS_4_TX, 0, 1}, + [SLIMBUS_5_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX, + SLIMBUS_5_TX, 0, 1}, + [SLIMBUS_6_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX, + SLIMBUS_6_TX, 0, 1}, [PRIMARY_MI2S_RX] = { AFE_PORT_ID_PRIMARY_MI2S_RX, PRIMARY_MI2S_RX, 1, 1}, [PRIMARY_MI2S_TX] = { AFE_PORT_ID_PRIMARY_MI2S_TX, @@ -777,7 +790,7 @@ static int q6afe_callback(struct apr_device *adev, struct apr_resp_pkt *data) */ int q6afe_get_port_id(int index) { - if (index < 0 || index > AFE_PORT_MAX) + if (index < 0 || index >= AFE_PORT_MAX) return -EINVAL; return port_maps[index].port_id; @@ -1014,7 +1027,7 @@ int q6afe_port_stop(struct q6afe_port *port) port_id = port->id; index = port->token; - if (index < 0 || index > AFE_PORT_MAX) { + if (index < 0 || index >= AFE_PORT_MAX) { dev_err(afe->dev, "AFE port index[%d] invalid!\n", index); return -EINVAL; } @@ -1355,7 +1368,7 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) unsigned long flags; int cfg_type; - if (id < 0 || id > AFE_PORT_MAX) { + if (id < 0 || id >= AFE_PORT_MAX) { dev_err(dev, "AFE port token[%d] invalid!\n", id); return ERR_PTR(-EINVAL); } @@ -1373,6 +1386,13 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) case AFE_PORT_ID_MULTICHAN_HDMI_RX: cfg_type = AFE_PARAM_ID_HDMI_CONFIG; break; + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX: case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX: case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX: case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX: @@ -1438,7 +1458,6 @@ static int q6afe_probe(struct apr_device *adev) { struct q6afe *afe; struct device *dev = &adev->dev; - struct device_node *dais_np; afe = devm_kzalloc(dev, sizeof(*afe), GFP_KERNEL); if (!afe) @@ -1453,22 +1472,12 @@ static int q6afe_probe(struct apr_device *adev) dev_set_drvdata(dev, afe); - dais_np = of_get_child_by_name(dev->of_node, "dais"); - if (dais_np) { - afe->pdev_dais = of_platform_device_create(dais_np, - "q6afe-dai", dev); - of_node_put(dais_np); - } - - return 0; + return of_platform_populate(dev->of_node, NULL, NULL, dev); } static int q6afe_remove(struct apr_device *adev) { - struct q6afe *afe = dev_get_drvdata(&adev->dev); - - if (afe->pdev_dais) - of_platform_device_destroy(&afe->pdev_dais->dev, NULL); + of_platform_depopulate(&adev->dev); return 0; } diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 349c6a883c63..9db9a2944ef2 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -7,7 +7,6 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> -#include <linux/component.h> #include <sound/soc.h> #include <sound/soc.h> #include <sound/soc-dapm.h> @@ -390,7 +389,9 @@ static int q6asm_dai_close(struct snd_pcm_substream *substream) struct q6asm_dai_rtd *prtd = runtime->private_data; if (prtd->audio_client) { - q6asm_cmd(prtd->audio_client, CMD_CLOSE); + if (prtd->state) + q6asm_cmd(prtd->audio_client, CMD_CLOSE); + q6asm_unmap_memory_regions(substream->stream, prtd->audio_client); q6asm_audio_client_free(prtd->audio_client); @@ -561,14 +562,15 @@ static struct snd_soc_dai_driver q6asm_fe_dais[] = { Q6ASM_FEDAI_DRIVER(8), }; -static int q6asm_dai_bind(struct device *dev, struct device *master, void *data) +static int q6asm_dai_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; struct device_node *node = dev->of_node; struct of_phandle_args args; struct q6asm_dai_data *pdata; int rc; - pdata = kzalloc(sizeof(struct q6asm_dai_data), GFP_KERNEL); + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); if (!pdata) return -ENOMEM; @@ -580,43 +582,23 @@ static int q6asm_dai_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(dev, pdata); - return snd_soc_register_component(dev, &q6asm_fe_dai_component, + return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component, q6asm_fe_dais, ARRAY_SIZE(q6asm_fe_dais)); } -static void q6asm_dai_unbind(struct device *dev, struct device *master, - void *data) -{ - struct q6asm_dai_data *pdata = dev_get_drvdata(dev); - - snd_soc_unregister_component(dev); - - kfree(pdata); - -} -static const struct component_ops q6asm_dai_comp_ops = { - .bind = q6asm_dai_bind, - .unbind = q6asm_dai_unbind, +static const struct of_device_id q6asm_dai_device_id[] = { + { .compatible = "qcom,q6asm-dais" }, + {}, }; - -static int q6asm_dai_probe(struct platform_device *pdev) -{ - return component_add(&pdev->dev, &q6asm_dai_comp_ops); -} - -static int q6asm_dai_dev_remove(struct platform_device *pdev) -{ - component_del(&pdev->dev, &q6asm_dai_comp_ops); - return 0; -} +MODULE_DEVICE_TABLE(of, q6asm_dai_device_id); static struct platform_driver q6asm_dai_platform_driver = { .driver = { .name = "q6asm-dai", + .of_match_table = of_match_ptr(q6asm_dai_device_id), }, .probe = q6asm_dai_probe, - .remove = q6asm_dai_dev_remove, }; module_platform_driver(q6asm_dai_platform_driver); diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 530852385cad..2b2c7233bb5f 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -174,10 +174,8 @@ struct q6asm { struct device *dev; struct q6core_svc_api_info ainfo; wait_queue_head_t mem_wait; - struct platform_device *pcmdev; spinlock_t slock; struct audio_client *session[MAX_SESSIONS + 1]; - struct platform_device *pdev_dais; }; struct audio_client { @@ -1344,7 +1342,6 @@ EXPORT_SYMBOL_GPL(q6asm_cmd_nowait); static int q6asm_probe(struct apr_device *adev) { struct device *dev = &adev->dev; - struct device_node *dais_np; struct q6asm *q6asm; q6asm = devm_kzalloc(dev, sizeof(*q6asm), GFP_KERNEL); @@ -1359,22 +1356,12 @@ static int q6asm_probe(struct apr_device *adev) spin_lock_init(&q6asm->slock); dev_set_drvdata(dev, q6asm); - dais_np = of_get_child_by_name(dev->of_node, "dais"); - if (dais_np) { - q6asm->pdev_dais = of_platform_device_create(dais_np, - "q6asm-dai", dev); - of_node_put(dais_np); - } - - return 0; + return of_platform_populate(dev->of_node, NULL, NULL, dev); } static int q6asm_remove(struct apr_device *adev) { - struct q6asm *q6asm = dev_get_drvdata(&adev->dev); - - if (q6asm->pdev_dais) - of_platform_device_destroy(&q6asm->pdev_dais->dev, NULL); + of_platform_depopulate(&adev->dev); return 0; } diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 593f66b8622f..dc94c5c53788 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -8,7 +8,6 @@ #include <linux/platform_device.h> #include <linux/of_platform.h> #include <linux/bitops.h> -#include <linux/component.h> #include <linux/mutex.h> #include <linux/of_device.h> #include <linux/slab.h> @@ -68,6 +67,13 @@ { mix_name, "SEC_MI2S_TX", "SEC_MI2S_TX" }, \ { mix_name, "QUAT_MI2S_TX", "QUAT_MI2S_TX" }, \ { mix_name, "TERT_MI2S_TX", "TERT_MI2S_TX" }, \ + { mix_name, "SLIMBUS_0_TX", "SLIMBUS_0_TX" }, \ + { mix_name, "SLIMBUS_1_TX", "SLIMBUS_1_TX" }, \ + { mix_name, "SLIMBUS_2_TX", "SLIMBUS_2_TX" }, \ + { mix_name, "SLIMBUS_3_TX", "SLIMBUS_3_TX" }, \ + { mix_name, "SLIMBUS_4_TX", "SLIMBUS_4_TX" }, \ + { mix_name, "SLIMBUS_5_TX", "SLIMBUS_5_TX" }, \ + { mix_name, "SLIMBUS_6_TX", "SLIMBUS_6_TX" }, \ { mix_name, "PRIMARY_TDM_TX_0", "PRIMARY_TDM_TX_0"}, \ { mix_name, "PRIMARY_TDM_TX_1", "PRIMARY_TDM_TX_1"}, \ { mix_name, "PRIMARY_TDM_TX_2", "PRIMARY_TDM_TX_2"}, \ @@ -122,6 +128,27 @@ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_0_TX", SLIMBUS_0_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_1_TX", SLIMBUS_1_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_2_TX", SLIMBUS_2_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_3_TX", SLIMBUS_3_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_4_TX", SLIMBUS_4_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_5_TX", SLIMBUS_5_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_6_TX", SLIMBUS_6_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ SOC_SINGLE_EXT("PRIMARY_TDM_TX_0", PRIMARY_TDM_TX_0, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \ @@ -310,7 +337,7 @@ int q6routing_stream_open(int fedai_id, int perf_mode, session->channels, topology, perf_mode, session->bits_per_sample, 0, 0); - if (!copp) { + if (IS_ERR_OR_NULL(copp)) { mutex_unlock(&routing_data->lock); return -EINVAL; } @@ -899,7 +926,7 @@ static int routing_hw_params(struct snd_pcm_substream *substream, else path_type = ADM_PATH_LIVE_REC; - if (be_id > AFE_MAX_PORTS) + if (be_id >= AFE_MAX_PORTS) return -EINVAL; session = &data->port_data[be_id]; @@ -949,9 +976,10 @@ static const struct snd_soc_component_driver msm_soc_routing_component = { .num_dapm_routes = ARRAY_SIZE(intercon), }; -static int q6routing_dai_bind(struct device *dev, struct device *master, - void *data) +static int q6pcm_routing_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; + routing_data = kzalloc(sizeof(*routing_data), GFP_KERNEL); if (!routing_data) return -ENOMEM; @@ -961,41 +989,28 @@ static int q6routing_dai_bind(struct device *dev, struct device *master, mutex_init(&routing_data->lock); dev_set_drvdata(dev, routing_data); - return snd_soc_register_component(dev, &msm_soc_routing_component, + return devm_snd_soc_register_component(dev, &msm_soc_routing_component, NULL, 0); } -static void q6routing_dai_unbind(struct device *dev, struct device *master, - void *d) +static int q6pcm_routing_remove(struct platform_device *pdev) { - struct msm_routing_data *data = dev_get_drvdata(dev); - - snd_soc_unregister_component(dev); - - kfree(data); - + kfree(routing_data); routing_data = NULL; -} - -static const struct component_ops q6routing_dai_comp_ops = { - .bind = q6routing_dai_bind, - .unbind = q6routing_dai_unbind, -}; -static int q6pcm_routing_probe(struct platform_device *pdev) -{ - return component_add(&pdev->dev, &q6routing_dai_comp_ops); -} - -static int q6pcm_routing_remove(struct platform_device *pdev) -{ - component_del(&pdev->dev, &q6routing_dai_comp_ops); return 0; } +static const struct of_device_id q6pcm_routing_device_id[] = { + { .compatible = "qcom,q6adm-routing" }, + {}, +}; +MODULE_DEVICE_TABLE(of, q6pcm_routing_device_id); + static struct platform_driver q6pcm_routing_platform_driver = { .driver = { .name = "q6routing", + .of_match_table = of_match_ptr(q6pcm_routing_device_id), }, .probe = q6pcm_routing_probe, .remove = q6pcm_routing_remove, diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c new file mode 100644 index 000000000000..2a781d87ee65 --- /dev/null +++ b/sound/soc/qcom/sdm845.c @@ -0,0 +1,285 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2018, The Linux Foundation. All rights reserved. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/of_device.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include "common.h" +#include "qdsp6/q6afe.h" + +#define DEFAULT_SAMPLE_RATE_48K 48000 +#define DEFAULT_MCLK_RATE 24576000 +#define DEFAULT_BCLK_RATE 12288000 + +struct sdm845_snd_data { + struct snd_soc_card *card; + uint32_t pri_mi2s_clk_count; + uint32_t quat_tdm_clk_count; +}; + +static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28}; + +static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret = 0; + int channels, slot_width; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + slot_width = 32; + break; + default: + dev_err(rtd->dev, "%s: invalid param format 0x%x\n", + __func__, params_format(params)); + return -EINVAL; + } + + channels = params_channels(params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0, 0x3, + 8, slot_width); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n", + __func__, ret); + goto end; + } + + ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, + channels, tdm_slot_offset); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n", + __func__, ret); + goto end; + } + } else { + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xf, 0, + 8, slot_width); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n", + __func__, ret); + goto end; + } + + ret = snd_soc_dai_set_channel_map(cpu_dai, channels, + tdm_slot_offset, 0, NULL); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n", + __func__, ret); + goto end; + } + } +end: + return ret; +} + +static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret = 0; + + switch (cpu_dai->id) { + case QUATERNARY_TDM_RX_0: + case QUATERNARY_TDM_TX_0: + ret = sdm845_tdm_snd_hw_params(substream, params); + break; + default: + pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); + break; + } + return ret; +} + +static int sdm845_snd_startup(struct snd_pcm_substream *substream) +{ + unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + case PRIMARY_MI2S_TX: + if (++(data->pri_mi2s_clk_count) == 1) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_MCLK_1, + DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT, + DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + } + snd_soc_dai_set_fmt(cpu_dai, fmt); + break; + + case QUATERNARY_TDM_RX_0: + case QUATERNARY_TDM_TX_0: + if (++(data->quat_tdm_clk_count) == 1) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT, + DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + } + break; + + default: + pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); + break; + } + return 0; +} + +static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + case PRIMARY_MI2S_TX: + if (--(data->pri_mi2s_clk_count) == 0) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_MCLK_1, + 0, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT, + 0, SNDRV_PCM_STREAM_PLAYBACK); + }; + break; + + case QUATERNARY_TDM_RX_0: + case QUATERNARY_TDM_TX_0: + if (--(data->quat_tdm_clk_count) == 0) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT, + 0, SNDRV_PCM_STREAM_PLAYBACK); + } + break; + + default: + pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); + break; + } +} + +static struct snd_soc_ops sdm845_be_ops = { + .hw_params = sdm845_snd_hw_params, + .startup = sdm845_snd_startup, + .shutdown = sdm845_snd_shutdown, +}; + +static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + rate->min = rate->max = DEFAULT_SAMPLE_RATE_48K; + channels->min = channels->max = 2; + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + + return 0; +} + +static void sdm845_add_be_ops(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *link = card->dai_link; + int i, num_links = card->num_links; + + for (i = 0; i < num_links; i++) { + if (link->no_pcm == 1) { + link->ops = &sdm845_be_ops; + link->be_hw_params_fixup = sdm845_be_hw_params_fixup; + } + link++; + } +} + +static int sdm845_snd_platform_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card; + struct sdm845_snd_data *data; + struct device *dev = &pdev->dev; + int ret; + + card = kzalloc(sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + /* Allocate the private data */ + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto data_alloc_fail; + } + + card->dev = dev; + dev_set_drvdata(dev, card); + ret = qcom_snd_parse_of(card); + if (ret) { + dev_err(dev, "Error parsing OF data\n"); + goto parse_dt_fail; + } + + data->card = card; + snd_soc_card_set_drvdata(card, data); + + sdm845_add_be_ops(card); + ret = snd_soc_register_card(card); + if (ret) { + dev_err(dev, "Sound card registration failed\n"); + goto register_card_fail; + } + return ret; + +register_card_fail: + kfree(card->dai_link); +parse_dt_fail: + kfree(data); +data_alloc_fail: + kfree(card); + return ret; +} + +static int sdm845_snd_platform_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = dev_get_drvdata(&pdev->dev); + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + kfree(card->dai_link); + kfree(data); + kfree(card); + return 0; +} + +static const struct of_device_id sdm845_snd_device_id[] = { + { .compatible = "qcom,sdm845-sndcard" }, + {}, +}; +MODULE_DEVICE_TABLE(of, sdm845_snd_device_id); + +static struct platform_driver sdm845_snd_driver = { + .probe = sdm845_snd_platform_probe, + .remove = sdm845_snd_platform_remove, + .driver = { + .name = "msm-snd-sdm845", + .of_match_table = sdm845_snd_device_id, + }, +}; +module_platform_driver(sdm845_snd_driver); + +MODULE_DESCRIPTION("sdm845 ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 05b078e7b87f..65e814d46006 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,10 +1,11 @@ # SPDX-License-Identifier: GPL-2.0 # ROCKCHIP Platform Support snd-soc-rockchip-i2s-objs := rockchip_i2s.o +snd-soc-rockchip-pcm-objs := rockchip_pcm.o snd-soc-rockchip-pdm-objs := rockchip_pdm.o snd-soc-rockchip-spdif-objs := rockchip_spdif.o -obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o snd-soc-rockchip-pcm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 950823d69e9c..60d43d53a8f5 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -22,6 +22,7 @@ #include <sound/dmaengine_pcm.h> #include "rockchip_i2s.h" +#include "rockchip_pcm.h" #define DRV_NAME "rockchip-i2s" @@ -674,7 +675,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_suspend; } - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = rockchip_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); return ret; diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c new file mode 100644 index 000000000000..f77538319221 --- /dev/null +++ b/sound/soc/rockchip/rockchip_pcm.c @@ -0,0 +1,45 @@ +/* + * Copyright (c) 2018 Rockchip Electronics Co. Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/device.h> +#include <linux/init.h> +#include <linux/module.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "rockchip_pcm.h" + +static const struct snd_pcm_hardware snd_rockchip_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 52, + .buffer_bytes_max = 64 * 1024, + .fifo_size = 32, +}; + +static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = { + .pcm_hardware = &snd_rockchip_hardware, + .prealloc_buffer_size = 32 * 1024, +}; + +int rockchip_pcm_platform_register(struct device *dev) +{ + return devm_snd_dmaengine_pcm_register(dev, &rk_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_COMPAT); +} +EXPORT_SYMBOL_GPL(rockchip_pcm_platform_register); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/rockchip_pcm.h b/sound/soc/rockchip/rockchip_pcm.h new file mode 100644 index 000000000000..d6c36115c60a --- /dev/null +++ b/sound/soc/rockchip/rockchip_pcm.h @@ -0,0 +1,14 @@ +/* + * Copyright (c) 2018 Rockchip Electronics Co. Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ROCKCHIP_PCM_H +#define _ROCKCHIP_PCM_H + +int rockchip_pcm_platform_register(struct device *dev); + +#endif diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 4db4fd56db35..881c32498808 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -181,7 +181,8 @@ static int snd_rk_mc_probe(struct platform_device *pdev) if (!rk_dailink.cpu_of_node) { dev_err(&pdev->dev, "Property 'rockchip,i2s-controller' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_codec_of_node; } rk_dailink.platform_of_node = rk_dailink.cpu_of_node; @@ -190,17 +191,36 @@ static int snd_rk_mc_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "Soc parse card name failed %d\n", ret); - return ret; + goto put_cpu_of_node; } ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "Soc register card failed %d\n", ret); - return ret; + goto put_cpu_of_node; } return ret; + +put_cpu_of_node: + of_node_put(rk_dailink.cpu_of_node); + rk_dailink.cpu_of_node = NULL; +put_codec_of_node: + of_node_put(rk_dailink.codec_of_node); + rk_dailink.codec_of_node = NULL; + + return ret; +} + +static int snd_rk_mc_remove(struct platform_device *pdev) +{ + of_node_put(rk_dailink.cpu_of_node); + rk_dailink.cpu_of_node = NULL; + of_node_put(rk_dailink.codec_of_node); + rk_dailink.codec_of_node = NULL; + + return 0; } static const struct of_device_id rockchip_rt5645_of_match[] = { @@ -212,6 +232,7 @@ MODULE_DEVICE_TABLE(of, rockchip_rt5645_of_match); static struct platform_driver snd_rk_mc_driver = { .probe = snd_rk_mc_probe, + .remove = snd_rk_mc_remove, .driver = { .name = DRV_NAME, .pm = &snd_soc_pm_ops, diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index f914ed45db7d..d6c62aa13041 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -710,6 +710,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, switch (params_channels(params)) { case 6: val |= MOD_DC2_EN; + /* fall through */ case 4: val |= MOD_DC1_EN; break; diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 0ae0800bf3a8..dc20f0f7080a 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -1,3 +1,4 @@ +# SPDX-License-Identifier: GPL-2.0 menu "SoC Audio support for Renesas SoCs" depends on SUPERH || ARCH_RENESAS || COMPILE_TEST diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 2dc3b762fdd9..922fb6aa3ed1 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -1,16 +1,14 @@ -/* - * SH7760 ("camelot") DMABRG audio DMA unit support - * - * Copyright (C) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> - * licensed under the terms outlined in the file COPYING at the root - * of the linux kernel sources. - * - * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which - * trigger an interrupt when one half of the programmed transfer size - * has been xmitted. - * - * FIXME: little-endian only for now - */ +// SPDX-License-Identifier: GPL-2.0 +// +// SH7760 ("camelot") DMABRG audio DMA unit support +// +// Copyright (C) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> +// +// The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which +// trigger an interrupt when one half of the programmed transfer size +// has been xmitted. +// +// FIXME: little-endian only for now #include <linux/module.h> #include <linux/gfp.h> @@ -341,6 +339,6 @@ static struct platform_driver sh7760_pcm_driver = { module_platform_driver(sh7760_pcm_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver"); MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3bae06dd121f..aa7e902f0c02 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1,16 +1,12 @@ -/* - * Fifo-attached Serial Interface (FSI) support for SH7724 - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Kuninori Morimoto <morimoto.kuninori@renesas.com> - * - * Based on ssi.c - * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Fifo-attached Serial Interface (FSI) support for SH7724 +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Kuninori Morimoto <morimoto.kuninori@renesas.com> +// +// Based on ssi.c +// Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> #include <linux/delay.h> #include <linux/dma-mapping.h> diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 624aaf569fef..c2b496398e6b 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -1,13 +1,11 @@ -/* - * Hitachi Audio Controller (AC97) support for SH7760/SH7780 - * - * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> - * licensed under the terms outlined in the file COPYING at the root - * of the linux kernel sources. - * - * dont forget to set IPSEL/OMSEL register bits (in your board code) to - * enable HAC output pins! - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Hitachi Audio Controller (AC97) support for SH7760/SH7780 +// +// Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> +// +// dont forget to set IPSEL/OMSEL register bits (in your board code) to +// enable HAC output pins! /* BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only * the FIRST can be used since ASoC does not pass any information to the @@ -343,6 +341,6 @@ static struct platform_driver hac_pcm_driver = { module_platform_driver(hac_pcm_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver"); MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index ecb057ff9fbb..8739c9f60672 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -1,12 +1,8 @@ -/* - * ALSA SoC driver for Migo-R - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC driver for Migo-R +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> #include <linux/clkdev.h> #include <linux/device.h> diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile index 9c3d5aed99d1..5d1ff8ef26f9 100644 --- a/sound/soc/sh/rcar/Makefile +++ b/sound/soc/sh/rcar/Makefile @@ -1,2 +1,3 @@ +# SPDX-License-Identifier: GPL-2.0 snd-soc-rcar-objs := core.o gen.o dma.o adg.o ssi.o ssiu.o src.o ctu.o mix.o dvc.o cmd.o obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 4672688cac32..3a3064dda57f 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -1,12 +1,9 @@ -/* - * Helper routines for R-Car sound ADG. - * - * Copyright (C) 2013 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This file is subject to the terms and conditions of the GNU General Public - * License. See the file "COPYING" in the main directory of this archive - * for more details. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Helper routines for R-Car sound ADG. +// +// Copyright (C) 2013 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #include <linux/clk-provider.h> #include "rsnd.h" diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index 5900fb535a2b..cc191cd5fb82 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -1,13 +1,10 @@ -/* - * Renesas R-Car CMD support - * - * Copyright (C) 2015 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car CMD support +// +// Copyright (C) 2015 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #include "rsnd.h" struct rsnd_cmd { @@ -89,7 +86,7 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, cmd_case[rsnd_mod_id(src)] << 16; } - dev_dbg(dev, "ctu/mix path = 0x%08x", data); + dev_dbg(dev, "ctu/mix path = 0x%08x\n", data); rsnd_mod_write(mod, CMD_ROUTE_SLCT, data); rsnd_mod_write(mod, CMD_BUSIF_MODE, rsnd_get_busif_shift(io, mod) | 1); diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index f237002180c0..f8425d8b44d2 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1,16 +1,12 @@ -/* - * Renesas R-Car SRU/SCU/SSIU/SSI support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * Based on fsi.c - * Kuninori Morimoto <morimoto.kuninori@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SRU/SCU/SSIU/SSI support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> +// +// Based on fsi.c +// Kuninori Morimoto <morimoto.kuninori@renesas.com> /* * Renesas R-Car sound device structure @@ -552,6 +548,15 @@ struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id) return priv->rdai + id; } +static struct snd_soc_dai_driver +*rsnd_daidrv_get(struct rsnd_priv *priv, int id) +{ + if ((id < 0) || (id >= rsnd_rdai_nr(priv))) + return NULL; + + return priv->daidrv + id; +} + #define rsnd_dai_to_priv(dai) snd_soc_dai_get_drvdata(dai) static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai) { @@ -1037,7 +1042,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, int io_i; rdai = rsnd_rdai_get(priv, dai_i); - drv = priv->daidrv + dai_i; + drv = rsnd_daidrv_get(priv, dai_i); io_playback = &rdai->playback; io_capture = &rdai->capture; @@ -1085,6 +1090,12 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, of_node_put(capture); } + if (rsnd_ssi_is_pin_sharing(io_capture) || + rsnd_ssi_is_pin_sharing(io_playback)) { + /* should have symmetric_rates if pin sharing */ + drv->symmetric_rates = 1; + } + dev_dbg(dev, "%s (%s/%s)\n", rdai->name, rsnd_io_to_mod_ssi(io_playback) ? "play" : " -- ", rsnd_io_to_mod_ssi(io_capture) ? "capture" : " -- "); @@ -1606,7 +1617,7 @@ static struct platform_driver rsnd_driver = { }; module_platform_driver(rsnd_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Renesas R-Car audio driver"); MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>"); MODULE_ALIAS("platform:rcar-pcm-audio"); diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 83be7d3ae0a8..6a55aa753003 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -1,12 +1,9 @@ -/* - * ctu.c - * - * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ctu.c +// +// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #include "rsnd.h" #define CTU_NAME_SIZE 16 diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index ef82b94d038b..fe63ef8600d0 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -1,13 +1,10 @@ -/* - * Renesas R-Car Audio DMAC support - * - * Copyright (C) 2015 Renesas Electronics Corp. - * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car Audio DMAC support +// +// Copyright (C) 2015 Renesas Electronics Corp. +// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #include <linux/delay.h> #include <linux/of_dma.h> #include "rsnd.h" diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index ca1780e0b830..2b16e0ce6bc5 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -1,13 +1,9 @@ -/* - * Renesas R-Car DVC support - * - * Copyright (C) 2014 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car DVC support +// +// Copyright (C) 2014 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> /* * Playback Volume diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 25642e92dae0..0230301fe078 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -1,13 +1,9 @@ -/* - * Renesas R-Car Gen1 SRU/SSI support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car Gen1 SRU/SSI support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> /* * #define DEBUG diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 1881b2de9126..8e3b57eaa708 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -1,12 +1,8 @@ -/* - * mix.c - * - * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// mix.c +// +// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> /* * CTUn MIXn diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 6d7280d2d9be..96d93330b1e1 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -1,13 +1,10 @@ -/* - * Renesas R-Car - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #ifndef RSND_H #define RSND_H diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 6c72d1a81cf5..beccfbac7581 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -1,13 +1,9 @@ -/* - * Renesas R-Car SRC support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SRC support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> /* * you can enable below define if you don't need diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 6e1166ec24a0..8304e4ec9242 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1,16 +1,12 @@ -/* - * Renesas R-Car SSIU/SSI support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * Based on fsi.c - * Kuninori Morimoto <morimoto.kuninori@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SSIU/SSI support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> +// +// Based on fsi.c +// Kuninori Morimoto <morimoto.kuninori@renesas.com> /* * you can enable below define if you don't need @@ -37,6 +33,7 @@ #define CHNL_4 (1 << 22) /* Channels */ #define CHNL_6 (2 << 22) /* Channels */ #define CHNL_8 (3 << 22) /* Channels */ +#define DWL_MASK (7 << 19) /* Data Word Length mask */ #define DWL_8 (0 << 19) /* Data Word Length */ #define DWL_16 (1 << 19) /* Data Word Length */ #define DWL_18 (2 << 19) /* Data Word Length */ @@ -353,21 +350,18 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, struct rsnd_dai *rdai = rsnd_io_to_rdai(io); struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - u32 cr_own; - u32 cr_mode; - u32 wsr; + u32 cr_own = ssi->cr_own; + u32 cr_mode = ssi->cr_mode; + u32 wsr = ssi->wsr; int is_tdm; - if (rsnd_ssi_is_parent(mod, io)) - return; - is_tdm = rsnd_runtime_is_ssi_tdm(io); /* * always use 32bit system word. * see also rsnd_ssi_master_clk_enable() */ - cr_own = FORCE | SWL_32; + cr_own |= FORCE | SWL_32; if (rdai->bit_clk_inv) cr_own |= SCKP; @@ -377,9 +371,18 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_own |= SDTA; if (rdai->sys_delay) cr_own |= DEL; + + /* + * We shouldn't exchange SWSP after running. + * This means, parent needs to care it. + */ + if (rsnd_ssi_is_parent(mod, io)) + goto init_end; + if (rsnd_io_is_play(io)) cr_own |= TRMD; + cr_own &= ~DWL_MASK; switch (snd_pcm_format_width(runtime->format)) { case 16: cr_own |= DWL_16; @@ -406,7 +409,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, wsr |= WS_MODE; cr_own |= CHNL_8; } - +init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; ssi->wsr = wsr; @@ -470,15 +473,18 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod, return -EIO; } - if (!rsnd_ssi_is_parent(mod, io)) - ssi->cr_own = 0; - rsnd_ssi_master_clk_stop(mod, io); rsnd_mod_power_off(mod); ssi->usrcnt--; + if (!ssi->usrcnt) { + ssi->cr_own = 0; + ssi->cr_mode = 0; + ssi->wsr = 0; + } + return 0; } @@ -1055,9 +1061,10 @@ struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id) int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod) { - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + if (!mod) + return 0; - return !!(rsnd_flags_has(ssi, RSND_SSI_CLK_PIN_SHARE)); + return !!(rsnd_flags_has(rsnd_mod_to_ssi(mod), RSND_SSI_CLK_PIN_SHARE)); } static u32 *rsnd_ssi_get_status(struct rsnd_dai_stream *io, diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 47bdba9fc582..016fbf5ac242 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -1,12 +1,9 @@ -/* - * Renesas R-Car SSIU support - * - * Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SSIU support +// +// Copyright (c) 2015 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + #include "rsnd.h" #define SSIU_NAME "ssiu" diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 4a3568a9bf59..4bb4c13cf860 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -1,10 +1,8 @@ -/* - * Generic AC97 sound support for SH7760 - * - * (c) 2007 Manuel Lauss - * - * Licensed under the GPLv2. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Generic AC97 sound support for SH7760 +// +// (c) 2007 Manuel Lauss #include <linux/module.h> #include <linux/moduleparam.h> @@ -68,6 +66,6 @@ static void __exit sh7760_ac97_exit(void) module_init(sh7760_ac97_init); module_exit(sh7760_ac97_exit); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Generic SH7760 AC97 sound machine"); MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/siu.h b/sound/soc/sh/siu.h index 6088d627c0e4..63a508fdfe78 100644 --- a/sound/soc/sh/siu.h +++ b/sound/soc/sh/siu.h @@ -1,23 +1,9 @@ -/* - * siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> - * Copyright (C) 2006 Carlos Munoz <carlos@kenati.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> +// Copyright (C) 2006 Carlos Munoz <carlos@kenati.com> #ifndef SIU_H #define SIU_H diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index ee2211635e92..f2a386fcd92e 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -1,23 +1,9 @@ -/* - * siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> - * Copyright (C) 2006 Carlos Munoz <carlos@kenati.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> +// Copyright (C) 2006 Carlos Munoz <carlos@kenati.com> #include <linux/delay.h> #include <linux/firmware.h> diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index 172909570ed5..e263757e4a69 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -1,23 +1,10 @@ -/* - * siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral. - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> - * Copyright (C) 2006 Carlos Munoz <carlos@kenati.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral. +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de> +// Copyright (C) 2006 Carlos Munoz <carlos@kenati.com> + #include <linux/delay.h> #include <linux/dma-mapping.h> #include <linux/dmaengine.h> diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 89ed1b107ac5..8125fa3840b6 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -1,14 +1,11 @@ -/* - * Serial Sound Interface (I2S) support for SH7760/SH7780 - * - * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> - * - * licensed under the terms outlined in the file COPYING at the root - * of the linux kernel sources. - * - * dont forget to set IPSEL/OMSEL register bits (in your board code) to - * enable SSI output pins! - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Serial Sound Interface (I2S) support for SH7760/SH7780 +// +// Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> +// +// dont forget to set IPSEL/OMSEL register bits (in your board code) to +// enable SSI output pins! /* * LIMITATIONS: @@ -400,6 +397,6 @@ static struct platform_driver sh4_ssi_driver = { module_platform_driver(sh4_ssi_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver"); MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c index 77e7dcf969d0..d70fcd4a1adf 100644 --- a/sound/soc/sirf/sirf-usp.c +++ b/sound/soc/sirf/sirf-usp.c @@ -370,10 +370,9 @@ static int sirf_usp_pcm_probe(struct platform_device *pdev) platform_set_drvdata(pdev, usp); mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap(&pdev->dev, mem_res->start, - resource_size(mem_res)); - if (base == NULL) - return -ENOMEM; + base = devm_ioremap_resource(&pdev->dev, mem_res); + if (IS_ERR(base)) + return PTR_ERR(base); usp->regmap = devm_regmap_init_mmio(&pdev->dev, base, &sirf_usp_regmap_config); if (IS_ERR(usp->regmap)) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 3f424f214bca..c086786e4471 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -1,20 +1,15 @@ -/* - * soc-ac97.c -- ALSA SoC Audio Layer AC97 support - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Author: Liam Girdwood <lrg@slimlogic.co.uk> - * with code, comments and ideas from :- - * Richard Purdie <richard@openedhand.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-ac97.c -- ALSA SoC Audio Layer AC97 support +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Author: Liam Girdwood <lrg@slimlogic.co.uk> +// with code, comments and ideas from :- +// Richard Purdie <richard@openedhand.com> #include <linux/ctype.h> #include <linux/delay.h> diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index 3d7e1ff79139..b8e72b52db30 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -1,18 +1,8 @@ -/* - * soc-apci.c - support for ACPI enumeration. - * - * Copyright (c) 2013-15, Intel Corporation. - * - * - * This program is free software; you can redistribute it and/or modify it - * under the terms and conditions of the GNU General Public License, - * version 2, as published by the Free Software Foundation. - * - * This program is distributed in the hope it will be useful, but WITHOUT - * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or - * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for - * more details. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// soc-apci.c - support for ACPI enumeration. +// +// Copyright (c) 2013-15, Intel Corporation. #include <sound/soc-acpi.h> diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index e095115fa9f9..409d082e80d1 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -1,18 +1,12 @@ -/* - * soc-compress.c -- ALSA SoC Compress - * - * Copyright (C) 2012 Intel Corp. - * - * Authors: Namarta Kohli <namartax.kohli@intel.com> - * Ramesh Babu K V <ramesh.babu@linux.intel.com> - * Vinod Koul <vinod.koul@linux.intel.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-compress.c -- ALSA SoC Compress +// +// Copyright (C) 2012 Intel Corp. +// +// Authors: Namarta Kohli <namartax.kohli@intel.com> +// Ramesh Babu K V <ramesh.babu@linux.intel.com> +// Vinod Koul <vinod.koul@linux.intel.com> #include <linux/kernel.h> #include <linux/init.h> @@ -146,6 +140,30 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) stream = SNDRV_PCM_STREAM_CAPTURE; mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + fe->dpcm[stream].runtime = fe_substream->runtime; + + ret = dpcm_path_get(fe, stream, &list); + if (ret < 0) + goto be_err; + else if (ret == 0) + dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n", + fe->dai_link->name, stream ? "capture" : "playback"); + /* calculate valid and active FE <-> BE dpcms */ + dpcm_process_paths(fe, stream, &list, 1); + fe->dpcm[stream].runtime = fe_substream->runtime; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_startup(fe, stream); + if (ret < 0) { + /* clean up all links */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + dpcm_be_disconnect(fe, stream); + fe->dpcm[stream].runtime = NULL; + goto out; + } if (cpu_dai->driver->cops && cpu_dai->driver->cops->startup) { ret = cpu_dai->driver->cops->startup(cstream, cpu_dai); @@ -159,7 +177,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) ret = soc_compr_components_open(cstream, &component); if (ret < 0) - goto machine_err; + goto open_err; if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->startup) { ret = fe->dai_link->compr_ops->startup(cstream); @@ -170,31 +188,6 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) } } - fe->dpcm[stream].runtime = fe_substream->runtime; - - ret = dpcm_path_get(fe, stream, &list); - if (ret < 0) - goto fe_err; - else if (ret == 0) - dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n", - fe->dai_link->name, stream ? "capture" : "playback"); - - /* calculate valid and active FE <-> BE dpcms */ - dpcm_process_paths(fe, stream, &list, 1); - - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; - - ret = dpcm_be_dai_startup(fe, stream); - if (ret < 0) { - /* clean up all links */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) - dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; - - dpcm_be_disconnect(fe, stream); - fe->dpcm[stream].runtime = NULL; - goto path_err; - } - dpcm_clear_pending_state(fe, stream); dpcm_path_put(&list); @@ -207,17 +200,14 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) return 0; -path_err: - dpcm_path_put(&list); -fe_err: - if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown) - fe->dai_link->compr_ops->shutdown(cstream); machine_err: soc_compr_components_free(cstream, component); - +open_err: if (cpu_dai->driver->cops && cpu_dai->driver->cops->shutdown) cpu_dai->driver->cops->shutdown(cstream, cpu_dai); out: + dpcm_path_put(&list); +be_err: fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; mutex_unlock(&fe->card->mutex); return ret; @@ -557,6 +547,24 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + /* + * Create an empty hw_params for the BE as the machine driver must + * fix this up to match DSP decoder and ASRC configuration. + * I.e. machine driver fixup for compressed BE is mandatory. + */ + memset(&fe->dpcm[fe_substream->stream].hw_params, 0, + sizeof(struct snd_pcm_hw_params)); + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_hw_params(fe, stream); + if (ret < 0) + goto out; + + ret = dpcm_be_dai_prepare(fe, stream); + if (ret < 0) + goto out; + if (cpu_dai->driver->cops && cpu_dai->driver->cops->set_params) { ret = cpu_dai->driver->cops->set_params(cstream, params, cpu_dai); if (ret < 0) @@ -583,24 +591,6 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, goto out; } - /* - * Create an empty hw_params for the BE as the machine driver must - * fix this up to match DSP decoder and ASRC configuration. - * I.e. machine driver fixup for compressed BE is mandatory. - */ - memset(&fe->dpcm[fe_substream->stream].hw_params, 0, - sizeof(struct snd_pcm_hw_params)); - - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; - - ret = dpcm_be_dai_hw_params(fe, stream); - if (ret < 0) - goto out; - - ret = dpcm_be_dai_prepare(fe, stream); - if (ret < 0) - goto out; - dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4663de3cf495..9cfe10d8040c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1,26 +1,21 @@ -/* - * soc-core.c -- ALSA SoC Audio Layer - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Author: Liam Girdwood <lrg@slimlogic.co.uk> - * with code, comments and ideas from :- - * Richard Purdie <richard@openedhand.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * TODO: - * o Add hw rules to enforce rates, etc. - * o More testing with other codecs/machines. - * o Add more codecs and platforms to ensure good API coverage. - * o Support TDM on PCM and I2S - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-core.c -- ALSA SoC Audio Layer +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Author: Liam Girdwood <lrg@slimlogic.co.uk> +// with code, comments and ideas from :- +// Richard Purdie <richard@openedhand.com> +// +// TODO: +// o Add hw rules to enforce rates, etc. +// o More testing with other codecs/machines. +// o Add more codecs and platforms to ensure good API coverage. +// o Support TDM on PCM and I2S #include <linux/module.h> #include <linux/moduleparam.h> @@ -533,6 +528,7 @@ int snd_soc_suspend(struct device *dev) "ASoC: idle_bias_off CODEC on over suspend\n"); break; } + /* fall through */ case SND_SOC_BIAS_OFF: if (component->driver->suspend) @@ -852,6 +848,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, const char *platform_name; int i; + if (dai_link->ignore) + return 0; + dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name); if (soc_is_dai_link_bound(card, dai_link)) { @@ -1195,15 +1194,27 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_remove_dai_link); +static void soc_set_of_name_prefix(struct snd_soc_component *component) +{ + struct device_node *component_of_node = component->dev->of_node; + const char *str; + int ret; + + if (!component_of_node && component->dev->parent) + component_of_node = component->dev->parent->of_node; + + ret = of_property_read_string(component_of_node, "sound-name-prefix", + &str); + if (!ret) + component->name_prefix = str; +} + static void soc_set_name_prefix(struct snd_soc_card *card, struct snd_soc_component *component) { int i; - if (card->codec_conf == NULL) - return; - - for (i = 0; i < card->num_configs; i++) { + for (i = 0; i < card->num_configs && card->codec_conf; i++) { struct snd_soc_codec_conf *map = &card->codec_conf[i]; struct device_node *component_of_node = component->dev->of_node; @@ -1215,8 +1226,14 @@ static void soc_set_name_prefix(struct snd_soc_card *card, if (map->dev_name && strcmp(component->name, map->dev_name)) continue; component->name_prefix = map->name_prefix; - break; + return; } + + /* + * If there is no configuration table or no match in the table, + * check if a prefix is provided in the node + */ + soc_set_of_name_prefix(component); } static int soc_probe_component(struct snd_soc_card *card, @@ -1461,7 +1478,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, ret; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + int i, ret, num; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, rtd->num, order); @@ -1507,9 +1526,28 @@ static int soc_probe_link_dais(struct snd_soc_card *card, soc_dpcm_debugfs_add(rtd); #endif + num = rtd->num; + + /* + * most drivers will register their PCMs using DAI link ordering but + * topology based drivers can use the DAI link id field to set PCM + * device number and then use rtd + a base offset of the BEs. + */ + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (!component->driver->use_dai_pcm_id) + continue; + + if (rtd->dai_link->no_pcm) + num += component->driver->be_pcm_base; + else + num = rtd->dai_link->id; + } + if (cpu_dai->driver->compress_new) { /*create compress_device"*/ - ret = cpu_dai->driver->compress_new(rtd, rtd->num); + ret = cpu_dai->driver->compress_new(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); @@ -1519,7 +1557,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, if (!dai_link->params) { /* create the pcm */ - ret = soc_new_pcm(rtd, rtd->num); + ret = soc_new_pcm(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); @@ -1846,6 +1884,74 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) EXPORT_SYMBOL_GPL(snd_soc_set_dmi_name); #endif /* CONFIG_DMI */ +static void soc_check_tplg_fes(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + const struct snd_soc_component_driver *comp_drv; + struct snd_soc_dai_link *dai_link; + int i; + + list_for_each_entry(component, &component_list, list) { + + /* does this component override FEs ? */ + if (!component->driver->ignore_machine) + continue; + + /* for this machine ? */ + if (strcmp(component->driver->ignore_machine, + card->dev->driver->name)) + continue; + + /* machine matches, so override the rtd data */ + for (i = 0; i < card->num_links; i++) { + + dai_link = &card->dai_link[i]; + + /* ignore this FE */ + if (dai_link->dynamic) { + dai_link->ignore = true; + continue; + } + + dev_info(card->dev, "info: override FE DAI link %s\n", + card->dai_link[i].name); + + /* override platform component */ + dai_link->platform_name = component->name; + + /* convert non BE into BE */ + dai_link->no_pcm = 1; + + /* override any BE fixups */ + dai_link->be_hw_params_fixup = + component->driver->be_hw_params_fixup; + + /* most BE links don't set stream name, so set it to + * dai link name if it's NULL to help bind widgets. + */ + if (!dai_link->stream_name) + dai_link->stream_name = dai_link->name; + } + + /* Inform userspace we are using alternate topology */ + if (component->driver->topology_name_prefix) { + + /* topology shortname created ? */ + if (!card->topology_shortname_created) { + comp_drv = component->driver; + + snprintf(card->topology_shortname, 32, "%s-%s", + comp_drv->topology_name_prefix, + card->name); + card->topology_shortname_created = true; + } + + /* use topology shortname */ + card->name = card->topology_shortname; + } + } +} + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; @@ -1855,6 +1961,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); + /* check whether any platform is ignore machine FE and using topology */ + soc_check_tplg_fes(card); + /* bind DAIs */ for (i = 0; i < card->num_links; i++) { ret = soc_bind_dai_link(card, &card->dai_link[i]); @@ -2523,6 +2632,28 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); /** + * snd_soc_dai_get_channel_map - Get DAI audio channel map + * @dai: DAI + * @tx_num: how many TX channels + * @tx_slot: pointer to an array which imply the TX slot number channel + * 0~num-1 uses + * @rx_num: how many RX channels + * @rx_slot: pointer to an array which imply the RX slot number channel + * 0~num-1 uses + */ +int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai, + unsigned int *tx_num, unsigned int *tx_slot, + unsigned int *rx_num, unsigned int *rx_slot) +{ + if (dai->driver->ops->get_channel_map) + return dai->driver->ops->get_channel_map(dai, tx_num, tx_slot, + rx_num, rx_slot); + else + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_get_channel_map); + +/** * snd_soc_dai_set_tristate - configure DAI system or master clock. * @dai: DAI * @tristate: tristate enable @@ -3258,9 +3389,9 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets); -static int snd_soc_of_get_slot_mask(struct device_node *np, - const char *prop_name, - unsigned int *mask) +int snd_soc_of_get_slot_mask(struct device_node *np, + const char *prop_name, + unsigned int *mask) { u32 val; const __be32 *of_slot_mask = of_get_property(np, prop_name, &val); @@ -3275,6 +3406,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np, return val; } +EXPORT_SYMBOL_GPL(snd_soc_of_get_slot_mask); int snd_soc_of_parse_tdm_slot(struct device_node *np, unsigned int *tx_mask, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 229c12349803..7e96793050c9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1,27 +1,21 @@ -/* - * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood <lrg@slimlogic.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * Features: - * o Changes power status of internal codec blocks depending on the - * dynamic configuration of codec internal audio paths and active - * DACs/ADCs. - * o Platform power domain - can support external components i.e. amps and - * mic/headphone insertion events. - * o Automatic Mic Bias support - * o Jack insertion power event initiation - e.g. hp insertion will enable - * sinks, dacs, etc - * o Delayed power down of audio subsystem to reduce pops between a quick - * device reopen. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-dapm.c -- ALSA SoC Dynamic Audio Power Management +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Author: Liam Girdwood <lrg@slimlogic.co.uk> +// +// Features: +// o Changes power status of internal codec blocks depending on the +// dynamic configuration of codec internal audio paths and active +// DACs/ADCs. +// o Platform power domain - can support external components i.e. amps and +// mic/headphone insertion events. +// o Automatic Mic Bias support +// o Jack insertion power event initiation - e.g. hp insertion will enable +// sinks, dacs, etc +// o Delayed power down of audio subsystem to reduce pops between a quick +// device reopen. #include <linux/module.h> #include <linux/moduleparam.h> @@ -3662,7 +3656,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; - u64 fmt; + unsigned int fmt; int ret; if (WARN_ON(!config) || @@ -4073,6 +4067,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) continue; } + /* let users know there is no DAI to link */ + if (!dai_w->priv) { + dev_dbg(card->dev, "dai widget %s has no DAI\n", + dai_w->name); + continue; + } + dai = dai_w->priv; /* ...find all widgets with the same stream and link them */ diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 7ac745df1412..a9ea172a66a7 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -1,13 +1,8 @@ -/* - * soc-devres.c -- ALSA SoC Audio Layer devres functions - * - * Copyright (C) 2013 Linaro Ltd - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-devres.c -- ALSA SoC Audio Layer devres functions +// +// Copyright (C) 2013 Linaro Ltd #include <linux/module.h> #include <linux/moduleparam.h> diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 56a541b9ff9e..52fd7af952a5 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -1,17 +1,8 @@ -/* - * Copyright (C) 2013, Analog Devices Inc. - * Author: Lars-Peter Clausen <lars@metafoo.de> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 675 Mass Ave, Cambridge, MA 02139, USA. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright (C) 2013, Analog Devices Inc. +// Author: Lars-Peter Clausen <lars@metafoo.de> + #include <linux/module.h> #include <linux/init.h> #include <linux/dmaengine.h> @@ -197,7 +188,7 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea case 32: case 64: if (addr_widths & (1 << (bits / 8))) - hw.formats |= (1LL << i); + hw.formats |= pcm_format_to_bits(i); break; default: /* Unsupported types */ @@ -343,7 +334,7 @@ static snd_pcm_uframes_t dmaengine_pcm_pointer( static int dmaengine_copy_user(struct snd_pcm_substream *substream, int channel, unsigned long hwoff, - void *buf, unsigned long bytes) + void __user *buf, unsigned long bytes) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component = @@ -359,18 +350,17 @@ static int dmaengine_copy_user(struct snd_pcm_substream *substream, int ret; if (is_playback) - if (copy_from_user(dma_ptr, (void __user *)buf, bytes)) + if (copy_from_user(dma_ptr, buf, bytes)) return -EFAULT; if (process) { - ret = process(substream, channel, hwoff, - (void __user *)buf, bytes); + ret = process(substream, channel, hwoff, (__force void *)buf, bytes); if (ret < 0) return ret; } if (!is_playback) - if (copy_to_user((void __user *)buf, dma_ptr, bytes)) + if (copy_to_user(buf, dma_ptr, bytes)) return -EFAULT; return 0; diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 026cd5347e53..1ff9175e9d5e 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -1,15 +1,10 @@ -/* - * soc-io.c -- ASoC register I/O helpers - * - * Copyright 2009-2011 Wolfson Microelectronics PLC. - * - * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-io.c -- ASoC register I/O helpers +// +// Copyright 2009-2011 Wolfson Microelectronics PLC. +// +// Author: Mark Brown <broonie@opensource.wolfsonmicro.com> #include <linux/i2c.h> #include <linux/spi/spi.h> diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index b2b16044ae80..c7b990abdbaa 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -1,15 +1,10 @@ -/* - * soc-jack.c -- ALSA SoC jack handling - * - * Copyright 2008 Wolfson Microelectronics PLC. - * - * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-jack.c -- ALSA SoC jack handling +// +// Copyright 2008 Wolfson Microelectronics PLC. +// +// Author: Mark Brown <broonie@opensource.wolfsonmicro.com> #include <sound/jack.h> #include <sound/soc.h> diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 7144a51ddfa9..592efb370c44 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -1,20 +1,15 @@ -/* - * soc-ops.c -- Generic ASoC operations - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Author: Liam Girdwood <lrg@slimlogic.co.uk> - * with code, comments and ideas from :- - * Richard Purdie <richard@openedhand.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-ops.c -- Generic ASoC operations +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Author: Liam Girdwood <lrg@slimlogic.co.uk> +// with code, comments and ideas from :- +// Richard Purdie <richard@openedhand.com> #include <linux/module.h> #include <linux/moduleparam.h> diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5e7ae47a9658..e8b98bfd4cf1 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1,20 +1,14 @@ -/* - * soc-pcm.c -- ALSA SoC PCM - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Authors: Liam Girdwood <lrg@ti.com> - * Mark Brown <broonie@opensource.wolfsonmicro.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-pcm.c -- ALSA SoC PCM +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Authors: Liam Girdwood <lrg@ti.com> +// Mark Brown <broonie@opensource.wolfsonmicro.com> #include <linux/kernel.h> #include <linux/init.h> @@ -448,6 +442,29 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) hw->rate_max = min_not_zero(hw->rate_max, rate_max); } +static int soc_pcm_components_close(struct snd_pcm_substream *substream, + struct snd_soc_component *last) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (component == last) + break; + + if (!component->driver->ops || + !component->driver->ops->close) + continue; + + component->driver->ops->close(substream); + } + + return 0; +} + /* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls @@ -462,7 +479,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; const char *codec_dai_name = "multicodec"; - int i, ret = 0, __ret; + int i, ret = 0; pinctrl_pm_select_default_state(cpu_dai->dev); for (i = 0; i < rtd->num_codecs; i++) @@ -486,7 +503,6 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - ret = 0; for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -494,16 +510,15 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) !component->driver->ops->open) continue; - __ret = component->driver->ops->open(substream); - if (__ret < 0) { + ret = component->driver->ops->open(substream); + if (ret < 0) { dev_err(component->dev, "ASoC: can't open component %s: %d\n", - component->name, __ret); - ret = __ret; + component->name, ret); + goto component_err; } } - if (ret < 0) - goto component_err; + component = NULL; for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; @@ -612,15 +627,7 @@ codec_dai_err: } component_err: - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->close) - continue; - - component->driver->ops->close(substream); - } + soc_pcm_components_close(substream, component); if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); @@ -714,15 +721,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) if (rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->close) - continue; - - component->driver->ops->close(substream); - } + soc_pcm_components_close(substream, NULL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (snd_soc_runtime_ignore_pmdown_time(rtd)) { @@ -860,8 +859,20 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; int ret; + /* perform any topology hw_params fixups before DAI */ + if (rtd->dai_link->be_hw_params_fixup) { + ret = rtd->dai_link->be_hw_params_fixup(rtd, params); + if (ret < 0) { + dev_err(rtd->dev, + "ASoC: hw_params topology fixup failed %d\n", + ret); + return ret; + } + } + if (dai->driver->ops->hw_params) { ret = dai->driver->ops->hw_params(substream, params, dai); if (ret < 0) { @@ -874,6 +885,29 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, return 0; } +static int soc_pcm_components_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_component *last) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (component == last) + break; + + if (!component->driver->ops || + !component->driver->ops->hw_free) + continue; + + component->driver->ops->hw_free(substream); + } + + return 0; +} + /* * Called by ALSA when the hardware params are set by application. This * function can also be called multiple times and can allocate buffers @@ -886,7 +920,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, ret = 0, __ret; + int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); if (rtd->dai_link->ops->hw_params) { @@ -944,7 +978,6 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (ret < 0) goto interface_err; - ret = 0; for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -952,16 +985,15 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, !component->driver->ops->hw_params) continue; - __ret = component->driver->ops->hw_params(substream, params); - if (__ret < 0) { + ret = component->driver->ops->hw_params(substream, params); + if (ret < 0) { dev_err(component->dev, "ASoC: %s hw params failed: %d\n", - component->name, __ret); - ret = __ret; + component->name, ret); + goto component_err; } } - if (ret < 0) - goto component_err; + component = NULL; /* store the parameters for each DAIs */ cpu_dai->rate = params_rate(params); @@ -977,15 +1009,7 @@ out: return ret; component_err: - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->hw_free) - continue; - - component->driver->ops->hw_free(substream); - } + soc_pcm_components_hw_free(substream, component); if (cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); @@ -1014,8 +1038,6 @@ codec_err: static int soc_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *component; - struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; @@ -1052,15 +1074,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) rtd->dai_link->ops->hw_free(substream); /* free any component resources */ - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->hw_free) - continue; - - component->driver->ops->hw_free(substream); - } + soc_pcm_components_hw_free(substream, NULL); /* now free hw params for the DAIs */ for (i = 0; i < rtd->num_codecs; i++) { @@ -1165,6 +1179,9 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) snd_pcm_sframes_t codec_delay = 0; int i; + /* clearing the previous total delay */ + runtime->delay = 0; + for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -1176,6 +1193,8 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) offset = component->driver->ops->pointer(substream); break; } + /* base delay if assigned in pointer callback */ + delay = runtime->delay; if (cpu_dai->driver->ops->delay) delay += cpu_dai->driver->ops->delay(substream, cpu_dai); @@ -1658,29 +1677,28 @@ unwind: } static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, - struct snd_soc_pcm_stream *stream, - u64 formats) + struct snd_soc_pcm_stream *stream) { runtime->hw.rate_min = stream->rate_min; runtime->hw.rate_max = stream->rate_max; runtime->hw.channels_min = stream->channels_min; runtime->hw.channels_max = stream->channels_max; if (runtime->hw.formats) - runtime->hw.formats &= formats & stream->formats; + runtime->hw.formats &= stream->formats; else - runtime->hw.formats = formats & stream->formats; + runtime->hw.formats = stream->formats; runtime->hw.rates = stream->rates; } -static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) +static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, + u64 *formats) { struct snd_soc_pcm_runtime *fe = substream->private_data; struct snd_soc_dpcm *dpcm; - u64 formats = ULLONG_MAX; int stream = substream->stream; if (!fe->dai_link->dpcm_merged_format) - return formats; + return; /* * It returns merged BE codec format @@ -1694,17 +1712,132 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) int i; for (i = 0; i < be->num_codecs; i++) { + /* + * Skip CODECs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(be->codec_dais[i], + stream)) + continue; + codec_dai_drv = be->codec_dais[i]->driver; if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; else codec_stream = &codec_dai_drv->capture; - formats &= codec_stream->formats; + *formats &= codec_stream->formats; } } +} + +static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, + unsigned int *channels_min, + unsigned int *channels_max) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + struct snd_soc_dpcm *dpcm; + int stream = substream->stream; - return formats; + if (!fe->dai_link->dpcm_merged_chan) + return; + + /* + * It returns merged BE codec channel; + * if FE want to use it (= dpcm_merged_chan) + */ + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_pcm_stream *codec_stream; + struct snd_soc_pcm_stream *cpu_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_stream = &cpu_dai_drv->playback; + else + cpu_stream = &cpu_dai_drv->capture; + + *channels_min = max(*channels_min, cpu_stream->channels_min); + *channels_max = min(*channels_max, cpu_stream->channels_max); + + /* + * chan min/max cannot be enforced if there are multiple CODEC + * DAIs connected to a single CPU DAI, use CPU DAI's directly + */ + if (be->num_codecs == 1) { + codec_dai_drv = be->codec_dais[0]->driver; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &codec_dai_drv->playback; + else + codec_stream = &codec_dai_drv->capture; + + *channels_min = max(*channels_min, + codec_stream->channels_min); + *channels_max = min(*channels_max, + codec_stream->channels_max); + } + } +} + +static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, + unsigned int *rates, + unsigned int *rate_min, + unsigned int *rate_max) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + struct snd_soc_dpcm *dpcm; + int stream = substream->stream; + + if (!fe->dai_link->dpcm_merged_rate) + return; + + /* + * It returns merged BE codec channel; + * if FE want to use it (= dpcm_merged_chan) + */ + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_pcm_stream *codec_stream; + struct snd_soc_pcm_stream *cpu_stream; + int i; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_stream = &cpu_dai_drv->playback; + else + cpu_stream = &cpu_dai_drv->capture; + + *rate_min = max(*rate_min, cpu_stream->rate_min); + *rate_max = min_not_zero(*rate_max, cpu_stream->rate_max); + *rates = snd_pcm_rate_mask_intersect(*rates, cpu_stream->rates); + + for (i = 0; i < be->num_codecs; i++) { + /* + * Skip CODECs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(be->codec_dais[i], + stream)) + continue; + + codec_dai_drv = be->codec_dais[i]->driver; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &codec_dai_drv->playback; + else + codec_stream = &codec_dai_drv->capture; + + *rate_min = max(*rate_min, codec_stream->rate_min); + *rate_max = min_not_zero(*rate_max, + codec_stream->rate_max); + *rates = snd_pcm_rate_mask_intersect(*rates, + codec_stream->rates); + } + } } static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) @@ -1713,12 +1846,17 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - u64 format = dpcm_runtime_base_format(substream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback, format); + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback); else - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture, format); + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture); + + dpcm_runtime_merge_format(substream, &runtime->hw.formats); + dpcm_runtime_merge_chan(substream, &runtime->hw.channels_min, + &runtime->hw.channels_max); + dpcm_runtime_merge_rate(substream, &runtime->hw.rates, + &runtime->hw.rate_min, &runtime->hw.rate_max); } static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd); @@ -2543,106 +2681,113 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) return ret; } -/* Called by DAPM mixer/mux changes to update audio routing between PCMs and - * any DAI links. - */ -int soc_dpcm_runtime_update(struct snd_soc_card *card) +static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) { - struct snd_soc_pcm_runtime *fe; - int old, new, paths; + struct snd_soc_dapm_widget_list *list; + int count, paths; - mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - list_for_each_entry(fe, &card->rtd_list, list) { - struct snd_soc_dapm_widget_list *list; + if (!fe->dai_link->dynamic) + return 0; - /* make sure link is FE */ - if (!fe->dai_link->dynamic) - continue; + /* only check active links */ + if (!fe->cpu_dai->active) + return 0; - /* only check active links */ - if (!fe->cpu_dai->active) - continue; + /* DAPM sync will call this to update DSP paths */ + dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n", + new ? "new" : "old", fe->dai_link->name); - /* DAPM sync will call this to update DSP paths */ - dev_dbg(fe->dev, "ASoC: DPCM runtime update for FE %s\n", - fe->dai_link->name); + /* skip if FE doesn't have playback capability */ + if (!fe->cpu_dai->driver->playback.channels_min || + !fe->codec_dai->driver->playback.channels_min) + goto capture; - /* skip if FE doesn't have playback capability */ - if (!fe->cpu_dai->driver->playback.channels_min - || !fe->codec_dai->driver->playback.channels_min) - goto capture; - - /* skip if FE isn't currently playing */ - if (!fe->cpu_dai->playback_active - || !fe->codec_dai->playback_active) - goto capture; - - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "playback"); - mutex_unlock(&card->mutex); - return paths; - } + /* skip if FE isn't currently playing */ + if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active) + goto capture; - /* update any new playback paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 1); - if (new) { - dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "playback"); + return paths; + } - /* update any old playback paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 0); - if (old) { + /* update any playback paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new); + if (count) { + if (new) + dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); + else dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } - dpcm_path_put(&list); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); + } + + dpcm_path_put(&list); + capture: - /* skip if FE doesn't have capture capability */ - if (!fe->cpu_dai->driver->capture.channels_min - || !fe->codec_dai->driver->capture.channels_min) - continue; + /* skip if FE doesn't have capture capability */ + if (!fe->cpu_dai->driver->capture.channels_min || + !fe->codec_dai->driver->capture.channels_min) + return 0; - /* skip if FE isn't currently capturing */ - if (!fe->cpu_dai->capture_active - || !fe->codec_dai->capture_active) - continue; + /* skip if FE isn't currently capturing */ + if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active) + return 0; - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "capture"); - mutex_unlock(&card->mutex); - return paths; - } + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "capture"); + return paths; + } - /* update any new capture paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 1); - if (new) { + /* update any old capture paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new); + if (count) { + if (new) dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - - /* update any old capture paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 0); - if (old) { + else dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - dpcm_path_put(&list); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); } - mutex_unlock(&card->mutex); + dpcm_path_put(&list); + return 0; } + +/* Called by DAPM mixer/mux changes to update audio routing between PCMs and + * any DAI links. + */ +int soc_dpcm_runtime_update(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *fe; + int ret = 0; + + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + /* shutdown all old paths first */ + list_for_each_entry(fe, &card->rtd_list, list) { + ret = soc_dpcm_fe_runtime_update(fe, 0); + if (ret) + goto out; + } + + /* bring new paths up */ + list_for_each_entry(fe, &card->rtd_list, list) { + ret = soc_dpcm_fe_runtime_update(fe, 1); + if (ret) + goto out; + } + +out: + mutex_unlock(&card->mutex); + return ret; +} int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) { struct snd_soc_dpcm *dpcm; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 53f121a50c97..66e77e020745 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1,29 +1,24 @@ -/* - * soc-topology.c -- ALSA SoC Topology - * - * Copyright (C) 2012 Texas Instruments Inc. - * Copyright (C) 2015 Intel Corporation. - * - * Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com> - * K, Mythri P <mythri.p.k@intel.com> - * Prusty, Subhransu S <subhransu.s.prusty@intel.com> - * B, Jayachandran <jayachandran.b@intel.com> - * Abdullah, Omair M <omair.m.abdullah@intel.com> - * Jin, Yao <yao.jin@intel.com> - * Lin, Mengdong <mengdong.lin@intel.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * Add support to read audio firmware topology alongside firmware text. The - * topology data can contain kcontrols, DAPM graphs, widgets, DAIs, DAI links, - * equalizers, firmware, coefficients etc. - * - * This file only manages the core ALSA and ASoC components, all other bespoke - * firmware topology data is passed to component drivers for bespoke handling. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-topology.c -- ALSA SoC Topology +// +// Copyright (C) 2012 Texas Instruments Inc. +// Copyright (C) 2015 Intel Corporation. +// +// Authors: Liam Girdwood <liam.r.girdwood@linux.intel.com> +// K, Mythri P <mythri.p.k@intel.com> +// Prusty, Subhransu S <subhransu.s.prusty@intel.com> +// B, Jayachandran <jayachandran.b@intel.com> +// Abdullah, Omair M <omair.m.abdullah@intel.com> +// Jin, Yao <yao.jin@intel.com> +// Lin, Mengdong <mengdong.lin@intel.com> +// +// Add support to read audio firmware topology alongside firmware text. The +// topology data can contain kcontrols, DAPM graphs, widgets, DAIs, DAI links, +// equalizers, firmware, coefficients etc. +// +// This file only manages the core ALSA and ASoC components, all other bespoke +// firmware topology data is passed to component drivers for bespoke handling. #include <linux/kernel.h> #include <linux/export.h> @@ -259,7 +254,7 @@ static int soc_tplg_vendor_load_(struct soc_tplg *tplg, int ret = 0; if (tplg->comp && tplg->ops && tplg->ops->vendor_load) - ret = tplg->ops->vendor_load(tplg->comp, hdr); + ret = tplg->ops->vendor_load(tplg->comp, tplg->index, hdr); else { dev_err(tplg->dev, "ASoC: no vendor load callback for ID %d\n", hdr->vendor_type); @@ -291,7 +286,8 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { if (tplg->comp && tplg->ops && tplg->ops->widget_load) - return tplg->ops->widget_load(tplg->comp, w, tplg_w); + return tplg->ops->widget_load(tplg->comp, tplg->index, w, + tplg_w); return 0; } @@ -302,27 +298,30 @@ static int soc_tplg_widget_ready(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { if (tplg->comp && tplg->ops && tplg->ops->widget_ready) - return tplg->ops->widget_ready(tplg->comp, w, tplg_w); + return tplg->ops->widget_ready(tplg->comp, tplg->index, w, + tplg_w); return 0; } /* pass DAI configurations to component driver for extra initialization */ static int soc_tplg_dai_load(struct soc_tplg *tplg, - struct snd_soc_dai_driver *dai_drv) + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { if (tplg->comp && tplg->ops && tplg->ops->dai_load) - return tplg->ops->dai_load(tplg->comp, dai_drv); + return tplg->ops->dai_load(tplg->comp, tplg->index, dai_drv, + pcm, dai); return 0; } /* pass link configurations to component driver for extra initialization */ static int soc_tplg_dai_link_load(struct soc_tplg *tplg, - struct snd_soc_dai_link *link) + struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg) { if (tplg->comp && tplg->ops && tplg->ops->link_load) - return tplg->ops->link_load(tplg->comp, link); + return tplg->ops->link_load(tplg->comp, tplg->index, link, cfg); return 0; } @@ -643,7 +642,8 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, struct snd_kcontrol_new *k, struct snd_soc_tplg_ctl_hdr *hdr) { if (tplg->comp && tplg->ops && tplg->ops->control_load) - return tplg->ops->control_load(tplg->comp, k, hdr); + return tplg->ops->control_load(tplg->comp, tplg->index, k, + hdr); return 0; } @@ -1100,6 +1100,17 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, return 0; } +/* optionally pass new dynamic kcontrol to component driver. */ +static int soc_tplg_add_route(struct soc_tplg *tplg, + struct snd_soc_dapm_route *route) +{ + if (tplg->comp && tplg->ops && tplg->ops->dapm_route_load) + return tplg->ops->dapm_route_load(tplg->comp, tplg->index, + route); + + return 0; +} + static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { @@ -1148,6 +1159,8 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, else route.control = elem->control; + soc_tplg_add_route(tplg, &route); + /* add route, but keep going if some fail */ snd_soc_dapm_add_routes(dapm, &route, 1); } @@ -1702,7 +1715,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, dai_drv->compress_new = snd_soc_new_compress; /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_load(tplg, dai_drv); + ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); kfree(dai_drv); @@ -1772,7 +1785,7 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, set_link_flags(link, pcm->flag_mask, pcm->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_link_load(tplg, link); + ret = soc_tplg_dai_link_load(tplg, link, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n"); kfree(link); @@ -2080,7 +2093,7 @@ static int soc_tplg_link_config(struct soc_tplg *tplg, set_link_flags(link, cfg->flag_mask, cfg->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_link_load(tplg, link); + ret = soc_tplg_dai_link_load(tplg, link, cfg); if (ret < 0) { dev_err(tplg->dev, "ASoC: physical link loading failed\n"); return ret; @@ -2202,7 +2215,7 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, set_dai_flags(dai_drv, d->flag_mask, d->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_load(tplg, dai_drv); + ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); return ret; @@ -2311,7 +2324,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, /* pass control to component driver for optional further init */ if (tplg->comp && tplg->ops && tplg->ops->manifest) - return tplg->ops->manifest(tplg->comp, _manifest); + return tplg->ops->manifest(tplg->comp, tplg->index, _manifest); if (!abi_match) /* free the duplicated one */ kfree(_manifest); diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 2d9e98bd1530..e0c93496c0cd 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -1,17 +1,11 @@ -/* - * soc-util.c -- ALSA SoC Audio Layer utility functions - * - * Copyright 2009 Wolfson Microelectronics PLC. - * - * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> - * Liam Girdwood <lrg@slimlogic.co.uk> - * - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-util.c -- ALSA SoC Audio Layer utility functions +// +// Copyright 2009 Wolfson Microelectronics PLC. +// +// Author: Mark Brown <broonie@opensource.wolfsonmicro.com> +// Liam Girdwood <lrg@slimlogic.co.uk> #include <linux/platform_device.h> #include <linux/export.h> @@ -381,6 +375,6 @@ int __init snd_soc_util_init(void) void __exit snd_soc_util_exit(void) { - platform_device_unregister(soc_dummy_dev); platform_driver_unregister(&soc_dummy_driver); + platform_device_unregister(soc_dummy_dev); } diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index d8b6936e544e..313dab2857ef 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -91,7 +91,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_FIFO_ERROR(player); /* Stop the player */ - snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(player->substream); } ret = IRQ_HANDLED; @@ -105,7 +105,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player); /* Stop the player */ - snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(player->substream); ret = IRQ_HANDLED; } @@ -138,7 +138,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) dev_err(player->dev, "Underflow recovery failed\n"); /* Stop the player */ - snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(player->substream); ret = IRQ_HANDLED; } diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index ee0055e60852..7b63d35ef428 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -65,7 +65,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) { dev_err(reader->dev, "FIFO error detected\n"); - snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(reader->substream); ret = IRQ_HANDLED; } diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig index 48f9ddd94016..9b2681397dba 100644 --- a/sound/soc/stm/Kconfig +++ b/sound/soc/stm/Kconfig @@ -6,6 +6,7 @@ config SND_SOC_STM32_SAI depends on SND_SOC select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO + select SND_PCM_IEC958 help Say Y if you want to enable SAI for STM32 diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index db73fef3e500..706ff005234f 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -149,7 +149,7 @@ static int stm32_afsdm_pcm_cb(const void *data, size_t size, void *private) unsigned int old_pos = priv->pos; unsigned int cur_size = size; - dev_dbg(rtd->dev, "%s: buff_add :%p, pos = %d, size = %zu\n", + dev_dbg(rtd->dev, "%s: buff_add :%pK, pos = %d, size = %zu\n", __func__, &pcm_buff[priv->pos], priv->pos, size); if ((priv->pos + size) > buff_size) { @@ -269,16 +269,10 @@ static int stm32_adfsdm_pcm_new(struct snd_soc_pcm_runtime *rtd) static void stm32_adfsdm_pcm_free(struct snd_pcm *pcm) { struct snd_pcm_substream *substream; - struct snd_soc_pcm_runtime *rtd; - struct stm32_adfsdm_priv *priv; substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; - if (substream) { - rtd = substream->private_data; - priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); - + if (substream) snd_pcm_lib_preallocate_free_for_all(pcm); - } } static struct snd_soc_component_driver stm32_adfsdm_soc_platform = { diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index cfeb219e1d78..06fba9650ac4 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -96,7 +96,8 @@ * @slot_mask: rx or tx active slots mask. set at init or at runtime * @data_size: PCM data width. corresponds to PCM substream width. * @spdif_frm_cnt: S/PDIF playback frame counter - * @spdif_status_bits: S/PDIF status bits + * @snd_aes_iec958: iec958 data + * @ctrl_lock: control lock */ struct stm32_sai_sub_data { struct platform_device *pdev; @@ -125,7 +126,8 @@ struct stm32_sai_sub_data { int slot_mask; int data_size; unsigned int spdif_frm_cnt; - unsigned char spdif_status_bits[SAI_IEC60958_STATUS_BYTES]; + struct snd_aes_iec958 iec958; + struct mutex ctrl_lock; /* protect resources accessed by controls */ }; enum stm32_sai_fifo_th { @@ -184,10 +186,6 @@ static bool stm32_sai_sub_writeable_reg(struct device *dev, unsigned int reg) } } -static const unsigned char default_status_bits[SAI_IEC60958_STATUS_BYTES] = { - 0, 0, 0, IEC958_AES3_CON_FS_48000, -}; - static const struct regmap_config stm32_sai_sub_regmap_config_f4 = { .reg_bits = 32, .reg_stride = 4, @@ -210,6 +208,49 @@ static const struct regmap_config stm32_sai_sub_regmap_config_h7 = { .fast_io = true, }; +static int snd_pcm_iec958_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + + return 0; +} + +static int snd_pcm_iec958_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct stm32_sai_sub_data *sai = snd_kcontrol_chip(kcontrol); + + mutex_lock(&sai->ctrl_lock); + memcpy(uctl->value.iec958.status, sai->iec958.status, 4); + mutex_unlock(&sai->ctrl_lock); + + return 0; +} + +static int snd_pcm_iec958_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct stm32_sai_sub_data *sai = snd_kcontrol_chip(kcontrol); + + mutex_lock(&sai->ctrl_lock); + memcpy(sai->iec958.status, uctl->value.iec958.status, 4); + mutex_unlock(&sai->ctrl_lock); + + return 0; +} + +static const struct snd_kcontrol_new iec958_ctls = { + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE), + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = snd_pcm_iec958_info, + .get = snd_pcm_iec958_get, + .put = snd_pcm_iec958_put, +}; + static irqreturn_t stm32_sai_isr(int irq, void *devid) { struct stm32_sai_sub_data *sai = (struct stm32_sai_sub_data *)devid; @@ -259,11 +300,8 @@ static irqreturn_t stm32_sai_isr(int irq, void *devid) status = SNDRV_PCM_STATE_XRUN; } - if (status != SNDRV_PCM_STATE_RUNNING) { - snd_pcm_stream_lock(sai->substream); - snd_pcm_stop(sai->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(sai->substream); - } + if (status != SNDRV_PCM_STATE_RUNNING) + snd_pcm_stop_xrun(sai->substream); return IRQ_HANDLED; } @@ -619,6 +657,59 @@ static void stm32_sai_set_frame(struct snd_soc_dai *cpu_dai) } } +static void stm32_sai_init_iec958_status(struct stm32_sai_sub_data *sai) +{ + unsigned char *cs = sai->iec958.status; + + cs[0] = IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_NONE; + cs[1] = IEC958_AES1_CON_GENERAL; + cs[2] = IEC958_AES2_CON_SOURCE_UNSPEC | IEC958_AES2_CON_CHANNEL_UNSPEC; + cs[3] = IEC958_AES3_CON_CLOCK_1000PPM | IEC958_AES3_CON_FS_NOTID; +} + +static void stm32_sai_set_iec958_status(struct stm32_sai_sub_data *sai, + struct snd_pcm_runtime *runtime) +{ + if (!runtime) + return; + + /* Force the sample rate according to runtime rate */ + mutex_lock(&sai->ctrl_lock); + switch (runtime->rate) { + case 22050: + sai->iec958.status[3] = IEC958_AES3_CON_FS_22050; + break; + case 44100: + sai->iec958.status[3] = IEC958_AES3_CON_FS_44100; + break; + case 88200: + sai->iec958.status[3] = IEC958_AES3_CON_FS_88200; + break; + case 176400: + sai->iec958.status[3] = IEC958_AES3_CON_FS_176400; + break; + case 24000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_24000; + break; + case 48000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_48000; + break; + case 96000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_96000; + break; + case 192000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_192000; + break; + case 32000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_32000; + break; + default: + sai->iec958.status[3] = IEC958_AES3_CON_FS_NOTID; + break; + } + mutex_unlock(&sai->ctrl_lock); +} + static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, struct snd_pcm_hw_params *params) { @@ -709,7 +800,11 @@ static int stm32_sai_hw_params(struct snd_pcm_substream *substream, sai->data_size = params_width(params); - if (!STM_SAI_PROTOCOL_IS_SPDIF(sai)) { + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) { + /* Rate not already set in runtime structure */ + substream->runtime->rate = params_rate(params); + stm32_sai_set_iec958_status(sai, substream->runtime); + } else { ret = stm32_sai_set_slots(cpu_dai); if (ret < 0) return ret; @@ -789,6 +884,20 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, sai->substream = NULL; } +static int stm32_sai_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *cpu_dai) +{ + struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); + + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) { + dev_dbg(&sai->pdev->dev, "%s: register iec controls", __func__); + return snd_ctl_add(rtd->pcm->card, + snd_ctl_new1(&iec958_ctls, sai)); + } + + return 0; +} + static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); @@ -809,6 +918,10 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) else snd_soc_dai_init_dma_data(cpu_dai, NULL, &sai->dma_params); + /* Next settings are not relevant for spdif mode */ + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) + return 0; + cr1_mask = SAI_XCR1_RX_TX; if (STM_SAI_IS_CAPTURE(sai)) cr1 |= SAI_XCR1_RX_TX; @@ -820,10 +933,6 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) sai->synco, sai->synci); } - if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) - memcpy(sai->spdif_status_bits, default_status_bits, - sizeof(default_status_bits)); - cr1_mask |= SAI_XCR1_SYNCEN_MASK; cr1 |= SAI_XCR1_SYNCEN_SET(sai->sync); @@ -861,7 +970,7 @@ static int stm32_sai_pcm_process_spdif(struct snd_pcm_substream *substream, /* Set channel status bit */ byte = frm_cnt >> 3; mask = 1 << (frm_cnt - (byte << 3)); - if (sai->spdif_status_bits[byte] & mask) + if (sai->iec958.status[byte] & mask) *ptr |= 0x04000000; ptr++; @@ -888,6 +997,7 @@ static const struct snd_pcm_hardware stm32_sai_pcm_hw = { static struct snd_soc_dai_driver stm32_sai_playback_dai[] = { { .probe = stm32_sai_dai_probe, + .pcm_new = stm32_sai_pcm_new, .id = 1, /* avoid call to fmt_single_name() */ .playback = { .channels_min = 1, @@ -998,6 +1108,7 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, dev_err(&pdev->dev, "S/PDIF IEC60958 not supported\n"); return -EINVAL; } + stm32_sai_init_iec958_status(sai); sai->spdif = true; sai->master = true; } @@ -1114,6 +1225,7 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) sai->id = (uintptr_t)of_id->data; sai->pdev = pdev; + mutex_init(&sai->ctrl_lock); platform_set_drvdata(pdev, sai); sai->pdata = dev_get_drvdata(pdev->dev.parent); diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index affad46bf188..682ef33afb5f 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -377,7 +377,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ret = clk_prepare_enable(ac97->clk_ac97); if (ret) { dev_err(&pdev->dev, "clk_enable failed: %d\n", ret); - goto err; + goto err_clk_put; } ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops); diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index 774fc6ad2026..2e561e946de2 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -173,7 +173,7 @@ /* Number of slots in frame, minus 1 */ #define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT 16 #define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US 7 -#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK (TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_SHIFT) +#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK (TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT) /* TDM mode slot enable bitmask */ #define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT 8 diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 5197d6b18cb6..98d87801d57a 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -190,14 +190,14 @@ static int tegra_alc5632_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_of_node; ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -210,6 +210,13 @@ static int tegra_alc5632_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&alc5632->util_data); +err_put_cpu_of_node: + of_node_put(tegra_alc5632_dai.cpu_of_node); + tegra_alc5632_dai.cpu_of_node = NULL; + tegra_alc5632_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_alc5632_dai.codec_of_node); + tegra_alc5632_dai.codec_of_node = NULL; err: return ret; } @@ -223,6 +230,12 @@ static int tegra_alc5632_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + of_node_put(tegra_alc5632_dai.cpu_of_node); + tegra_alc5632_dai.cpu_of_node = NULL; + tegra_alc5632_dai.platform_of_node = NULL; + of_node_put(tegra_alc5632_dai.codec_of_node); + tegra_alc5632_dai.codec_of_node = NULL; + return 0; } diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index 0e4805c7b4ca..7081f15302cc 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -264,13 +264,13 @@ static int tegra_rt5677_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_rt5677_dai.platform_of_node = tegra_rt5677_dai.cpu_of_node; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -283,6 +283,13 @@ static int tegra_rt5677_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); +err_put_cpu_of_node: + of_node_put(tegra_rt5677_dai.cpu_of_node); + tegra_rt5677_dai.cpu_of_node = NULL; + tegra_rt5677_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_rt5677_dai.codec_of_node); + tegra_rt5677_dai.codec_of_node = NULL; err: return ret; } @@ -296,6 +303,12 @@ static int tegra_rt5677_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + tegra_rt5677_dai.platform_of_node = NULL; + of_node_put(tegra_rt5677_dai.codec_of_node); + tegra_rt5677_dai.codec_of_node = NULL; + of_node_put(tegra_rt5677_dai.cpu_of_node); + tegra_rt5677_dai.cpu_of_node = NULL; + return 0; } diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 638cb3fc5f7b..9bcba06ba52e 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -265,6 +265,57 @@ void aio_port_reset(struct uniphier_aio_sub *sub) } /** + * aio_port_set_ch - set channels of LPCM + * @sub: the AIO substream pointer, PCM substream only + * @ch : count of channels + * + * Set suitable slot selecting to input/output port block of AIO. + * + * This function may return error if non-PCM substream. + * + * Return: Zero if successful, otherwise a negative value on error. + */ +static int aio_port_set_ch(struct uniphier_aio_sub *sub) +{ + struct regmap *r = sub->aio->chip->regmap; + u32 slotsel_2ch[] = { + 0, 0, 0, 0, 0, + }; + u32 slotsel_multi[] = { + OPORTMXTYSLOTCTR_SLOTSEL_SLOT0, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT1, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT2, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT3, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT4, + }; + u32 mode, *slotsel; + int i; + + switch (params_channels(&sub->params)) { + case 8: + case 6: + mode = OPORTMXTYSLOTCTR_MODE; + slotsel = slotsel_multi; + break; + case 2: + mode = 0; + slotsel = slotsel_2ch; + break; + default: + return -EINVAL; + } + + for (i = 0; i < AUD_MAX_SLOTSEL; i++) { + regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i), + OPORTMXTYSLOTCTR_MODE, mode); + regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i), + OPORTMXTYSLOTCTR_SLOTSEL_MASK, slotsel[i]); + } + + return 0; +} + +/** * aio_port_set_rate - set sampling rate of LPCM * @sub: the AIO substream pointer, PCM substream only * @rate: Sampling rate in Hz. @@ -276,7 +327,7 @@ void aio_port_reset(struct uniphier_aio_sub *sub) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) +static int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) { struct regmap *r = sub->aio->chip->regmap; struct device *dev = &sub->aio->chip->pdev->dev; @@ -395,7 +446,7 @@ int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_fmt(struct uniphier_aio_sub *sub) +static int aio_port_set_fmt(struct uniphier_aio_sub *sub) { struct regmap *r = sub->aio->chip->regmap; struct device *dev = &sub->aio->chip->pdev->dev; @@ -460,7 +511,7 @@ int aio_port_set_fmt(struct uniphier_aio_sub *sub) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_clk(struct uniphier_aio_sub *sub) +static int aio_port_set_clk(struct uniphier_aio_sub *sub) { struct uniphier_aio_chip *chip = sub->aio->chip; struct device *dev = &sub->aio->chip->pdev->dev; @@ -575,6 +626,10 @@ int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, rate = params_rate(params); } + ret = aio_port_set_ch(sub); + if (ret) + return ret; + ret = aio_port_set_rate(sub, rate); if (ret) return ret; @@ -731,15 +786,28 @@ void aio_port_set_volume(struct uniphier_aio_sub *sub, int vol) int aio_if_set_param(struct uniphier_aio_sub *sub, int pass_through) { struct regmap *r = sub->aio->chip->regmap; - u32 v; + u32 memfmt, v; if (sub->swm->dir == PORT_DIR_OUTPUT) { - if (pass_through) + if (pass_through) { v = PBOUTMXCTR0_ENDIAN_0123 | PBOUTMXCTR0_MEMFMT_STREAM; - else - v = PBOUTMXCTR0_ENDIAN_3210 | - PBOUTMXCTR0_MEMFMT_2CH; + } else { + switch (params_channels(&sub->params)) { + case 2: + memfmt = PBOUTMXCTR0_MEMFMT_2CH; + break; + case 6: + memfmt = PBOUTMXCTR0_MEMFMT_6CH; + break; + case 8: + memfmt = PBOUTMXCTR0_MEMFMT_8CH; + break; + default: + return -EINVAL; + } + v = PBOUTMXCTR0_ENDIAN_3210 | memfmt; + } regmap_write(r, PBOUTMXCTR0(sub->swm->oif.map), v); regmap_write(r, PBOUTMXCTR1(sub->swm->oif.map), 0); diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c index 2d9b7dde2ffa..ee90e6c3937c 100644 --- a/sound/soc/uniphier/aio-cpu.c +++ b/sound/soc/uniphier/aio-cpu.c @@ -219,15 +219,10 @@ static int uniphier_aio_set_pll(struct snd_soc_dai *dai, int pll_id, unsigned int freq_out) { struct uniphier_aio *aio = uniphier_priv(dai); - struct device *dev = &aio->chip->pdev->dev; int ret; if (!is_valid_pll(aio->chip, pll_id)) return -EINVAL; - if (!aio->chip->plls[pll_id].enable) { - dev_err(dev, "PLL(%d) is not implemented\n", pll_id); - return -ENOTSUPP; - } ret = aio_chip_set_pll(aio->chip, pll_id, freq_out); if (ret < 0) diff --git a/sound/soc/uniphier/aio-ld11.c b/sound/soc/uniphier/aio-ld11.c index ab04d3331be9..de962df245ba 100644 --- a/sound/soc/uniphier/aio-ld11.c +++ b/sound/soc/uniphier/aio-ld11.c @@ -286,7 +286,7 @@ static struct snd_soc_dai_driver uniphier_aio_dai_ld11[] = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_48000, .channels_min = 2, - .channels_max = 2, + .channels_max = 8, }, .ops = &uniphier_aio_i2s_ops, }, diff --git a/sound/soc/uniphier/aio-reg.h b/sound/soc/uniphier/aio-reg.h index 45fdc6ae358a..734395dbcffb 100644 --- a/sound/soc/uniphier/aio-reg.h +++ b/sound/soc/uniphier/aio-reg.h @@ -374,6 +374,7 @@ #define OPORTMXTYVOLGAINSTATUS(n, m) (0x42108 + 0x400 * (n) + 0x20 * (m)) #define OPORTMXTYVOLGAINSTATUS_CUR_MASK GENMASK(15, 0) #define OPORTMXTYSLOTCTR(n, m) (0x42114 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYSLOTCTR_MODE BIT(15) #define OPORTMXTYSLOTCTR_SLOTSEL_MASK GENMASK(11, 8) #define OPORTMXTYSLOTCTR_SLOTSEL_SLOT0 (0x8 << 8) #define OPORTMXTYSLOTCTR_SLOTSEL_SLOT1 (0x9 << 8) diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index aa89c2f6fa24..ca6ccbae0ee8 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -141,6 +141,9 @@ enum IEC61937_PC { #define AUD_MIN_FRAGMENT_SIZE (4 * 1024) #define AUD_MAX_FRAGMENT_SIZE (16 * 1024) +/* max 5 slots, 10 channels, 2 channel in 1 slot */ +#define AUD_MAX_SLOTSEL 5 + /* * This is a selector for virtual register map of AIO. * @@ -322,9 +325,6 @@ int aio_chip_set_pll(struct uniphier_aio_chip *chip, int pll_id, void aio_chip_init(struct uniphier_aio_chip *chip); int aio_init(struct uniphier_aio_sub *sub); void aio_port_reset(struct uniphier_aio_sub *sub); -int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate); -int aio_port_set_fmt(struct uniphier_aio_sub *sub); -int aio_port_set_clk(struct uniphier_aio_sub *sub); int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, const struct snd_pcm_hw_params *params); void aio_port_set_enable(struct uniphier_aio_sub *sub, int enable); diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c index dc955272f58b..389272eeba9a 100644 --- a/sound/soc/zte/zx-tdm.c +++ b/sound/soc/zte/zx-tdm.c @@ -144,8 +144,8 @@ static void zx_tdm_rx_dma_en(struct zx_tdm_info *tdm, bool on) #define ZX_TDM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) #define ZX_TDM_FMTBIT \ - (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_MU_LAW | \ - SNDRV_PCM_FORMAT_A_LAW) + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_MU_LAW | \ + SNDRV_PCM_FMTBIT_A_LAW) static int zx_tdm_dai_probe(struct snd_soc_dai *dai) { diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index b840ff2dcfbb..64f3141a3e1b 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -163,20 +163,3 @@ int snd_emux_free(struct snd_emux *emu) } EXPORT_SYMBOL(snd_emux_free); - - -/* - * INIT part - */ - -static int __init alsa_emux_init(void) -{ - return 0; -} - -static void __exit alsa_emux_exit(void) -{ -} - -module_init(alsa_emux_init) -module_exit(alsa_emux_exit) diff --git a/sound/synth/util_mem.c b/sound/synth/util_mem.c index 8e34bc4e07ec..4bd1e98200d2 100644 --- a/sound/synth/util_mem.c +++ b/sound/synth/util_mem.c @@ -193,19 +193,3 @@ EXPORT_SYMBOL(snd_util_mem_avail); EXPORT_SYMBOL(__snd_util_mem_alloc); EXPORT_SYMBOL(__snd_util_mem_free); EXPORT_SYMBOL(__snd_util_memblk_new); - -/* - * INIT part - */ - -static int __init alsa_util_mem_init(void) -{ - return 0; -} - -static void __exit alsa_util_mem_exit(void) -{ -} - -module_init(alsa_util_mem_init) -module_exit(alsa_util_mem_exit) diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 2dd2518a71d3..f8ef3e2a8ca0 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -565,7 +565,6 @@ static const struct snd_pcm_ops pcm_ops = { .trigger = usb6fire_pcm_trigger, .pointer = usb6fire_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static void usb6fire_pcm_init_urb(struct pcm_urb *urb, diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 05440e2df8d9..d330f74c90e6 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -13,6 +13,7 @@ snd-usb-audio-objs := card.o \ mixer_scarlett.o \ mixer_us16x08.o \ pcm.o \ + power.o \ proc.o \ quirks.o \ stream.o diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index f35d29f49ffe..c6108a3d7f8f 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -348,7 +348,6 @@ static const struct snd_pcm_ops snd_usb_caiaq_ops = { .trigger = snd_usb_caiaq_pcm_trigger, .pointer = snd_usb_caiaq_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static void check_for_elapsed_periods(struct snd_usb_caiaqdev *cdev, @@ -636,6 +635,7 @@ static void read_completed(struct urb *urb) struct device *dev; struct urb *out = NULL; int i, frame, len, send_it = 0, outframe = 0; + unsigned long flags; size_t offset = 0; if (urb->status || !info) @@ -672,10 +672,10 @@ static void read_completed(struct urb *urb) offset += len; if (len > 0) { - spin_lock(&cdev->spinlock); + spin_lock_irqsave(&cdev->spinlock, flags); fill_out_urb(cdev, out, &out->iso_frame_desc[outframe]); read_in_urb(cdev, urb, &urb->iso_frame_desc[frame]); - spin_unlock(&cdev->spinlock); + spin_unlock_irqrestore(&cdev->spinlock, flags); check_for_elapsed_periods(cdev, cdev->sub_playback); check_for_elapsed_periods(cdev, cdev->sub_capture); send_it = 1; diff --git a/sound/usb/card.c b/sound/usb/card.c index a1ed798a1c6b..2bfe4e80a6b9 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -809,6 +809,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) if (!chip->num_suspended_intf++) { list_for_each_entry(as, &chip->pcm_list, list) { snd_pcm_suspend_all(as->pcm); + snd_usb_pcm_suspend(as); as->substream[0].need_setup_ep = as->substream[1].need_setup_ep = true; } @@ -824,6 +825,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) { struct snd_usb_audio *chip = usb_get_intfdata(intf); + struct snd_usb_stream *as; struct usb_mixer_interface *mixer; struct list_head *p; int err = 0; @@ -834,6 +836,13 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) return 0; atomic_inc(&chip->active); /* avoid autopm */ + + list_for_each_entry(as, &chip->pcm_list, list) { + err = snd_usb_pcm_resume(as); + if (err < 0) + goto err_out; + } + /* * ALSA leaves material resumption to user space * we just notify and restart the mixers diff --git a/sound/usb/card.h b/sound/usb/card.h index 9b41b7dda84f..ac785d15ced4 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -37,6 +37,7 @@ struct audioformat { struct snd_usb_substream; struct snd_usb_endpoint; +struct snd_usb_power_domain; struct snd_urb_ctx { struct urb *urb; @@ -115,6 +116,7 @@ struct snd_usb_substream { int interface; /* current interface */ int endpoint; /* assigned endpoint */ struct audioformat *cur_audiofmt; /* current audioformat pointer (for hw_params callback) */ + struct snd_usb_power_domain *str_pd; /* UAC3 Power Domain for streaming path */ snd_pcm_format_t pcm_format; /* current audio format (for hw_params callback) */ unsigned int channels; /* current number of channels (for hw_params callback) */ unsigned int channels_max; /* max channels in the all audiofmts */ diff --git a/sound/usb/clock.c b/sound/usb/clock.c index c79749613fa6..db5e39d67a90 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -513,14 +513,28 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, bool writeable; u32 bmControls; + /* First, try to find a valid clock. This may trigger + * automatic clock selection if the current clock is not + * valid. + */ clock = snd_usb_clock_find_source(chip, fmt->protocol, fmt->clock, true); - if (clock < 0) - return clock; + if (clock < 0) { + /* We did not find a valid clock, but that might be + * because the current sample rate does not match an + * external clock source. Try again without validation + * and we will do another validation after setting the + * rate. + */ + clock = snd_usb_clock_find_source(chip, fmt->protocol, + fmt->clock, false); + if (clock < 0) + return clock; + } prev_rate = get_sample_rate_v2v3(chip, iface, fmt->altsetting, clock); if (prev_rate == rate) - return 0; + goto validation; if (fmt->protocol == UAC_VERSION_3) { struct uac3_clock_source_descriptor *cs_desc; @@ -577,6 +591,10 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, snd_usb_set_interface_quirk(dev); } +validation: + /* validate clock after rate change */ + if (!uac_clock_source_is_valid(chip, fmt->protocol, clock)) + return -ENXIO; return 0; } diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index c90607ebe155..d86be8bfe412 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -325,7 +325,6 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) unsigned long flags; struct snd_usb_packet_info *uninitialized_var(packet); struct snd_urb_ctx *ctx = NULL; - struct urb *urb; int err, i; spin_lock_irqsave(&ep->lock, flags); @@ -345,7 +344,6 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) return; list_del_init(&ctx->ready_list); - urb = ctx->urb; /* copy over the length information */ for (i = 0; i < packet->packets; i++) diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 396c317115b1..e1fbb9cc9ea7 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -523,7 +523,6 @@ static const struct snd_pcm_ops pcm_ops = { .trigger = hiface_pcm_trigger, .pointer = hiface_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static int hiface_pcm_init_urb(struct pcm_urb *urb, diff --git a/sound/usb/line6/toneport.c b/sound/usb/line6/toneport.c index 750467fb95db..f47ba94e6f4a 100644 --- a/sound/usb/line6/toneport.c +++ b/sound/usb/line6/toneport.c @@ -367,12 +367,13 @@ static bool toneport_has_source_select(struct usb_line6_toneport *toneport) */ static void toneport_setup(struct usb_line6_toneport *toneport) { - int ticks; + u32 ticks; struct usb_line6 *line6 = &toneport->line6; struct usb_device *usbdev = line6->usbdev; /* sync time on device with host: */ - ticks = (int)get_seconds(); + /* note: 32-bit timestamps overflow in year 2106 */ + ticks = (u32)ktime_get_real_seconds(); line6_write_data(line6, 0x80c6, &ticks, 4); /* enable device: */ diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 2c1aaa3292bf..dcfc546d81b9 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -281,15 +281,16 @@ static void snd_usbmidi_out_urb_complete(struct urb *urb) struct out_urb_context *context = urb->context; struct snd_usb_midi_out_endpoint *ep = context->ep; unsigned int urb_index; + unsigned long flags; - spin_lock(&ep->buffer_lock); + spin_lock_irqsave(&ep->buffer_lock, flags); urb_index = context - ep->urbs; ep->active_urbs &= ~(1 << urb_index); if (unlikely(ep->drain_urbs)) { ep->drain_urbs &= ~(1 << urb_index); wake_up(&ep->drain_wait); } - spin_unlock(&ep->buffer_lock); + spin_unlock_irqrestore(&ep->buffer_lock, flags); if (urb->status < 0) { int err = snd_usbmidi_urb_error(urb); if (err < 0) { diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 386fbfd5c617..a0b6d039017f 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -900,7 +900,6 @@ static const struct snd_pcm_ops capture_pcm_ops = { .trigger = capture_pcm_trigger, .pointer = capture_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops playback_pcm_ops = { @@ -913,7 +912,6 @@ static const struct snd_pcm_ops playback_pcm_ops = { .trigger = playback_pcm_trigger, .pointer = playback_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct uac_format_type_i_discrete_descriptor * diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ca963e94ec03..c63c84b54969 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -675,16 +675,16 @@ static int get_term_name(struct snd_usb_audio *chip, struct usb_audio_term *iter if (term_only) return 0; switch (iterm->type >> 16) { - case UAC_SELECTOR_UNIT: + case UAC3_SELECTOR_UNIT: strcpy(name, "Selector"); return 8; - case UAC1_PROCESSING_UNIT: + case UAC3_PROCESSING_UNIT: strcpy(name, "Process Unit"); return 12; - case UAC1_EXTENSION_UNIT: + case UAC3_EXTENSION_UNIT: strcpy(name, "Ext Unit"); return 8; - case UAC_MIXER_UNIT: + case UAC3_MIXER_UNIT: strcpy(name, "Mixer"); return 5; default: @@ -832,7 +832,7 @@ static int check_input_term(struct mixer_build *state, int id, case UAC_MIXER_UNIT: { struct uac_mixer_unit_descriptor *d = p1; - term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->type = UAC3_MIXER_UNIT << 16; /* virtual type */ term->channels = uac_mixer_unit_bNrChannels(d); term->chconfig = uac_mixer_unit_wChannelConfig(d, protocol); term->name = uac_mixer_unit_iMixer(d); @@ -845,15 +845,25 @@ static int check_input_term(struct mixer_build *state, int id, err = check_input_term(state, d->baSourceID[0], term); if (err < 0) return err; - term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */ term->id = id; term->name = uac_selector_unit_iSelector(d); return 0; } case UAC1_PROCESSING_UNIT: + /* UAC2_EFFECT_UNIT */ + if (protocol == UAC_VERSION_1) + term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ + else /* UAC_VERSION_2 */ + term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */ + /* fall through */ case UAC1_EXTENSION_UNIT: /* UAC2_PROCESSING_UNIT_V2 */ - /* UAC2_EFFECT_UNIT */ + if (protocol == UAC_VERSION_1 && !term->type) + term->type = UAC3_EXTENSION_UNIT << 16; /* virtual type */ + else if (protocol == UAC_VERSION_2 && !term->type) + term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ + /* fall through */ case UAC2_EXTENSION_UNIT_V2: { struct uac_processing_unit_descriptor *d = p1; @@ -869,7 +879,9 @@ static int check_input_term(struct mixer_build *state, int id, id = d->baSourceID[0]; break; /* continue to parse */ } - term->type = d->bDescriptorSubtype << 16; /* virtual type */ + if (!term->type) + term->type = UAC3_EXTENSION_UNIT << 16; /* virtual type */ + term->channels = uac_processing_unit_bNrChannels(d); term->chconfig = uac_processing_unit_wChannelConfig(d, protocol); term->name = uac_processing_unit_iProcessing(d, protocol); @@ -878,7 +890,7 @@ static int check_input_term(struct mixer_build *state, int id, case UAC2_CLOCK_SOURCE: { struct uac_clock_source_descriptor *d = p1; - term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */ term->id = id; term->name = d->iClockSource; return 0; @@ -923,7 +935,7 @@ static int check_input_term(struct mixer_build *state, int id, case UAC3_CLOCK_SOURCE: { struct uac3_clock_source_descriptor *d = p1; - term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */ term->id = id; term->name = le16_to_cpu(d->wClockSourceStr); return 0; @@ -936,7 +948,37 @@ static int check_input_term(struct mixer_build *state, int id, return err; term->channels = err; - term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->type = UAC3_MIXER_UNIT << 16; /* virtual type */ + + return 0; + } + case UAC3_SELECTOR_UNIT: + case UAC3_CLOCK_SELECTOR: { + struct uac_selector_unit_descriptor *d = p1; + /* call recursively to retrieve the channel info */ + err = check_input_term(state, d->baSourceID[0], term); + if (err < 0) + return err; + term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */ + term->id = id; + term->name = 0; /* TODO: UAC3 Class-specific strings */ + + return 0; + } + case UAC3_PROCESSING_UNIT: { + struct uac_processing_unit_descriptor *d = p1; + + if (!d->bNrInPins) + return -EINVAL; + + /* call recursively to retrieve the channel info */ + err = check_input_term(state, d->baSourceID[0], term); + if (err < 0) + return err; + + term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ + term->id = id; + term->name = 0; /* TODO: UAC3 Class-specific strings */ return 0; } @@ -2167,6 +2209,11 @@ struct procunit_info { struct procunit_value_info *values; }; +static struct procunit_value_info undefined_proc_info[] = { + { 0x00, "Control Undefined", 0 }, + { 0 } +}; + static struct procunit_value_info updown_proc_info[] = { { UAC_UD_ENABLE, "Switch", USB_MIXER_BOOLEAN }, { UAC_UD_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 }, @@ -2215,6 +2262,23 @@ static struct procunit_info procunits[] = { { UAC_PROCESS_DYN_RANGE_COMP, "DCR", dcr_proc_info }, { 0 }, }; + +static struct procunit_value_info uac3_updown_proc_info[] = { + { UAC3_UD_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 }, + { 0 } +}; +static struct procunit_value_info uac3_stereo_ext_proc_info[] = { + { UAC3_EXT_WIDTH_CONTROL, "Width Control", USB_MIXER_U8 }, + { 0 } +}; + +static struct procunit_info uac3_procunits[] = { + { UAC3_PROCESS_UP_DOWNMIX, "Up Down", uac3_updown_proc_info }, + { UAC3_PROCESS_STEREO_EXTENDER, "3D Stereo Extender", uac3_stereo_ext_proc_info }, + { UAC3_PROCESS_MULTI_FUNCTION, "Multi-Function", undefined_proc_info }, + { 0 }, +}; + /* * predefined data for extension units */ @@ -2287,8 +2351,16 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, for (valinfo = info->values; valinfo->control; valinfo++) { __u8 *controls = uac_processing_unit_bmControls(desc, state->mixer->protocol); - if (!(controls[valinfo->control / 8] & (1 << ((valinfo->control % 8) - 1)))) - continue; + if (state->mixer->protocol == UAC_VERSION_1) { + if (!(controls[valinfo->control / 8] & + (1 << ((valinfo->control % 8) - 1)))) + continue; + } else { /* UAC_VERSION_2/3 */ + if (!uac_v2v3_control_is_readable(controls[valinfo->control / 8], + valinfo->control)) + continue; + } + map = find_map(state->map, unitid, valinfo->control); if (check_ignored_ctl(map)) continue; @@ -2300,26 +2372,55 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, cval->val_type = valinfo->val_type; cval->channels = 1; + if (state->mixer->protocol > UAC_VERSION_1 && + !uac_v2v3_control_is_writeable(controls[valinfo->control / 8], + valinfo->control)) + cval->master_readonly = 1; + /* get min/max values */ - if (type == UAC_PROCESS_UP_DOWNMIX && cval->control == UAC_UD_MODE_SELECT) { - __u8 *control_spec = uac_processing_unit_specific(desc, state->mixer->protocol); - /* FIXME: hard-coded */ - cval->min = 1; - cval->max = control_spec[0]; - cval->res = 1; - cval->initialized = 1; - } else { - if (type == USB_XU_CLOCK_RATE) { - /* - * E-Mu USB 0404/0202/TrackerPre/0204 - * samplerate control quirk - */ - cval->min = 0; - cval->max = 5; + switch (type) { + case UAC_PROCESS_UP_DOWNMIX: { + bool mode_sel = false; + + switch (state->mixer->protocol) { + case UAC_VERSION_1: + case UAC_VERSION_2: + default: + if (cval->control == UAC_UD_MODE_SELECT) + mode_sel = true; + break; + case UAC_VERSION_3: + if (cval->control == UAC3_UD_MODE_SELECT) + mode_sel = true; + break; + } + + if (mode_sel) { + __u8 *control_spec = uac_processing_unit_specific(desc, + state->mixer->protocol); + cval->min = 1; + cval->max = control_spec[0]; cval->res = 1; cval->initialized = 1; - } else - get_min_max(cval, valinfo->min_value); + break; + } + + get_min_max(cval, valinfo->min_value); + break; + } + case USB_XU_CLOCK_RATE: + /* + * E-Mu USB 0404/0202/TrackerPre/0204 + * samplerate control quirk + */ + cval->min = 0; + cval->max = 5; + cval->res = 1; + cval->initialized = 1; + break; + default: + get_min_max(cval, valinfo->min_value); + break; } kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); @@ -2362,8 +2463,16 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, static int parse_audio_processing_unit(struct mixer_build *state, int unitid, void *raw_desc) { - return build_audio_procunit(state, unitid, raw_desc, - procunits, "Processing Unit"); + switch (state->mixer->protocol) { + case UAC_VERSION_1: + case UAC_VERSION_2: + default: + return build_audio_procunit(state, unitid, raw_desc, + procunits, "Processing Unit"); + case UAC_VERSION_3: + return build_audio_procunit(state, unitid, raw_desc, + uac3_procunits, "Processing Unit"); + } } static int parse_audio_extension_unit(struct mixer_build *state, int unitid, @@ -2509,11 +2618,20 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, cval->res = 1; cval->initialized = 1; - if (state->mixer->protocol == UAC_VERSION_1) + switch (state->mixer->protocol) { + case UAC_VERSION_1: + default: cval->control = 0; - else /* UAC_VERSION_2 */ - cval->control = (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) ? - UAC2_CX_CLOCK_SELECTOR : UAC2_SU_SELECTOR; + break; + case UAC_VERSION_2: + case UAC_VERSION_3: + if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR || + desc->bDescriptorSubtype == UAC3_CLOCK_SELECTOR) + cval->control = UAC2_CX_CLOCK_SELECTOR; + else /* UAC2/3_SELECTOR_UNIT */ + cval->control = UAC2_SU_SELECTOR; + break; + } namelist = kmalloc_array(desc->bNrInPins, sizeof(char *), GFP_KERNEL); if (!namelist) { @@ -2555,12 +2673,22 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); if (!len) { /* no mapping ? */ + switch (state->mixer->protocol) { + case UAC_VERSION_1: + case UAC_VERSION_2: + default: /* if iSelector is given, use it */ - nameid = uac_selector_unit_iSelector(desc); - if (nameid) - len = snd_usb_copy_string_desc(state->chip, nameid, - kctl->id.name, - sizeof(kctl->id.name)); + nameid = uac_selector_unit_iSelector(desc); + if (nameid) + len = snd_usb_copy_string_desc(state->chip, + nameid, kctl->id.name, + sizeof(kctl->id.name)); + break; + case UAC_VERSION_3: + /* TODO: Class-Specific strings not yet supported */ + break; + } + /* ... or pick up the terminal name at next */ if (!len) len = get_term_name(state->chip, &state->oterm, @@ -2570,7 +2698,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); /* and add the proper suffix */ - if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) + if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR || + desc->bDescriptorSubtype == UAC3_CLOCK_SELECTOR) append_ctl_name(kctl, " Clock Source"); else if ((state->oterm.type & 0xff00) == 0x0100) append_ctl_name(kctl, " Capture Source"); @@ -2641,6 +2770,7 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return parse_audio_mixer_unit(state, unitid, p1); case UAC3_CLOCK_SOURCE: return parse_clock_source_unit(state, unitid, p1); + case UAC3_SELECTOR_UNIT: case UAC3_CLOCK_SELECTOR: return parse_audio_selector_unit(state, unitid, p1); case UAC3_FEATURE_UNIT: diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index e02653465e29..3d12af8bf191 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -109,4 +109,6 @@ int snd_usb_get_cur_mix_value(struct usb_mixer_elem_info *cval, extern void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl); +extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl; + #endif /* __USBMIXER_H */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index e82a72fea9a1..cbfb48bdea51 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -47,8 +47,6 @@ #include "mixer_us16x08.h" #include "helper.h" -extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl; - struct std_mono_table { unsigned int unitid, control, cmask; int val_type; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 160f52c4871b..382847154227 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -711,6 +711,54 @@ static int configure_endpoint(struct snd_usb_substream *subs) return ret; } +static int snd_usb_pcm_change_state(struct snd_usb_substream *subs, int state) +{ + int ret; + + if (!subs->str_pd) + return 0; + + ret = snd_usb_power_domain_set(subs->stream->chip, subs->str_pd, state); + if (ret < 0) { + dev_err(&subs->dev->dev, + "Cannot change Power Domain ID: %d to state: %d. Err: %d\n", + subs->str_pd->pd_id, state, ret); + return ret; + } + + return 0; +} + +int snd_usb_pcm_suspend(struct snd_usb_stream *as) +{ + int ret; + + ret = snd_usb_pcm_change_state(&as->substream[0], UAC3_PD_STATE_D2); + if (ret < 0) + return ret; + + ret = snd_usb_pcm_change_state(&as->substream[1], UAC3_PD_STATE_D2); + if (ret < 0) + return ret; + + return 0; +} + +int snd_usb_pcm_resume(struct snd_usb_stream *as) +{ + int ret; + + ret = snd_usb_pcm_change_state(&as->substream[0], UAC3_PD_STATE_D1); + if (ret < 0) + return ret; + + ret = snd_usb_pcm_change_state(&as->substream[1], UAC3_PD_STATE_D1); + if (ret < 0) + return ret; + + return 0; +} + /* * hw_params callback * @@ -755,16 +803,22 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, ret = snd_usb_lock_shutdown(subs->stream->chip); if (ret < 0) return ret; + + ret = snd_usb_pcm_change_state(subs, UAC3_PD_STATE_D0); + if (ret < 0) + goto unlock; + ret = set_format(subs, fmt); - snd_usb_unlock_shutdown(subs->stream->chip); if (ret < 0) - return ret; + goto unlock; subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; subs->need_setup_ep = true; - return 0; + unlock: + snd_usb_unlock_shutdown(subs->stream->chip); + return ret; } /* @@ -821,6 +875,10 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) snd_usb_endpoint_sync_pending_stop(subs->sync_endpoint); snd_usb_endpoint_sync_pending_stop(subs->data_endpoint); + ret = snd_usb_pcm_change_state(subs, UAC3_PD_STATE_D0); + if (ret < 0) + goto unlock; + ret = set_format(subs, subs->cur_audiofmt); if (ret < 0) goto unlock; @@ -1265,6 +1323,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream) int direction = substream->stream; struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[direction]; + int ret; stop_endpoints(subs, true); @@ -1273,7 +1332,10 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream) !snd_usb_lock_shutdown(subs->stream->chip)) { usb_set_interface(subs->dev, subs->interface, 0); subs->interface = -1; + ret = snd_usb_pcm_change_state(subs, UAC3_PD_STATE_D1); snd_usb_unlock_shutdown(subs->stream->chip); + if (ret < 0) + return ret; } subs->pcm_substream = NULL; @@ -1632,6 +1694,7 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea switch (cmd) { case SNDRV_PCM_TRIGGER_START: subs->trigger_tstamp_pending_update = true; + /* fall through */ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: subs->data_endpoint->prepare_data_urb = prepare_playback_urb; subs->data_endpoint->retire_data_urb = retire_playback_urb; @@ -1694,7 +1757,6 @@ static const struct snd_pcm_ops snd_usb_playback_ops = { .trigger = snd_usb_substream_playback_trigger, .pointer = snd_usb_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops snd_usb_capture_ops = { @@ -1707,7 +1769,6 @@ static const struct snd_pcm_ops snd_usb_capture_ops = { .trigger = snd_usb_substream_capture_trigger, .pointer = snd_usb_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct snd_pcm_ops snd_usb_playback_dev_ops = { diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h index f77ec58bf1a1..9833627c1eca 100644 --- a/sound/usb/pcm.h +++ b/sound/usb/pcm.h @@ -6,6 +6,8 @@ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, unsigned int rate); void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream); +int snd_usb_pcm_suspend(struct snd_usb_stream *as); +int snd_usb_pcm_resume(struct snd_usb_stream *as); int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, diff --git a/sound/usb/power.c b/sound/usb/power.c new file mode 100644 index 000000000000..bd303a1ba1b7 --- /dev/null +++ b/sound/usb/power.c @@ -0,0 +1,104 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * UAC3 Power Domain state management functions + */ + +#include <linux/slab.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <linux/usb/audio-v2.h> +#include <linux/usb/audio-v3.h> + +#include "usbaudio.h" +#include "helper.h" +#include "power.h" + +struct snd_usb_power_domain * +snd_usb_find_power_domain(struct usb_host_interface *ctrl_iface, + unsigned char id) +{ + struct snd_usb_power_domain *pd; + void *p; + + pd = kzalloc(sizeof(*pd), GFP_KERNEL); + if (!pd) + return NULL; + + p = NULL; + while ((p = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + p, UAC3_POWER_DOMAIN)) != NULL) { + struct uac3_power_domain_descriptor *pd_desc = p; + int i; + + for (i = 0; i < pd_desc->bNrEntities; i++) { + if (pd_desc->baEntityID[i] == id) { + pd->pd_id = pd_desc->bPowerDomainID; + pd->pd_d1d0_rec = + le16_to_cpu(pd_desc->waRecoveryTime1); + pd->pd_d2d0_rec = + le16_to_cpu(pd_desc->waRecoveryTime2); + return pd; + } + } + } + + kfree(pd); + return NULL; +} + +int snd_usb_power_domain_set(struct snd_usb_audio *chip, + struct snd_usb_power_domain *pd, + unsigned char state) +{ + struct usb_device *dev = chip->dev; + unsigned char current_state; + int err, idx; + + idx = snd_usb_ctrl_intf(chip) | (pd->pd_id << 8); + + err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), + UAC2_CS_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + UAC3_AC_POWER_DOMAIN_CONTROL << 8, idx, + ¤t_state, sizeof(current_state)); + if (err < 0) { + dev_err(&dev->dev, "Can't get UAC3 power state for id %d\n", + pd->pd_id); + return err; + } + + if (current_state == state) { + dev_dbg(&dev->dev, "UAC3 power domain id %d already in state %d\n", + pd->pd_id, state); + return 0; + } + + err = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), UAC2_CS_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + UAC3_AC_POWER_DOMAIN_CONTROL << 8, idx, + &state, sizeof(state)); + if (err < 0) { + dev_err(&dev->dev, "Can't set UAC3 power state to %d for id %d\n", + state, pd->pd_id); + return err; + } + + if (state == UAC3_PD_STATE_D0) { + switch (current_state) { + case UAC3_PD_STATE_D2: + udelay(pd->pd_d2d0_rec * 50); + break; + case UAC3_PD_STATE_D1: + udelay(pd->pd_d1d0_rec * 50); + break; + default: + return -EINVAL; + } + } + + dev_dbg(&dev->dev, "UAC3 power domain id %d change to state %d\n", + pd->pd_id, state); + + return 0; +} diff --git a/sound/usb/power.h b/sound/usb/power.h index b2e25f60c5a2..6004231a7c75 100644 --- a/sound/usb/power.h +++ b/sound/usb/power.h @@ -2,6 +2,25 @@ #ifndef __USBAUDIO_POWER_H #define __USBAUDIO_POWER_H +struct snd_usb_power_domain { + int pd_id; /* UAC3 Power Domain ID */ + int pd_d1d0_rec; /* D1 to D0 recovery time */ + int pd_d2d0_rec; /* D2 to D0 recovery time */ +}; + +enum { + UAC3_PD_STATE_D0, + UAC3_PD_STATE_D1, + UAC3_PD_STATE_D2, +}; + +int snd_usb_power_domain_set(struct snd_usb_audio *chip, + struct snd_usb_power_domain *pd, + unsigned char state); +struct snd_usb_power_domain * +snd_usb_find_power_domain(struct usb_host_interface *ctrl_iface, + unsigned char id); + #ifdef CONFIG_PM int snd_usb_autoresume(struct snd_usb_audio *chip); void snd_usb_autosuspend(struct snd_usb_audio *chip); diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8aac48f9c322..08aa78007020 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2875,7 +2875,8 @@ YAMAHA_DEVICE(0x7010, "UB99"), */ #define AU0828_DEVICE(vid, pid, vname, pname) { \ - USB_DEVICE_VENDOR_SPEC(vid, pid), \ + .idVendor = vid, \ + .idProduct = pid, \ .match_flags = USB_DEVICE_ID_MATCH_DEVICE | \ USB_DEVICE_ID_MATCH_INT_CLASS | \ USB_DEVICE_ID_MATCH_INT_SUBCLASS, \ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc02767f..8a945ece9869 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1213,7 +1213,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, if (err < 0) return err; - mdelay(20); /* Delay needed after setting the interface */ + msleep(20); /* Delay needed after setting the interface */ /* Vendor mode switch cmd is required. */ if (fmt->formats & SNDRV_PCM_FMTBIT_DSD_U32_BE) { @@ -1234,7 +1234,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, return err; } - mdelay(20); + msleep(20); } return 0; } @@ -1281,7 +1281,7 @@ void snd_usb_set_interface_quirk(struct usb_device *dev) switch (USB_ID_VENDOR(chip->usb_id)) { case 0x23ba: /* Playback Design */ case 0x0644: /* TEAC Corp. */ - mdelay(50); + msleep(50); break; } } @@ -1301,7 +1301,7 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, */ if (USB_ID_VENDOR(chip->usb_id) == 0x23ba && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + msleep(20); /* * "TEAC Corp." products need a 20ms delay after each @@ -1309,14 +1309,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, */ if (USB_ID_VENDOR(chip->usb_id) == 0x0644 && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + msleep(20); /* ITF-USB DSD based DACs functionality need a delay * after each class compliant request */ if (is_itf_usb_dsd_dac(chip->usb_id) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + msleep(20); /* Zoom R16/24, Logitech H650e, Jabra 550a needs a tiny delay here, * otherwise requests like get/set frequency return as failed despite @@ -1326,7 +1326,7 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, chip->usb_id == USB_ID(0x046d, 0x0a46) || chip->usb_id == USB_ID(0x0b0e, 0x0349)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(1); + usleep_range(1000, 2000); } /* @@ -1373,6 +1373,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ @@ -1443,6 +1444,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, */ switch (USB_ID_VENDOR(chip->usb_id)) { case 0x20b1: /* XMOS based devices */ + case 0x152a: /* Thesycon devices */ case 0x25ce: /* Mytek devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 729afd808cc4..67cf849aa16b 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -37,6 +37,7 @@ #include "format.h" #include "clock.h" #include "stream.h" +#include "power.h" /* * free a substream @@ -53,6 +54,7 @@ static void free_substream(struct snd_usb_substream *subs) kfree(fp); } kfree(subs->rate_list.list); + kfree(subs->str_pd); } @@ -82,7 +84,8 @@ static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) static void snd_usb_init_substream(struct snd_usb_stream *as, int stream, - struct audioformat *fp) + struct audioformat *fp, + struct snd_usb_power_domain *pd) { struct snd_usb_substream *subs = &as->substream[stream]; @@ -107,6 +110,13 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, if (fp->channels > subs->channels_max) subs->channels_max = fp->channels; + if (pd) { + subs->str_pd = pd; + /* Initialize Power Domain to idle status D1 */ + snd_usb_power_domain_set(subs->stream->chip, pd, + UAC3_PD_STATE_D1); + } + snd_usb_preallocate_buffer(subs); } @@ -468,9 +478,11 @@ snd_pcm_chmap_elem *convert_chmap_v3(struct uac3_cluster_header_descriptor * fmt_list and will be freed on the chip instance release. do not free * fp or do remove it from the substream fmt_list to avoid double-free. */ -int snd_usb_add_audio_stream(struct snd_usb_audio *chip, - int stream, - struct audioformat *fp) +static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, + int stream, + struct audioformat *fp, + struct snd_usb_power_domain *pd) + { struct snd_usb_stream *as; struct snd_usb_substream *subs; @@ -498,7 +510,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) return err; - snd_usb_init_substream(as, stream, fp); + snd_usb_init_substream(as, stream, fp, pd); return add_chmap(as->pcm, stream, subs); } @@ -526,7 +538,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, else strcpy(pcm->name, "USB Audio"); - snd_usb_init_substream(as, stream, fp); + snd_usb_init_substream(as, stream, fp, pd); /* * Keep using head insertion for M-Audio Audiophile USB (tm) which has a @@ -544,6 +556,21 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, return add_chmap(pcm, stream, &as->substream[stream]); } +int snd_usb_add_audio_stream(struct snd_usb_audio *chip, + int stream, + struct audioformat *fp) +{ + return __snd_usb_add_audio_stream(chip, stream, fp, NULL); +} + +static int snd_usb_add_audio_stream_v3(struct snd_usb_audio *chip, + int stream, + struct audioformat *fp, + struct snd_usb_power_domain *pd) +{ + return __snd_usb_add_audio_stream(chip, stream, fp, pd); +} + static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, struct usb_host_interface *alts, int protocol, int iface_no) @@ -819,6 +846,7 @@ found_clock: static struct audioformat * snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, struct usb_host_interface *alts, + struct snd_usb_power_domain **pd_out, int iface_no, int altset_idx, int altno, int stream) { @@ -829,6 +857,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, struct uac3_as_header_descriptor *as = NULL; struct uac3_hc_descriptor_header hc_header; struct snd_pcm_chmap_elem *chmap; + struct snd_usb_power_domain *pd; unsigned char badd_profile; u64 badd_formats = 0; unsigned int num_channels; @@ -1008,12 +1037,28 @@ found_clock: fp->rate_max = UAC3_BADD_SAMPLING_RATE; fp->rates = SNDRV_PCM_RATE_CONTINUOUS; + pd = kzalloc(sizeof(*pd), GFP_KERNEL); + if (!pd) { + kfree(fp->rate_table); + kfree(fp); + return NULL; + } + pd->pd_id = (stream == SNDRV_PCM_STREAM_PLAYBACK) ? + UAC3_BADD_PD_ID10 : UAC3_BADD_PD_ID11; + pd->pd_d1d0_rec = UAC3_BADD_PD_RECOVER_D1D0; + pd->pd_d2d0_rec = UAC3_BADD_PD_RECOVER_D2D0; + } else { fp->attributes = parse_uac_endpoint_attributes(chip, alts, UAC_VERSION_3, iface_no); + + pd = snd_usb_find_power_domain(chip->ctrl_intf, + as->bTerminalLink); + /* ok, let's parse further... */ if (snd_usb_parse_audio_format_v3(chip, fp, as, stream) < 0) { + kfree(pd); kfree(fp->chmap); kfree(fp->rate_table); kfree(fp); @@ -1021,6 +1066,9 @@ found_clock: } } + if (pd) + *pd_out = pd; + return fp; } @@ -1032,6 +1080,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) struct usb_interface_descriptor *altsd; int i, altno, err, stream; struct audioformat *fp = NULL; + struct snd_usb_power_domain *pd = NULL; int num, protocol; dev = chip->dev; @@ -1114,7 +1163,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) break; } case UAC_VERSION_3: - fp = snd_usb_get_audioformat_uac3(chip, alts, + fp = snd_usb_get_audioformat_uac3(chip, alts, &pd, iface_no, i, altno, stream); break; } @@ -1125,9 +1174,14 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) return PTR_ERR(fp); dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint); - err = snd_usb_add_audio_stream(chip, stream, fp); + if (protocol == UAC_VERSION_3) + err = snd_usb_add_audio_stream_v3(chip, stream, fp, pd); + else + err = snd_usb_add_audio_stream(chip, stream, fp); + if (err < 0) { list_del(&fp->list); /* unlink for avoiding double-free */ + kfree(pd); kfree(fp->rate_table); kfree(fp->chmap); kfree(fp); diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 4ed9d0c41843..fa7dca5a68c8 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -290,7 +290,6 @@ static void had_reset_audio(struct snd_intelhad *intelhaddata) static int had_prog_status_reg(struct snd_pcm_substream *substream, struct snd_intelhad *intelhaddata) { - union aud_cfg cfg_val = {.regval = 0}; union aud_ch_status_0 ch_stat0 = {.regval = 0}; union aud_ch_status_1 ch_stat1 = {.regval = 0}; @@ -298,7 +297,6 @@ static int had_prog_status_reg(struct snd_pcm_substream *substream, IEC958_AES0_NONAUDIO) >> 1; ch_stat0.regx.clk_acc = (intelhaddata->aes_bits & IEC958_AES3_CON_CLOCK) >> 4; - cfg_val.regx.val_bit = ch_stat0.regx.lpcm_id; switch (substream->runtime->rate) { case AUD_SAMPLE_RATE_32: @@ -1854,7 +1852,7 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) /* setup private data which can be retrieved when required */ pcm->private_data = ctx; pcm->info_flags = 0; - strncpy(pcm->name, card->shortname, strlen(card->shortname)); + strlcpy(pcm->name, card->shortname, strlen(card->shortname)); /* setup the ops for playabck */ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &had_pcm_ops); diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c index 5a2bd70a2fa1..129180e17db1 100644 --- a/sound/xen/xen_snd_front_alsa.c +++ b/sound/xen/xen_snd_front_alsa.c @@ -188,7 +188,7 @@ static u64 to_sndif_formats_mask(u64 alsa_formats) mask = 0; for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) - if (1 << ALSA_SNDIF_FORMATS[i].alsa & alsa_formats) + if (pcm_format_to_bits(ALSA_SNDIF_FORMATS[i].alsa) & alsa_formats) mask |= 1 << ALSA_SNDIF_FORMATS[i].sndif; return mask; @@ -202,7 +202,7 @@ static u64 to_alsa_formats_mask(u64 sndif_formats) mask = 0; for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) if (1 << ALSA_SNDIF_FORMATS[i].sndif & sndif_formats) - mask |= 1 << ALSA_SNDIF_FORMATS[i].alsa; + mask |= pcm_format_to_bits(ALSA_SNDIF_FORMATS[i].alsa); return mask; } |