diff options
| author | Linus Torvalds <torvalds@linux-foundation.org> | 2014-08-07 07:07:24 +0400 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2014-08-07 07:07:24 +0400 |
| commit | 930e0312bcdc96d15f02ed6812d4a6c947855a2d (patch) | |
| tree | d2d620c06359510562b25987cf329c77e41b7c11 | |
| parent | ec6c0a77786524e44003e70ea69651ad7fb35aec (diff) | |
| parent | a509574e5ea7b617268943526773ebf7e2d20a9b (diff) | |
| download | linux-930e0312bcdc96d15f02ed6812d4a6c947855a2d.tar.xz | |
Merge tag 'sound-3.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There've been many updates in ASoC side at this time, especially the
framework enhancement for multiple CODECs on a single DAI and more
componentization works.
The only major change in ALSA core is the addition of timestamp type
in sw_params field. This should behave in backward compatible way.
Other than that, there are lots of small changes and new drivers in
wide range, including a large code cut in HD-audio driver for
deprecated static quirks. Some highlights are below:
ALSA Core:
- Add the new timestamp type field to sw_params to choose
MONOTONIC_RAW type
HD-audio:
- Continued conversion to standard printk macros, generic code
cleanups
- Removal of obsoleted static quirk codes for Conexant and C-Media
codecs
- Fixups for HP Envy TS, Dell XPS 15, HP and Dell mute/mic LED,
Gigabyte BXBT-2807 mobo
- Intel Braswell support
ASoC:
- Support for multiple CODECs attached to a single DAI, enabling
systems with for example multiple DAC/speaker drivers on a single
link, contributed by Benoit Cousson based on work from Misael Lopez
Cruz
- Support for byte controls larger than 256 bytes based on the use of
TLVs contributed by Omair Mohammed Abdullah
- More componentisation work from Lars-Peter Clausen
- The remainder of the conversions of CODEC drivers to params_width()
by Mark Brown
- Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks,
Realtek RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas
Instruments TAS2552
- Lots of updates and fixes, especially to the DaVinci, Intel,
Freescale, Realtek, and rcar drivers"
* tag 'sound-3.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (402 commits)
ALSA: usb-audio: Whitespace cleanups for sound/usb/midi.*
ALSA: usb-audio: Respond to suspend and resume callbacks for MIDI input
sound/oss/pss: Remove typedefs pss_mixerdata and pss_confdata
sound/oss/opl3: Remove typedef opl_devinfo
ALSA: fireworks: fix specifiers in format strings for propper output
ASoC: imx-audmux: Use uintptr_t for port numbers
ASoC: davinci: Enable menuconfig entry for McASP
ASoC: fsl_asrc: Don't access members of config before checking it
ASoC: fsl_sarc_dma: Check pair before using it
ASoC: adau1977: Fix truncation warning on 64 bit architectures
ALSA: virtuoso: add Xonar Essence STX II support
ALSA: riptide: fix %d confusingly prefixed with 0x in format strings
ALSA: fireworks: fix %d confusingly prefixed with 0x in format strings
ALSA: hda - add codec ID for Braswell display audio codec
ALSA: hda - add PCI IDs for Intel Braswell
ALSA: usb-audio: Adjust Gamecom 780 volume level
ALSA: usb-audio: improve dmesg source grepability
ASoC: rt5670: Fix duplicate const warnings
ASoC: rt5670: Staticise non-exported symbols
ASoC: Intel: update stream only on stream IPC msgs
...
294 files changed, 17264 insertions, 7462 deletions
diff --git a/Documentation/devicetree/bindings/sound/ak5386.txt b/Documentation/devicetree/bindings/sound/ak5386.txt index dc3914fe6ce8..ec3df3abba0c 100644 --- a/Documentation/devicetree/bindings/sound/ak5386.txt +++ b/Documentation/devicetree/bindings/sound/ak5386.txt @@ -10,10 +10,14 @@ Optional properties: - reset-gpio : a GPIO spec for the reset/power down pin. If specified, it will be deasserted at probe time. + - va-supply : a regulator spec, providing 5.0V + - vd-supply : a regulator spec, providing 3.3V Example: spdif: ak5386@0 { compatible = "asahi-kasei,ak5386"; reset-gpio = <&gpio0 23>; + va-supply = <&vdd_5v0_reg>; + vd-supply = <&vdd_3v3_reg>; }; diff --git a/Documentation/devicetree/bindings/sound/cs4265.txt b/Documentation/devicetree/bindings/sound/cs4265.txt new file mode 100644 index 000000000000..380fff8e4e83 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4265.txt @@ -0,0 +1,29 @@ +CS4265 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "cirrus,cs4265" + + - reg : the I2C address of the device for I2C. The I2C address depends on + the state of the AD0 pin. If AD0 is high, the i2c address is 0x4f. + If it is low, the i2c address is 0x4e. + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + +Examples: + +codec_ad0_high: cs4265@4f { /* AD0 Pin is high */ + compatible = "cirrus,cs4265"; + reg = <0x4f>; +}; + + +codec_ad0_low: cs4265@4e { /* AD0 Pin is low */ + compatible = "cirrus,cs4265"; + reg = <0x4e>; +}; diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt new file mode 100644 index 000000000000..b93362a570be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt @@ -0,0 +1,60 @@ +Freescale Asynchronous Sample Rate Converter (ASRC) Controller + +The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of a +signal associated with an input clock into a signal associated with a different +output clock. The driver currently works as a Front End of DPCM with other Back +Ends Audio controller such as ESAI, SSI and SAI. It has three pairs to support +three substreams within totally 10 channels. + +Required properties: + + - compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc". + + - reg : Offset and length of the register set for the device. + + - interrupts : Contains the spdif interrupt. + + - dmas : Generic dma devicetree binding as described in + Documentation/devicetree/bindings/dma/dma.txt. + + - dma-names : Contains "rxa", "rxb", "rxc", "txa", "txb" and "txc". + + - clocks : Contains an entry for each entry in clock-names. + + - clock-names : Contains the following entries + "mem" Peripheral access clock to access registers. + "ipg" Peripheral clock to driver module. + "asrck_<0-f>" Clock sources for input and output clock. + + - big-endian : If this property is absent, the little endian mode + will be in use as default. Otherwise, the big endian + mode will be in use for all the device registers. + + - fsl,asrc-rate : Defines a mutual sample rate used by DPCM Back Ends. + + - fsl,asrc-width : Defines a mutual sample width used by DPCM Back Ends. + +Example: + +asrc: asrc@02034000 { + compatible = "fsl,imx53-asrc"; + reg = <0x02034000 0x4000>; + interrupts = <0 50 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clks 107>, <&clks 107>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 107>, <&clks 0>, <&clks 0>; + clock-names = "mem", "ipg", "asrck0", + "asrck_1", "asrck_2", "asrck_3", "asrck_4", + "asrck_5", "asrck_6", "asrck_7", "asrck_8", + "asrck_9", "asrck_a", "asrck_b", "asrck_c", + "asrck_d", "asrck_e", "asrck_f"; + dmas = <&sdma 17 23 1>, <&sdma 18 23 1>, <&sdma 19 23 1>, + <&sdma 20 23 1>, <&sdma 21 23 1>, <&sdma 22 23 1>; + dma-names = "rxa", "rxb", "rxc", + "txa", "txb", "txc"; + fsl,asrc-rate = <48000>; + fsl,asrc-width = <16>; + status = "okay"; +}; diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index a5e63fa47dc5..c454e67f54bb 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -4,7 +4,7 @@ This device supports I2C only. Required properties: -- compatible : "maxim,max98090". +- compatible : "maxim,max98090" or "maxim,max98091". - reg : The I2C address of the device. diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 8346cab046cd..aa697abf337e 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -13,6 +13,9 @@ Required properties: - rcar_sound,src : Should contain SRC feature. The number of SRC subnode should be same as HW. see below for detail. +- rcar_sound,dvc : Should contain DVC feature. + The number of DVC subnode should be same as HW. + see below for detail. - rcar_sound,dai : DAI contents. The number of DAI subnode should be same as HW. see below for detail. @@ -21,6 +24,7 @@ SSI subnode properties: - interrupts : Should contain SSI interrupt for PIO transfer - shared-pin : if shared clock pin - pio-transfer : use PIO transfer mode +- no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case SRC subnode properties: no properties at this point @@ -39,6 +43,11 @@ rcar_sound: rcar_sound@0xffd90000 { <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x1280>; /* SSI */ + rcar_sound,dvc { + dvc0: dvc@0 { }; + dvc1: dvc@1 { }; + }; + rcar_sound,src { src0: src@0 { }; src1: src@1 { }; diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt new file mode 100644 index 000000000000..6c55fcfe5e1d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -0,0 +1,37 @@ +* Rockchip I2S controller + +The I2S bus (Inter-IC sound bus) is a serial link for digital +audio data transfer between devices in the system. + +Required properties: + +- compatible: should be one of the followings + - "rockchip,rk3066-i2s": for rk3066 + - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 + - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288 +- reg: physical base address of the controller and length of memory mapped + region. +- interrupts: should contain the I2S interrupt. +- #address-cells: should be 1. +- #size-cells: should be 0. +- dmas: DMA specifiers for tx and rx dma. See the DMA client binding, + Documentation/devicetree/bindings/dma/dma.txt +- dma-names: should include "tx" and "rx". +- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. +- clock-names: should contain followings: + - "i2s_hclk": clock for I2S BUS + - "i2s_clk" : clock for I2S controller + +Example for rk3288 I2S controller: + +i2s@ff890000 { + compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; + reg = <0xff890000 0x10000>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + #address-cells = <1>; + #size-cells = <0>; + dmas = <&pdma1 0>, <&pdma1 1>; + dma-names = "rx", "tx"; + clock-names = "i2s_hclk", "i2s_clk"; + clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>; +}; diff --git a/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt new file mode 100644 index 000000000000..9148f72319e1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt @@ -0,0 +1,35 @@ +Samsung Exynos Odroid X2/U3 audio complex with MAX98090 codec + +Required properties: + - compatible : "samsung,odroidx2-audio" - for Odroid X2 board, + "samsung,odroidu3-audio" - for Odroid U3 board + - samsung,model : the user-visible name of this sound complex + - samsung,i2s-controller : the phandle of the I2S controller + - samsung,audio-codec : the phandle of the MAX98090 audio codec + - samsung,audio-routing : 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 for sources and sinks are the MAX98090's pins (as + documented in its binding), and the jacks on the board + For Odroid X2: + * Headphone Jack + * Mic Jack + * DMIC + + For Odroid U3: + * Headphone Jack + * Speakers + +Example: + +sound { + compatible = "samsung,odroidu3-audio"; + samsung,i2s-controller = <&i2s0>; + samsung,audio-codec = <&max98090>; + samsung,model = "Odroid-X2"; + samsung,audio-routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "IN1", "Mic Jack", + "Mic Jack", "MICBIAS"; +}; diff --git a/Documentation/devicetree/bindings/sound/sirf-usp.txt b/Documentation/devicetree/bindings/sound/sirf-usp.txt new file mode 100644 index 000000000000..02f85b32d359 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sirf-usp.txt @@ -0,0 +1,27 @@ +* SiRF SoC USP module + +Required properties: +- compatible: "sirf,prima2-usp-pcm" +- reg: Base address and size entries: +- dmas: List of DMA controller phandle and DMA request line ordered pairs. +- dma-names: Identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. + + One of the DMA channels will be responsible for transmission (should be + named "tx") and one for reception (should be named "rx"). + +- clocks: USP controller clock source +- pinctrl-names: Must contain a "default" entry. +- pinctrl-NNN: One property must exist for each entry in pinctrl-names. + +Example: +usp0: usp@b0080000 { + compatible = "sirf,prima2-usp-pcm"; + reg = <0xb0080000 0x10000>; + clocks = <&clks 28>; + dmas = <&dmac1 1>, <&dmac1 2>; + dma-names = "rx", "tx"; + pinctrl-names = "default"; + pinctrl-0 = <&usp0_only_utfs_pins_a>; +}; + diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt index 678b191c37b8..6df74f15687f 100644 --- a/Documentation/devicetree/bindings/sound/snow.txt +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -3,15 +3,20 @@ Audio Binding for Snow boards Required properties: - compatible : Can be one of the following, "google,snow-audio-max98090" or + "google,snow-audio-max98091" or "google,snow-audio-max98095" - samsung,i2s-controller: The phandle of the Samsung I2S controller - samsung,audio-codec: The phandle of the audio codec +Optional: +- samsung,model: The name of the sound-card + Example: sound { compatible = "google,snow-audio-max98095"; + samsung,model = "Snow-I2S-MAX98095"; samsung,i2s-controller = <&i2s0>; samsung,audio-codec = <&max98095>; }; diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt new file mode 100644 index 000000000000..55e2a0af5645 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2552.txt @@ -0,0 +1,26 @@ +Texas Instruments - tas2552 Codec module + +The tas2552 serial control bus communicates through I2C protocols + +Required properties: + - compatible - One of: + "ti,tas2552" - TAS2552 + - reg - I2C slave address + - supply-*: Required supply regulators are: + "vbat" battery voltage + "iovdd" I/O Voltage + "avdd" Analog DAC Voltage + +Optional properties: + - enable-gpio - gpio pin to enable/disable the device + +Example: + +tas2552: tas2552@41 { + compatible = "ti,tas2552"; + reg = <0x41>; + enable-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; +}; + +For more product information please see the link below: +http://www.ti.com/product/TAS2552 diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt index d2866a0d6a26..234dad296da7 100644 --- a/Documentation/devicetree/bindings/sound/ti,tas5086.txt +++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt @@ -31,6 +31,9 @@ Optional properties: Most systems should not set any of these properties. + - avdd-supply: Power supply for AVDD, providing 3.3V + - dvdd-supply: Power supply for DVDD, providing 3.3V + Examples: i2c_bus { @@ -39,5 +42,7 @@ Examples: reg = <0x1b>; reset-gpio = <&gpio 23 0>; ti,charge-period = <156000>; + avdd-supply = <&vdd_3v3_reg>; + dvdd-supply = <&vdd_3v3_reg>; }; }; diff --git a/Documentation/devicetree/bindings/sound/wm8904.txt b/Documentation/devicetree/bindings/sound/wm8904.txt new file mode 100644 index 000000000000..e99f4097c83c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8904.txt @@ -0,0 +1,33 @@ +WM8904 audio CODEC + +This device supports I2C only. + +Required properties: + - compatible: "wlf,wm8904" + - reg: the I2C address of the device. + - clock-names: "mclk" + - clocks: reference to + <Documentation/devicetree/bindings/clock/clock-bindings.txt> + +Pins on the device (for linking into audio routes): + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * MICBIAS + +Examples: + +codec: wm8904@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + clocks = <&pck0>; + clock-names = "mclk"; +}; diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 7ccf933bfbe0..48148d6d9307 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -2026,8 +2026,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ------------------- Module for sound cards based on the Asus AV66/AV100/AV200 chips, - i.e., Xonar D1, DX, D2, D2X, DS, Essence ST (Deluxe), Essence STX, - HDAV1.3 (Deluxe), and HDAV1.3 Slim. + i.e., Xonar D1, DX, D2, D2X, DS, DSX, Essence ST (Deluxe), + Essence STX (II), HDAV1.3 (Deluxe), and HDAV1.3 Slim. This module supports autoprobe and multiple cards. diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index d1ab5e17eb13..a5e754714344 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -284,6 +284,11 @@ STAC92HD83* hp-zephyr HP Zephyr hp-led HP with broken BIOS for mute LED hp-inv-led HP with broken BIOS for inverted mute LED + hp-mic-led HP with mic-mute LED + headset-jack Dell Latitude with a 4-pin headset jack + hp-envy-bass Pin fixup for HP Envy bass speaker (NID 0x0f) + hp-envy-ts-bass Pin fixup for HP Envy TS bass speaker (NID 0x10) + hp-bnb13-eq Hardware equalizer setup for HP laptops auto BIOS setup (default) STAC92HD95 diff --git a/MAINTAINERS b/MAINTAINERS index 12fee4ef936b..731c8a48e19c 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7526,6 +7526,13 @@ F: drivers/rtc/ F: include/linux/rtc.h F: include/uapi/linux/rtc.h +REALTEK AUDIO CODECS +M: Bard Liao <bardliao@realtek.com> +M: Oder Chiou <oder_chiou@realtek.com> +S: Maintained +F: sound/soc/codecs/rt* +F: include/sound/rt*.h + REISERFS FILE SYSTEM L: reiserfs-devel@vger.kernel.org S: Supported diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 30fcac73a540..689c121157ec 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -998,6 +998,8 @@ static struct platform_device fsi_wm8978_device = { .id = 0, .dev = { .platform_data = &fsi_wm8978_info, + .coherent_dma_mask = DMA_BIT_MASK(32), + .dma_mask = &fsi_wm8978_device.dev.coherent_dma_mask, }, }; @@ -1021,6 +1023,8 @@ static struct platform_device fsi_hdmi_device = { .id = 1, .dev = { .platform_data = &fsi2_hdmi_info, + .coherent_dma_mask = DMA_BIT_MASK(32), + .dma_mask = &fsi_hdmi_device.dev.coherent_dma_mask, }, }; diff --git a/arch/arm/mach-shmobile/board-kzm9g.c b/arch/arm/mach-shmobile/board-kzm9g.c index f94ec8ca42c1..01e0d1386db7 100644 --- a/arch/arm/mach-shmobile/board-kzm9g.c +++ b/arch/arm/mach-shmobile/board-kzm9g.c @@ -603,6 +603,8 @@ static struct platform_device fsi_ak4648_device = { .name = "asoc-simple-card", .dev = { .platform_data = &fsi2_ak4648_info, + .coherent_dma_mask = DMA_BIT_MASK(32), + .dma_mask = &fsi_ak4648_device.dev.coherent_dma_mask, }, }; diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c index 0ff4d8e45cf7..112553f0f9bf 100644 --- a/arch/arm/mach-shmobile/board-mackerel.c +++ b/arch/arm/mach-shmobile/board-mackerel.c @@ -523,6 +523,8 @@ static struct platform_device fsi_hdmi_device = { .id = 1, .dev = { .platform_data = &fsi2_hdmi_info, + .coherent_dma_mask = DMA_BIT_MASK(32), + .dma_mask = &fsi_hdmi_device.dev.coherent_dma_mask, }, }; @@ -919,6 +921,8 @@ static struct platform_device fsi_ak4643_device = { .name = "asoc-simple-card", .dev = { .platform_data = &fsi2_ak4643_info, + .coherent_dma_mask = DMA_BIT_MASK(32), + .dma_mask = &fsi_ak4643_device.dev.coherent_dma_mask, }, }; diff --git a/arch/sh/boards/mach-ecovec24/setup.c b/arch/sh/boards/mach-ecovec24/setup.c index 85d5255d259f..0d3049244cd3 100644 --- a/arch/sh/boards/mach-ecovec24/setup.c +++ b/arch/sh/boards/mach-ecovec24/setup.c @@ -874,6 +874,8 @@ static struct platform_device fsi_da7210_device = { .name = "asoc-simple-card", .dev = { .platform_data = &fsi_da7210_info, + .coherent_dma_mask = DMA_BIT_MASK(32), + .dma_mask = &fsi_da7210_device.dev.coherent_dma_mask, }, }; diff --git a/arch/x86/include/asm/platform_sst_audio.h b/arch/x86/include/asm/platform_sst_audio.h new file mode 100644 index 000000000000..0a4e140315b6 --- /dev/null +++ b/arch/x86/include/asm/platform_sst_audio.h @@ -0,0 +1,78 @@ +/* + * platform_sst_audio.h: sst audio platform data header file + * + * Copyright (C) 2012-14 Intel Corporation + * Author: Jeeja KP <jeeja.kp@intel.com> + * Omair Mohammed Abdullah <omair.m.abdullah@intel.com> + * Vinod Koul ,vinod.koul@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. + */ +#ifndef _PLATFORM_SST_AUDIO_H_ +#define _PLATFORM_SST_AUDIO_H_ + +#include <linux/sfi.h> + +enum sst_audio_task_id_mrfld { + SST_TASK_ID_NONE = 0, + SST_TASK_ID_SBA = 1, + SST_TASK_ID_MEDIA = 3, + SST_TASK_ID_MAX = SST_TASK_ID_MEDIA, +}; + +/* Device IDs for Merrifield are Pipe IDs, + * ref: DSP spec v0.75 */ +enum sst_audio_device_id_mrfld { + /* Output pipeline IDs */ + PIPE_ID_OUT_START = 0x0, + PIPE_CODEC_OUT0 = 0x2, + PIPE_CODEC_OUT1 = 0x3, + PIPE_SPROT_LOOP_OUT = 0x4, + PIPE_MEDIA_LOOP1_OUT = 0x5, + PIPE_MEDIA_LOOP2_OUT = 0x6, + PIPE_VOIP_OUT = 0xC, + PIPE_PCM0_OUT = 0xD, + PIPE_PCM1_OUT = 0xE, + PIPE_PCM2_OUT = 0xF, + PIPE_MEDIA0_OUT = 0x12, + PIPE_MEDIA1_OUT = 0x13, +/* Input Pipeline IDs */ + PIPE_ID_IN_START = 0x80, + PIPE_CODEC_IN0 = 0x82, + PIPE_CODEC_IN1 = 0x83, + PIPE_SPROT_LOOP_IN = 0x84, + PIPE_MEDIA_LOOP1_IN = 0x85, + PIPE_MEDIA_LOOP2_IN = 0x86, + PIPE_VOIP_IN = 0x8C, + PIPE_PCM0_IN = 0x8D, + PIPE_PCM1_IN = 0x8E, + PIPE_MEDIA0_IN = 0x8F, + PIPE_MEDIA1_IN = 0x90, + PIPE_MEDIA2_IN = 0x91, + PIPE_RSVD = 0xFF, +}; + +/* The stream map for each platform consists of an array of the below + * stream map structure. + */ +struct sst_dev_stream_map { + u8 dev_num; /* device id */ + u8 subdev_num; /* substream */ + u8 direction; + u8 device_id; /* fw id */ + u8 task_id; /* fw task */ + u8 status; +}; + +struct sst_platform_data { + /* Intel software platform id*/ + struct sst_dev_stream_map *pdev_strm_map; + unsigned int strm_map_size; +}; + +int add_sst_platform_device(void); +#endif + diff --git a/drivers/dma/edma.c b/drivers/dma/edma.c index d08c4dedef35..b512caf46944 100644 --- a/drivers/dma/edma.c +++ b/drivers/dma/edma.c @@ -982,6 +982,7 @@ static void __init edma_chan_init(struct edma_cc *ecc, #define EDMA_DMA_BUSWIDTHS (BIT(DMA_SLAVE_BUSWIDTH_1_BYTE) | \ BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | \ + BIT(DMA_SLAVE_BUSWIDTH_3_BYTES) | \ BIT(DMA_SLAVE_BUSWIDTH_4_BYTES)) static int edma_dma_device_slave_caps(struct dma_chan *dchan, diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c index 22de13727641..60843a275abd 100644 --- a/drivers/misc/atmel-ssc.c +++ b/drivers/misc/atmel-ssc.c @@ -83,10 +83,17 @@ EXPORT_SYMBOL(ssc_free); static struct atmel_ssc_platform_data at91rm9200_config = { .use_dma = 0, + .has_fslen_ext = 0, +}; + +static struct atmel_ssc_platform_data at91sam9rl_config = { + .use_dma = 0, + .has_fslen_ext = 1, }; static struct atmel_ssc_platform_data at91sam9g45_config = { .use_dma = 1, + .has_fslen_ext = 1, }; static const struct platform_device_id atmel_ssc_devtypes[] = { @@ -94,6 +101,9 @@ static const struct platform_device_id atmel_ssc_devtypes[] = { .name = "at91rm9200_ssc", .driver_data = (unsigned long) &at91rm9200_config, }, { + .name = "at91sam9rl_ssc", + .driver_data = (unsigned long) &at91sam9rl_config, + }, { .name = "at91sam9g45_ssc", .driver_data = (unsigned long) &at91sam9g45_config, }, { @@ -107,6 +117,9 @@ static const struct of_device_id atmel_ssc_dt_ids[] = { .compatible = "atmel,at91rm9200-ssc", .data = &at91rm9200_config, }, { + .compatible = "atmel,at91sam9rl-ssc", + .data = &at91sam9rl_config, + }, { .compatible = "atmel,at91sam9g45-ssc", .data = &at91sam9g45_config, }, { diff --git a/include/linux/atmel-ssc.h b/include/linux/atmel-ssc.h index 571a12ebb018..7c0f6549898b 100644 --- a/include/linux/atmel-ssc.h +++ b/include/linux/atmel-ssc.h @@ -7,6 +7,7 @@ struct atmel_ssc_platform_data { int use_dma; + int has_fslen_ext; }; struct ssc_device { @@ -71,6 +72,12 @@ void ssc_free(struct ssc_device *ssc); #define SSC_RFMR_DATNB_OFFSET 8 #define SSC_RFMR_FSEDGE_SIZE 1 #define SSC_RFMR_FSEDGE_OFFSET 24 +/* + * The FSLEN_EXT exist on at91sam9rl, at91sam9g10, + * at91sam9g20, and at91sam9g45 and newer SoCs + */ +#define SSC_RFMR_FSLEN_EXT_SIZE 4 +#define SSC_RFMR_FSLEN_EXT_OFFSET 28 #define SSC_RFMR_FSLEN_SIZE 4 #define SSC_RFMR_FSLEN_OFFSET 16 #define SSC_RFMR_FSOS_SIZE 4 @@ -109,6 +116,12 @@ void ssc_free(struct ssc_device *ssc); #define SSC_TFMR_FSDEN_OFFSET 23 #define SSC_TFMR_FSEDGE_SIZE 1 #define SSC_TFMR_FSEDGE_OFFSET 24 +/* + * The FSLEN_EXT exist on at91sam9rl, at91sam9g10, + * at91sam9g20, and at91sam9g45 and newer SoCs + */ +#define SSC_TFMR_FSLEN_EXT_SIZE 4 +#define SSC_TFMR_FSLEN_EXT_OFFSET 28 #define SSC_TFMR_FSLEN_SIZE 4 #define SSC_TFMR_FSLEN_OFFSET 16 #define SSC_TFMR_FSOS_SIZE 3 diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index d2c5cc7c583c..3d1c2aa51530 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -299,6 +299,7 @@ enum dma_slave_buswidth { DMA_SLAVE_BUSWIDTH_UNDEFINED = 0, DMA_SLAVE_BUSWIDTH_1_BYTE = 1, DMA_SLAVE_BUSWIDTH_2_BYTES = 2, + DMA_SLAVE_BUSWIDTH_3_BYTES = 3, DMA_SLAVE_BUSWIDTH_4_BYTES = 4, DMA_SLAVE_BUSWIDTH_8_BYTES = 8, }; diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index 6d9371f88875..a614b33d0a39 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -110,6 +110,12 @@ struct arizona { int clk32k_ref; struct snd_soc_dapm_context *dapm; + + int tdm_width[ARIZONA_MAX_AIF]; + int tdm_slots[ARIZONA_MAX_AIF]; + + uint16_t dac_comp_coeff; + uint8_t dac_comp_enabled; }; int arizona_clk32k_enable(struct arizona *arizona); diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h index 709c6f7e2f8c..a6591c693ebb 100644 --- a/include/linux/platform_data/asoc-s3c.h +++ b/include/linux/platform_data/asoc-s3c.h @@ -15,15 +15,6 @@ #define S3C64XX_AC97_GPE 1 extern void s3c64xx_ac97_setup_gpio(int); -/* - * The machine init code calls s5p*_spdif_setup_gpio with - * one of these defines in order to select appropriate bank - * of GPIO for S/PDIF pins - */ -#define S5PC100_SPDIF_GPD 0 -#define S5PC100_SPDIF_GPG3 1 -extern void s5pc100_spdif_setup_gpio(int); - struct samsung_i2s { /* If the Primary DAI has 5.1 Channels */ #define QUIRK_PRI_6CHAN (1 << 0) diff --git a/include/linux/platform_data/dma-imx.h b/include/linux/platform_data/dma-imx.h index bcbc6c3c14c0..d05542aafa3e 100644 --- a/include/linux/platform_data/dma-imx.h +++ b/include/linux/platform_data/dma-imx.h @@ -50,6 +50,7 @@ enum imx_dma_prio { struct imx_dma_data { int dma_request; /* DMA request line */ + int dma_request2; /* secondary DMA request line */ enum sdma_peripheral_type peripheral_type; int priority; }; diff --git a/include/sound/control.h b/include/sound/control.h index 5358892b1b39..042613938a1d 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -31,10 +31,15 @@ typedef int (snd_kcontrol_info_t) (struct snd_kcontrol * kcontrol, struct snd_ct typedef int (snd_kcontrol_get_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol); typedef int (snd_kcontrol_put_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol); typedef int (snd_kcontrol_tlv_rw_t)(struct snd_kcontrol *kcontrol, - int op_flag, /* 0=read,1=write,-1=command */ + int op_flag, /* SNDRV_CTL_TLV_OP_XXX */ unsigned int size, unsigned int __user *tlv); +enum { + SNDRV_CTL_TLV_OP_READ = 0, + SNDRV_CTL_TLV_OP_WRITE = 1, + SNDRV_CTL_TLV_OP_CMD = -1, +}; struct snd_kcontrol_new { snd_ctl_elem_iface_t iface; /* interface identifier */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index d854fb31c000..6f3e10ca0e32 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -931,10 +931,17 @@ void snd_pcm_timer_done(struct snd_pcm_substream *substream); static inline void snd_pcm_gettime(struct snd_pcm_runtime *runtime, struct timespec *tv) { - if (runtime->tstamp_type == SNDRV_PCM_TSTAMP_TYPE_MONOTONIC) + switch (runtime->tstamp_type) { + case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC: ktime_get_ts(tv); - else + break; + case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW: + getrawmonotonic(tv); + break; + default: getnstimeofday(tv); + break; + } } /* diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index f4a706f82cb7..d76412b84b48 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -34,6 +34,7 @@ * B : SSI direction */ #define RSND_SSI_CLK_PIN_SHARE (1 << 31) +#define RSND_SSI_NO_BUSIF (1 << 30) /* SSI+DMA without BUSIF */ #define RSND_SSI(_dma_id, _pio_irq, _flags) \ { .dma_id = _dma_id, .pio_irq = _pio_irq, .flags = _flags } diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 000000000000..eb773d1485f2 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 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_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool gpio2_en; /*GPIO2 enable*/ +}; + +#endif diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h new file mode 100644 index 000000000000..bd311197a3b5 --- /dev/null +++ b/include/sound/rt5670.h @@ -0,0 +1,27 @@ +/* + * linux/sound/rt5670.h -- Platform data for RT5670 + * + * Copyright 2014 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_RT5670_H +#define __LINUX_SND_RT5670_H + +struct rt5670_platform_data { + int jd_mode; + bool in2_diff; + + bool dmic_en; + unsigned int dmic1_data_pin; + /* 0 = GPIO6; 1 = IN2P; 3 = GPIO7*/ + unsigned int dmic2_data_pin; + /* 0 = GPIO8; 1 = IN3N; */ + unsigned int dmic3_data_pin; + /* 0 = GPIO9; 1 = GPIO10; 2 = GPIO5*/ +}; + +#endif diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 688f2ba8009f..e8b3080d196a 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -257,7 +257,6 @@ struct snd_soc_dai { struct snd_soc_dapm_widget *playback_widget; struct snd_soc_dapm_widget *capture_widget; - struct snd_soc_dapm_context dapm; /* DAI DMA data */ void *playback_dma_data; @@ -273,6 +272,10 @@ struct snd_soc_dai { struct snd_soc_codec *codec; struct snd_soc_component *component; + /* CODEC TDM slot masks and params (for fixup) */ + unsigned int tx_mask; + unsigned int rx_mask; + struct snd_soc_card *card; struct list_head list; diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 6b59471cdf44..aac04ff84eea 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -431,7 +431,7 @@ int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm, const char *pin); int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); -void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); +void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card); /* Mostly internal - should not normally be used */ void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm); @@ -441,6 +441,8 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget_list **list); struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol); +struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( + struct snd_kcontrol *kcontrol); /* dapm widget types */ enum snd_soc_dapm_type { @@ -524,7 +526,6 @@ struct snd_soc_dapm_widget { const char *name; /* widget name */ const char *sname; /* stream name */ struct snd_soc_codec *codec; - struct snd_soc_platform *platform; struct list_head list; struct snd_soc_dapm_context *dapm; @@ -593,7 +594,6 @@ struct snd_soc_dapm_context { struct device *dev; /* from parent - for debug */ struct snd_soc_component *component; /* parent component */ struct snd_soc_codec *codec; /* parent codec */ - struct snd_soc_platform *platform; /* parent platform */ struct snd_soc_card *card; /* parent card */ /* used during DAPM updates */ @@ -601,6 +601,8 @@ struct snd_soc_dapm_context { struct list_head list; int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + int (*set_bias_level)(struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level); #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7e5fdc..be6ecae247b0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -248,6 +248,8 @@ .info = snd_soc_info_enum_double, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&xenum } +#define SOC_VALUE_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \ + SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) #define SND_SOC_BYTES(xname, xbase, xregs) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -270,7 +272,14 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&(struct soc_bytes_ext) \ {.max = xcount} } - +#define SND_SOC_BYTES_TLV(xname, xcount, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ + .tlv.c = (snd_soc_bytes_tlv_callback), \ + .info = snd_soc_info_bytes_ext, \ + .private_value = (unsigned long)&(struct soc_bytes_ext) \ + {.max = xcount, .get = xhandler_get, .put = xhandler_put, } } #define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \ xmin, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -436,6 +445,10 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, int snd_soc_platform_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_platform *platform); +int soc_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); + /* Jack reporting */ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, struct snd_soc_jack *jack); @@ -503,10 +516,12 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, const char *prefix); struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, const char *name); +int snd_soc_add_component_controls(struct snd_soc_component *component, + const struct snd_kcontrol_new *controls, unsigned int num_controls); int snd_soc_add_codec_controls(struct snd_soc_codec *codec, - const struct snd_kcontrol_new *controls, int num_controls); + const struct snd_kcontrol_new *controls, unsigned int num_controls); int snd_soc_add_platform_controls(struct snd_soc_platform *platform, - const struct snd_kcontrol_new *controls, int num_controls); + const struct snd_kcontrol_new *controls, unsigned int num_controls); int snd_soc_add_card_controls(struct snd_soc_card *soc_card, const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_add_dai_controls(struct snd_soc_dai *dai, @@ -552,6 +567,8 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *ucontrol); +int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv); int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, @@ -677,12 +694,17 @@ struct snd_soc_component_driver { int (*of_xlate_dai_name)(struct snd_soc_component *component, struct of_phandle_args *args, const char **dai_name); + void (*seq_notifier)(struct snd_soc_component *, enum snd_soc_dapm_type, + int subseq); + int (*stream_event)(struct snd_soc_component *, int event); }; struct snd_soc_component { const char *name; int id; + const char *name_prefix; struct device *dev; + struct snd_soc_card *card; unsigned int active; @@ -705,18 +727,18 @@ struct snd_soc_component { int val_bytes; struct mutex io_mutex; + + /* Don't use these, use snd_soc_component_get_dapm() */ + struct snd_soc_dapm_context dapm; + struct snd_soc_dapm_context *dapm_ptr; }; /* SoC Audio Codec device */ struct snd_soc_codec { - const char *name; - const char *name_prefix; - int id; struct device *dev; const struct snd_soc_codec_driver *driver; struct mutex mutex; - struct snd_soc_card *card; struct list_head list; struct list_head card_list; @@ -790,9 +812,6 @@ struct snd_soc_codec_driver { void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); - /* codec stream completion event */ - int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); - bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */ /* probe ordering - for components with runtime dependencies */ @@ -834,9 +853,6 @@ struct snd_soc_platform_driver { /* platform stream compress ops */ const struct snd_compr_ops *compr_ops; - /* platform stream completion event */ - int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); - /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; @@ -847,23 +863,23 @@ struct snd_soc_platform_driver { int (*bespoke_trigger)(struct snd_pcm_substream *, int); }; -struct snd_soc_platform { +struct snd_soc_dai_link_component { const char *name; - int id; + const struct device_node *of_node; + const char *dai_name; +}; + +struct snd_soc_platform { struct device *dev; const struct snd_soc_platform_driver *driver; unsigned int suspended:1; /* platform is suspended */ unsigned int probed:1; - struct snd_soc_card *card; struct list_head list; - struct list_head card_list; struct snd_soc_component component; - struct snd_soc_dapm_context dapm; - #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_platform_root; #endif @@ -896,6 +912,10 @@ struct snd_soc_dai_link { const struct device_node *codec_of_node; /* You MUST specify the DAI name within the codec */ const char *codec_dai_name; + + struct snd_soc_dai_link_component *codecs; + unsigned int num_codecs; + /* * You MAY specify the link's platform/PCM/DMA driver, either by * device name, or by DT/OF node, but not both. Some forms of link @@ -1047,7 +1067,6 @@ struct snd_soc_card { /* lists of probed devices belonging to this card */ struct list_head codec_dev_list; - struct list_head platform_dev_list; struct list_head widgets; struct list_head paths; @@ -1094,6 +1113,9 @@ struct snd_soc_pcm_runtime { struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai; + struct snd_soc_dai **codec_dais; + unsigned int num_codecs; + struct delayed_work delayed_work; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dpcm_root; @@ -1119,6 +1141,9 @@ struct soc_bytes { struct soc_bytes_ext { int max; + /* used for TLV byte control */ + int (*get)(unsigned int __user *bytes, unsigned int size); + int (*put)(const unsigned int __user *bytes, unsigned int size); }; /* multi register control */ @@ -1165,6 +1190,21 @@ static inline struct snd_soc_platform *snd_soc_component_to_platform( } /** + * snd_soc_dapm_to_component() - Casts a DAPM context to the component it is + * embedded in + * @dapm: The DAPM context to cast to the component + * + * This function must only be used on DAPM contexts that are known to be part of + * a component (e.g. in a component driver). Otherwise the behavior is + * undefined. + */ +static inline struct snd_soc_component *snd_soc_dapm_to_component( + struct snd_soc_dapm_context *dapm) +{ + return container_of(dapm, struct snd_soc_component, dapm); +} + +/** * snd_soc_dapm_to_codec() - Casts a DAPM context to the CODEC it is embedded in * @dapm: The DAPM context to cast to the CODEC * @@ -1188,7 +1228,18 @@ static inline struct snd_soc_codec *snd_soc_dapm_to_codec( static inline struct snd_soc_platform *snd_soc_dapm_to_platform( struct snd_soc_dapm_context *dapm) { - return container_of(dapm, struct snd_soc_platform, dapm); + return snd_soc_component_to_platform(snd_soc_dapm_to_component(dapm)); +} + +/** + * snd_soc_component_get_dapm() - Returns the DAPM context associated with a + * component + * @component: The component for which to get the DAPM context + */ +static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm( + struct snd_soc_component *component) +{ + return component->dapm_ptr; } /* codec IO */ @@ -1261,7 +1312,6 @@ static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd) static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) { INIT_LIST_HEAD(&card->codec_dev_list); - INIT_LIST_HEAD(&card->platform_dev_list); INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); diff --git a/include/sound/tas2552-plat.h b/include/sound/tas2552-plat.h new file mode 100644 index 000000000000..65e7627ba38e --- /dev/null +++ b/include/sound/tas2552-plat.h @@ -0,0 +1,25 @@ +/* + * TAS2552 driver platform header + * + * Copyright (C) 2014 Texas Instruments Inc. + * + * Author: Dan Murphy <dmurphy@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. + * + * 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. + */ + +#ifndef TAS2552_PLAT_H +#define TAS2552_PLAT_H + +struct tas2552_platform_data { + int enable_gpio; +}; + +#endif diff --git a/include/sound/wm8962.h b/include/sound/wm8962.h index 79e6d427b858..0af7c1674cbf 100644 --- a/include/sound/wm8962.h +++ b/include/sound/wm8962.h @@ -37,6 +37,7 @@ #define WM8962_GPIO_FN_MICSCD 22 struct wm8962_pdata { + struct clk *mclk; int gpio_base; u32 gpio_init[WM8962_MAX_GPIO]; diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index c75c795a377b..0194a641e4e2 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -296,17 +296,17 @@ TRACE_EVENT(snd_soc_cache_sync, TP_ARGS(codec, type, status), TP_STRUCT__entry( - __string( name, codec->name ) + __string( name, codec->component.name) __string( status, status ) __string( type, type ) __field( int, id ) ), TP_fast_assign( - __assign_str(name, codec->name); + __assign_str(name, codec->component.name); __assign_str(status, status); __assign_str(type, type); - __entry->id = codec->id; + __entry->id = codec->component.id; ), TP_printk("codec=%s.%d type=%s status=%s", __get_str(name), diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 224948342f14..32168f7ffce3 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -139,7 +139,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 11) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 12) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -391,7 +391,9 @@ struct snd_pcm_sw_params { snd_pcm_uframes_t silence_threshold; /* min distance from noise for silence filling */ snd_pcm_uframes_t silence_size; /* silence block size */ snd_pcm_uframes_t boundary; /* pointers wrap point */ - unsigned char reserved[64]; /* reserved for future */ + unsigned int proto; /* protocol version */ + unsigned int tstamp_type; /* timestamp type (req. proto >= 2.0.12) */ + unsigned char reserved[56]; /* reserved for future */ }; struct snd_pcm_channel_info { @@ -462,7 +464,8 @@ struct snd_xfern { enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ - SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, + SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ + SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, }; /* channel positions */ diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 467836057ee5..a80d5ea87ccd 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -47,15 +47,11 @@ static int alloc_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev, /* We use the PCI APIs for now until the generic one gets fixed * enough or until we get some macio-specific versions */ - r->space = dma_alloc_coherent( - &macio_get_pci_dev(i2sdev->macio)->dev, - r->size, - &r->bus_addr, - GFP_KERNEL); + r->space = dma_zalloc_coherent(&macio_get_pci_dev(i2sdev->macio)->dev, + r->size, &r->bus_addr, GFP_KERNEL); + if (!r->space) + return -ENOMEM; - if (!r->space) return -ENOMEM; - - memset(r->space, 0, r->size); r->cmds = (void*)DBDMA_ALIGN(r->space); r->bus_cmd_start = r->bus_addr + (dma_addr_t)((char*)r->cmds - (char*)r->space); diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 66de90ed30ca..39c3969ac1c7 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -152,9 +152,9 @@ static inline void pxa_ac97_cold_pxa27x(void) gsr_bits = 0; /* PXA27x Developers Manual section 13.5.2.2.1 */ - clk_enable(ac97conf_clk); + clk_prepare_enable(ac97conf_clk); udelay(5); - clk_disable(ac97conf_clk); + clk_disable_unprepare(ac97conf_clk); GCR = GCR_COLD_RST | GCR_WARM_RST; } #endif @@ -299,14 +299,14 @@ static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id) int pxa2xx_ac97_hw_suspend(void) { GCR |= GCR_ACLINK_OFF; - clk_disable(ac97_clk); + clk_disable_unprepare(ac97_clk); return 0; } EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_suspend); int pxa2xx_ac97_hw_resume(void) { - clk_enable(ac97_clk); + clk_prepare_enable(ac97_clk); return 0; } EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); @@ -368,7 +368,7 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev) goto err_clk; } - ret = clk_enable(ac97_clk); + ret = clk_prepare_enable(ac97_clk); if (ret) goto err_clk2; @@ -403,7 +403,7 @@ void pxa2xx_ac97_hw_remove(struct platform_device *dev) clk_put(ac97conf_clk); ac97conf_clk = NULL; } - clk_disable(ac97_clk); + clk_disable_unprepare(ac97_clk); clk_put(ac97_clk); ac97_clk = NULL; } diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 7403f348ed14..89028fab64fd 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -491,7 +491,7 @@ static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ if (params->buffer.fragment_size == 0 || - params->buffer.fragments > SIZE_MAX / params->buffer.fragment_size) + params->buffer.fragments > INT_MAX / params->buffer.fragment_size) return -EINVAL; /* now codec parameters */ diff --git a/sound/core/control.c b/sound/core/control.c index f0b0e14497a5..b9611344ff9e 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1406,11 +1406,11 @@ static long snd_ctl_ioctl(struct file *file, unsigned int cmd, unsigned long arg case SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS: return snd_ctl_subscribe_events(ctl, ip); case SNDRV_CTL_IOCTL_TLV_READ: - return snd_ctl_tlv_ioctl(ctl, argp, 0); + return snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_READ); case SNDRV_CTL_IOCTL_TLV_WRITE: - return snd_ctl_tlv_ioctl(ctl, argp, 1); + return snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_WRITE); case SNDRV_CTL_IOCTL_TLV_COMMAND: - return snd_ctl_tlv_ioctl(ctl, argp, -1); + return snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_CMD); case SNDRV_CTL_IOCTL_POWER: return -ENOPROTOOPT; case SNDRV_CTL_IOCTL_POWER_STATE: diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index af49721ba0e3..102e8fd1d450 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -101,7 +101,9 @@ struct snd_pcm_sw_params32 { u32 silence_threshold; u32 silence_size; u32 boundary; - unsigned char reserved[64]; + u32 proto; + u32 tstamp_type; + unsigned char reserved[56]; }; /* recalcuate the boundary within 32bit */ @@ -133,7 +135,9 @@ static int snd_pcm_ioctl_sw_params_compat(struct snd_pcm_substream *substream, get_user(params.start_threshold, &src->start_threshold) || get_user(params.stop_threshold, &src->stop_threshold) || get_user(params.silence_threshold, &src->silence_threshold) || - get_user(params.silence_size, &src->silence_size)) + get_user(params.silence_size, &src->silence_size) || + get_user(params.tstamp_type, &src->tstamp_type) || + get_user(params.proto, &src->proto)) return -EFAULT; /* * Check silent_size parameter. Since we have 64bit boundary, diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 76cbb9ec953a..6542c4083594 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -65,13 +65,15 @@ int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, enum dma_slave_buswidth buswidth; int bits; - bits = snd_pcm_format_physical_width(params_format(params)); + bits = params_physical_width(params); if (bits < 8 || bits > 64) return -EINVAL; else if (bits == 8) buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE; else if (bits == 16) buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + else if (bits == 24) + buswidth = DMA_SLAVE_BUSWIDTH_3_BYTES; else if (bits <= 32) buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; else diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index b653ab001fba..8cd2f930ad0b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -543,6 +543,9 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST) return -EINVAL; + if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12) && + params->tstamp_type > SNDRV_PCM_TSTAMP_TYPE_LAST) + return -EINVAL; if (params->avail_min == 0) return -EINVAL; if (params->silence_size >= runtime->boundary) { @@ -557,6 +560,8 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, err = 0; snd_pcm_stream_lock_irq(substream); runtime->tstamp_mode = params->tstamp_mode; + if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12)) + runtime->tstamp_type = params->tstamp_type; runtime->period_step = params->period_step; runtime->control->avail_min = params->avail_min; runtime->start_threshold = params->start_threshold; @@ -2540,9 +2545,7 @@ static int snd_pcm_tstamp(struct snd_pcm_substream *substream, int __user *_arg) return -EFAULT; if (arg < 0 || arg > SNDRV_PCM_TSTAMP_TYPE_LAST) return -EINVAL; - runtime->tstamp_type = SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY; - if (arg == SNDRV_PCM_TSTAMP_TYPE_MONOTONIC) - runtime->tstamp_type = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC; + runtime->tstamp_type = arg; return 0; } diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index 1e206de0c2dd..ba8e4a64e13e 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -101,9 +101,9 @@ int snd_seq_dump_var_event(const struct snd_seq_event *event, len -= size; } return 0; - } if (! (event->data.ext.len & SNDRV_SEQ_EXT_CHAINED)) { - return func(private_data, event->data.ext.ptr, len); } + if (!(event->data.ext.len & SNDRV_SEQ_EXT_CHAINED)) + return func(private_data, event->data.ext.ptr, len); cell = (struct snd_seq_event_cell *)event->data.ext.ptr; for (; len > 0 && cell; cell = cell->next) { diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 775ef2efc296..46dff64908c8 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -83,8 +83,8 @@ config SND_BEBOB * Edirol FA-66/FA-101 * PreSonus FIREBOX/FIREPOD/FP10/Inspire1394 * BridgeCo RDAudio1/Audio5 - * Mackie Onyx 1220/1620/1640 (Firewire I/O Card) - * Mackie d.2 (Firewire Option) + * Mackie Onyx 1220/1620/1640 (FireWire I/O Card) + * Mackie d.2 (FireWire Option) * Stanton FinalScratch 2 (ScratchAmp) * Tascam IF-FW/DM * Behringer XENIX UFX 1204/1604 @@ -92,7 +92,7 @@ config SND_BEBOB * Apogee Rosetta 200/400 (X-FireWire card) * Apogee DA/AD/DD-16X (X-FireWire card) * Apogee Ensemble - * ESI Quotafire610 + * ESI QuataFire 610 * AcousticReality eARMasterOne * CME MatrixKFW * Phonic Helix Board 12 MkII/18 MkII/24 MkII @@ -101,13 +101,13 @@ config SND_BEBOB * ICON FireXon * PrismSound Orpheus/ADA-8XR * TerraTec PHASE 24 FW/PHASE X24 FW/PHASE 88 Rack FW - * Terratec EWS MIC2/EWS MIC4 - * Terratec Aureon 7.1 Firewire + * TerraTec EWS MIC2/EWS MIC8 + * TerraTec Aureon 7.1 FireWire * Yamaha GO44/GO46 * Focusrite Saffire/Saffire LE/SaffirePro10 IO/SaffirePro26 IO - * M-Audio Firewire410/AudioPhile/Solo + * M-Audio FireWire410/AudioPhile/Solo * M-Audio Ozonic/NRV10/ProfireLightBridge - * M-Audio Firewire 1814/ProjectMix IO + * M-Audio FireWire 1814/ProjectMix IO To compile this driver as a module, choose M here: the module will be called snd-bebob. diff --git a/sound/firewire/fireworks/fireworks_proc.c b/sound/firewire/fireworks/fireworks_proc.c index f29d4aaf56a1..0639dcb13f7d 100644 --- a/sound/firewire/fireworks/fireworks_proc.c +++ b/sound/firewire/fireworks/fireworks_proc.c @@ -64,7 +64,7 @@ proc_read_hwinfo(struct snd_info_entry *entry, struct snd_info_buffer *buffer) hwinfo->phys_in_grp_count); for (i = 0; i < hwinfo->phys_in_grp_count; i++) { snd_iprintf(buffer, - "phys in grp[0x%d]: type 0x%d, count 0x%d\n", + "phys in grp[%d]: type 0x%X, count 0x%X\n", i, hwinfo->phys_out_grps[i].type, hwinfo->phys_out_grps[i].count); } @@ -73,7 +73,7 @@ proc_read_hwinfo(struct snd_info_entry *entry, struct snd_info_buffer *buffer) hwinfo->phys_out_grp_count); for (i = 0; i < hwinfo->phys_out_grp_count; i++) { snd_iprintf(buffer, - "phys out grps[0x%d]: type 0x%d, count 0x%d\n", + "phys out grps[%d]: type 0x%X, count 0x%X\n", i, hwinfo->phys_out_grps[i].type, hwinfo->phys_out_grps[i].count); } diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 3bbc3ec5be82..862735005b43 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -316,6 +316,7 @@ static int mpu_input_scanner(struct mpu_config *devc, unsigned char midic) case 0xf6: /* printk( "tune_request\n"); */ devc->m_state = ST_INIT; + break; /* * Real time messages @@ -972,7 +973,6 @@ int attach_mpu401(struct address_info *hw_config, struct module *owner) devc->m_busy = 0; devc->m_state = ST_INIT; devc->shared_irq = hw_config->always_detect; - devc->irq = hw_config->irq; spin_lock_init(&devc->lock); if (devc->irq < 0) diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index 4709e592e2cc..607cee4d545e 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -52,7 +52,7 @@ struct voice_info int panning; /* 0xffff means not set */ }; -typedef struct opl_devinfo +struct opl_devinfo { int base; int left_io, right_io; @@ -73,7 +73,7 @@ typedef struct opl_devinfo unsigned char cmask; int is_opl4; -} opl_devinfo; +}; static struct opl_devinfo *devc = NULL; diff --git a/sound/oss/pss.c b/sound/oss/pss.c index 145e36b2cfd0..ca0d6e9f49f5 100644 --- a/sound/oss/pss.c +++ b/sound/oss/pss.c @@ -123,25 +123,25 @@ static bool pss_mixer; #endif -typedef struct pss_mixerdata { +struct pss_mixerdata { unsigned int volume_l; unsigned int volume_r; unsigned int bass; unsigned int treble; unsigned int synth; -} pss_mixerdata; +}; -typedef struct pss_confdata { +struct pss_confdata { int base; int irq; int dma; int *osp; - pss_mixerdata mixer; + struct pss_mixerdata mixer; int ad_mixer_dev; -} pss_confdata; +}; -static pss_confdata pss_data; -static pss_confdata *devc = &pss_data; +static struct pss_confdata pss_data; +static struct pss_confdata *devc = &pss_data; static DEFINE_SPINLOCK(lock); static int pss_initialized; @@ -150,7 +150,7 @@ static int pss_cdrom_port = -1; /* Parameter for the PSS cdrom port */ static bool pss_enable_joystick; /* Parameter for enabling the joystick */ static coproc_operations pss_coproc_operations; -static void pss_write(pss_confdata *devc, int data) +static void pss_write(struct pss_confdata *devc, int data) { unsigned long i, limit; @@ -206,7 +206,7 @@ static int __init probe_pss(struct address_info *hw_config) return 1; } -static int set_irq(pss_confdata * devc, int dev, int irq) +static int set_irq(struct pss_confdata *devc, int dev, int irq) { static unsigned short irq_bits[16] = { @@ -232,7 +232,7 @@ static int set_irq(pss_confdata * devc, int dev, int irq) return 1; } -static void set_io_base(pss_confdata * devc, int dev, int base) +static void set_io_base(struct pss_confdata *devc, int dev, int base) { unsigned short tmp = inw(REG(dev)) & 0x003f; unsigned short bits = (base & 0x0ffc) << 4; @@ -240,7 +240,7 @@ static void set_io_base(pss_confdata * devc, int dev, int base) outw(bits | tmp, REG(dev)); } -static int set_dma(pss_confdata * devc, int dev, int dma) +static int set_dma(struct pss_confdata *devc, int dev, int dma) { static unsigned short dma_bits[8] = { @@ -264,7 +264,7 @@ static int set_dma(pss_confdata * devc, int dev, int dma) return 1; } -static int pss_reset_dsp(pss_confdata * devc) +static int pss_reset_dsp(struct pss_confdata *devc) { unsigned long i, limit = jiffies + HZ/10; @@ -275,7 +275,7 @@ static int pss_reset_dsp(pss_confdata * devc) return 1; } -static int pss_put_dspword(pss_confdata * devc, unsigned short word) +static int pss_put_dspword(struct pss_confdata *devc, unsigned short word) { int i, val; @@ -291,7 +291,7 @@ static int pss_put_dspword(pss_confdata * devc, unsigned short word) return 0; } -static int pss_get_dspword(pss_confdata * devc, unsigned short *word) +static int pss_get_dspword(struct pss_confdata *devc, unsigned short *word) { int i, val; @@ -307,7 +307,8 @@ static int pss_get_dspword(pss_confdata * devc, unsigned short *word) return 0; } -static int pss_download_boot(pss_confdata * devc, unsigned char *block, int size, int flags) +static int pss_download_boot(struct pss_confdata *devc, unsigned char *block, + int size, int flags) { int i, val, count; unsigned long limit; @@ -397,7 +398,7 @@ static int pss_download_boot(pss_confdata * devc, unsigned char *block, int size } /* Mixer */ -static void set_master_volume(pss_confdata *devc, int left, int right) +static void set_master_volume(struct pss_confdata *devc, int left, int right) { static unsigned char log_scale[101] = { 0xdb, 0xe0, 0xe3, 0xe5, 0xe7, 0xe9, 0xea, 0xeb, 0xec, 0xed, 0xed, 0xee, @@ -416,7 +417,7 @@ static void set_master_volume(pss_confdata *devc, int left, int right) pss_write(devc, log_scale[right] | 0x0100); } -static void set_synth_volume(pss_confdata *devc, int volume) +static void set_synth_volume(struct pss_confdata *devc, int volume) { int vol = ((0x8000*volume)/100L); pss_write(devc, 0x0080); @@ -425,21 +426,21 @@ static void set_synth_volume(pss_confdata *devc, int volume) pss_write(devc, vol); } -static void set_bass(pss_confdata *devc, int level) +static void set_bass(struct pss_confdata *devc, int level) { int vol = (int)(((0xfd - 0xf0) * level)/100L) + 0xf0; pss_write(devc, 0x0010); pss_write(devc, vol | 0x0200); }; -static void set_treble(pss_confdata *devc, int level) +static void set_treble(struct pss_confdata *devc, int level) { int vol = (((0xfd - 0xf0) * level)/100L) + 0xf0; pss_write(devc, 0x0010); pss_write(devc, vol | 0x0300); }; -static void pss_mixer_reset(pss_confdata *devc) +static void pss_mixer_reset(struct pss_confdata *devc) { set_master_volume(devc, 33, 33); set_bass(devc, 50); @@ -499,7 +500,8 @@ static int ret_vol_stereo(int left, int right) return ((right << 8) | left); } -static int call_ad_mixer(pss_confdata *devc,unsigned int cmd, void __user *arg) +static int call_ad_mixer(struct pss_confdata *devc, unsigned int cmd, + void __user *arg) { if (devc->ad_mixer_dev != NO_WSS_MIXER) return mixer_devs[devc->ad_mixer_dev]->ioctl(devc->ad_mixer_dev, cmd, arg); @@ -509,7 +511,7 @@ static int call_ad_mixer(pss_confdata *devc,unsigned int cmd, void __user *arg) static int pss_mixer_ioctl (int dev, unsigned int cmd, void __user *arg) { - pss_confdata *devc = mixer_devs[dev]->devc; + struct pss_confdata *devc = mixer_devs[dev]->devc; int cmdf = cmd & 0xff; if ((cmdf != SOUND_MIXER_VOLUME) && (cmdf != SOUND_MIXER_BASS) && diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 3a3a3a71088b..50dd0086cfb1 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -858,8 +858,8 @@ config SND_VIRTUOSO select SND_JACK if INPUT=y || INPUT=SND help Say Y here to include support for sound cards based on the - Asus AV66/AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, DS, - Essence ST (Deluxe), and Essence STX. + Asus AV66/AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, DS, DSX, + Essence ST (Deluxe), and Essence STX (II). Support for the HDAV1.3 (Deluxe) and HDAV1.3 Slim is experimental; for the Xense, missing. diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 9f10c9e0df5e..631aaa4046ad 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1754,9 +1754,6 @@ static struct snd_kcontrol_new snd_echo_vumeters_switch = { static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - struct echoaudio *chip; - - chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 96; uinfo->value.integer.min = ECHOGAIN_MINOUT; @@ -1798,9 +1795,6 @@ static struct snd_kcontrol_new snd_echo_vumeters = { static int snd_echo_channels_info_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - struct echoaudio *chip; - - chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 6; uinfo->value.integer.min = 0; diff --git a/sound/pci/hda/dell_wmi_helper.c b/sound/pci/hda/dell_wmi_helper.c new file mode 100644 index 000000000000..9c22f95838ef --- /dev/null +++ b/sound/pci/hda/dell_wmi_helper.c @@ -0,0 +1,76 @@ +/* Helper functions for Dell Mic Mute LED control; + * to be included from codec driver + */ + +#if IS_ENABLED(CONFIG_LEDS_DELL_NETBOOKS) +#include <linux/dell-led.h> + +static int dell_led_value; +static int (*dell_led_set_func)(int, int); +static void (*dell_old_cap_hook)(struct hda_codec *, + struct snd_kcontrol *, + struct snd_ctl_elem_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_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? */ + int val = (ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]) ? 0 : 1; + if (val == dell_led_value) + return; + dell_led_value = val; + if (dell_led_set_func) + dell_led_set_func(DELL_LED_MICMUTE, dell_led_value); + } +} + + +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) { + if (!dell_led_set_func) + dell_led_set_func = symbol_request(dell_app_wmi_led_set); + if (!dell_led_set_func) { + codec_warn(codec, "Failed to find dell wmi symbol dell_app_wmi_led_set\n"); + return; + } + + removefunc = true; + if (dell_led_set_func(DELL_LED_MICMUTE, false) >= 0) { + dell_led_value = 0; + if (spec->gen.num_adc_nids > 1) + 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; + } + } + + } + + if (dell_led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + symbol_put(dell_app_wmi_led_set); + dell_led_set_func = NULL; + dell_old_cap_hook = NULL; + } +} + +#else /* CONFIG_LEDS_DELL_NETBOOKS */ +static void alc_fixup_dell_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ +} + +#endif /* CONFIG_LEDS_DELL_NETBOOKS */ diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index dabe41975a9d..51dea49aadd4 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -17,8 +17,6 @@ #include "hda_local.h" #include "hda_auto_parser.h" -#define SFX "hda_codec: " - /* * Helper for automatic pin configuration */ @@ -856,7 +854,7 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, { const struct snd_hda_pin_quirk *pq; - if (codec->fixup_forced) + if (codec->fixup_id != HDA_FIXUP_ID_NOT_SET) return; for (pq = pin_quirk; pq->subvendor; pq++) { @@ -882,14 +880,17 @@ void snd_hda_pick_fixup(struct hda_codec *codec, const struct hda_fixup *fixlist) { const struct snd_pci_quirk *q; - int id = -1; + int id = HDA_FIXUP_ID_NOT_SET; const char *name = NULL; + if (codec->fixup_id != HDA_FIXUP_ID_NOT_SET) + return; + /* when model=nofixup is given, don't pick up any fixups */ if (codec->modelname && !strcmp(codec->modelname, "nofixup")) { codec->fixup_list = NULL; - codec->fixup_id = -1; - codec->fixup_forced = 1; + codec->fixup_name = NULL; + codec->fixup_id = HDA_FIXUP_ID_NO_FIXUP; return; } @@ -899,13 +900,12 @@ void snd_hda_pick_fixup(struct hda_codec *codec, codec->fixup_id = models->id; codec->fixup_name = models->name; codec->fixup_list = fixlist; - codec->fixup_forced = 1; return; } models++; } } - if (id < 0 && quirk) { + if (quirk) { q = snd_pci_quirk_lookup(codec->bus->pci, quirk); if (q) { id = q->value; @@ -929,7 +929,6 @@ void snd_hda_pick_fixup(struct hda_codec *codec, } } - codec->fixup_forced = 0; codec->fixup_id = id; if (id >= 0) { codec->fixup_list = fixlist; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4c20277a6835..ec6a7d0d1886 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1476,6 +1476,7 @@ int snd_hda_codec_new(struct hda_bus *bus, INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); codec->depop_delay = -1; + codec->fixup_id = HDA_FIXUP_ID_NOT_SET; #ifdef CONFIG_PM spin_lock_init(&codec->power_lock); @@ -2727,7 +2728,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) return 0; } -typedef int (*map_slave_func_t)(void *, struct snd_kcontrol *); +typedef int (*map_slave_func_t)(struct hda_codec *, void *, struct snd_kcontrol *); /* apply the function to all matching slave ctls in the mixer list */ static int map_slaves(struct hda_codec *codec, const char * const *slaves, @@ -2751,7 +2752,7 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves, name = tmpname; } if (!strcmp(sctl->id.name, name)) { - err = func(data, sctl); + err = func(codec, data, sctl); if (err) return err; break; @@ -2761,13 +2762,15 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves, return 0; } -static int check_slave_present(void *data, struct snd_kcontrol *sctl) +static int check_slave_present(struct hda_codec *codec, + void *data, struct snd_kcontrol *sctl) { return 1; } /* guess the value corresponding to 0dB */ -static int get_kctl_0dB_offset(struct snd_kcontrol *kctl, int *step_to_check) +static int get_kctl_0dB_offset(struct hda_codec *codec, + struct snd_kcontrol *kctl, int *step_to_check) { int _tlv[4]; const int *tlv = NULL; @@ -2788,7 +2791,7 @@ static int get_kctl_0dB_offset(struct snd_kcontrol *kctl, int *step_to_check) if (!step) return -1; if (*step_to_check && *step_to_check != step) { - snd_printk(KERN_ERR "hda_codec: Mismatching dB step for vmaster slave (%d!=%d)\n", + codec_err(codec, "Mismatching dB step for vmaster slave (%d!=%d)\n", - *step_to_check, step); return -1; } @@ -2813,20 +2816,28 @@ static int put_kctl_with_value(struct snd_kcontrol *kctl, int val) } /* initialize the slave volume with 0dB */ -static int init_slave_0dB(void *data, struct snd_kcontrol *slave) +static int init_slave_0dB(struct hda_codec *codec, + void *data, struct snd_kcontrol *slave) { - int offset = get_kctl_0dB_offset(slave, data); + int offset = get_kctl_0dB_offset(codec, slave, data); if (offset > 0) put_kctl_with_value(slave, offset); return 0; } /* unmute the slave */ -static int init_slave_unmute(void *data, struct snd_kcontrol *slave) +static int init_slave_unmute(struct hda_codec *codec, + void *data, struct snd_kcontrol *slave) { return put_kctl_with_value(slave, 1); } +static int add_slave(struct hda_codec *codec, + void *data, struct snd_kcontrol *slave) +{ + return snd_ctl_add_slave(data, slave); +} + /** * snd_hda_add_vmaster - create a virtual master control and add slaves * @codec: HD-audio codec @@ -2869,8 +2880,7 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, if (err < 0) return err; - err = map_slaves(codec, slaves, suffix, - (map_slave_func_t)snd_ctl_add_slave, kctl); + err = map_slaves(codec, slaves, suffix, add_slave, kctl); if (err < 0) return err; @@ -4280,6 +4290,7 @@ static struct hda_rate_tbl rate_bits[] = { /** * snd_hda_calc_stream_format - calculate format bitset + * @codec: HD-audio codec * @rate: the sample rate * @channels: the number of channels * @format: the PCM format (SNDRV_PCM_FORMAT_XXX) @@ -4289,7 +4300,8 @@ static struct hda_rate_tbl rate_bits[] = { * * Return zero if invalid. */ -unsigned int snd_hda_calc_stream_format(unsigned int rate, +unsigned int snd_hda_calc_stream_format(struct hda_codec *codec, + unsigned int rate, unsigned int channels, unsigned int format, unsigned int maxbps, @@ -4304,12 +4316,12 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, break; } if (!rate_bits[i].hz) { - snd_printdd("invalid rate %d\n", rate); + codec_dbg(codec, "invalid rate %d\n", rate); return 0; } if (channels == 0 || channels > 8) { - snd_printdd("invalid channels %d\n", channels); + codec_dbg(codec, "invalid channels %d\n", channels); return 0; } val |= channels - 1; @@ -4332,7 +4344,7 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, val |= AC_FMT_BITS_20; break; default: - snd_printdd("invalid format width %d\n", + codec_dbg(codec, "invalid format width %d\n", snd_pcm_format_width(format)); return 0; } @@ -5670,12 +5682,13 @@ EXPORT_SYMBOL_GPL(_snd_hda_set_pin_ctl); * suffix is appended to the label. This label index number is stored * to type_idx when non-NULL pointer is given. */ -int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label, +int snd_hda_add_imux_item(struct hda_codec *codec, + struct hda_input_mux *imux, const char *label, int index, int *type_idx) { int i, label_idx = 0; if (imux->num_items >= HDA_MAX_NUM_INPUTS) { - snd_printd(KERN_ERR "hda_codec: Too many imux items!\n"); + codec_err(codec, "hda_codec: Too many imux items!\n"); return -EINVAL; } for (i = 0; i < imux->num_items; i++) { diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5825aa17d8e3..bbc5a1392c75 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -402,7 +402,6 @@ struct hda_codec { /* fix-up list */ int fixup_id; - unsigned int fixup_forced:1; /* fixup explicitly set by user */ const struct hda_fixup *fixup_list; const char *fixup_name; @@ -538,7 +537,8 @@ void __snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid, int do_now); #define snd_hda_codec_cleanup_stream(codec, nid) \ __snd_hda_codec_cleanup_stream(codec, nid, 0) -unsigned int snd_hda_calc_stream_format(unsigned int rate, +unsigned int snd_hda_calc_stream_format(struct hda_codec *codec, + unsigned int rate, unsigned int channels, unsigned int format, unsigned int maxbps, diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 6df04d91c93c..8337645aa7a5 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -27,6 +27,7 @@ #include <linux/module.h> #include <linux/pm_runtime.h> #include <linux/slab.h> +#include <linux/reboot.h> #include <sound/core.h> #include <sound/initval.h> #include "hda_priv.h" @@ -152,11 +153,11 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) upper_32_bits(azx_dev->bdl.addr)); /* enable the position buffer */ - if (chip->position_fix[0] != POS_FIX_LPIB || - chip->position_fix[1] != POS_FIX_LPIB) { - if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE)) + if (chip->get_position[0] != azx_get_pos_lpib || + chip->get_position[1] != azx_get_pos_lpib) { + if (!(azx_readl(chip, DPLBASE) & AZX_DPLBASE_ENABLE)) azx_writel(chip, DPLBASE, - (u32)chip->posbuf.addr | ICH6_DPLBASE_ENABLE); + (u32)chip->posbuf.addr | AZX_DPLBASE_ENABLE); } /* set the interrupt enable bits in the descriptor control register */ @@ -482,7 +483,8 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) } azx_stream_reset(chip, azx_dev); - format_val = snd_hda_calc_stream_format(runtime->rate, + format_val = snd_hda_calc_stream_format(apcm->codec, + runtime->rate, runtime->channels, runtime->format, hinfo->maxbps, @@ -673,125 +675,40 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } -/* get the current DMA position with correction on VIA chips */ -static unsigned int azx_via_get_position(struct azx *chip, - struct azx_dev *azx_dev) +unsigned int azx_get_pos_lpib(struct azx *chip, struct azx_dev *azx_dev) { - unsigned int link_pos, mini_pos, bound_pos; - unsigned int mod_link_pos, mod_dma_pos, mod_mini_pos; - unsigned int fifo_size; - - link_pos = azx_sd_readl(chip, azx_dev, SD_LPIB); - if (azx_dev->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* Playback, no problem using link position */ - return link_pos; - } - - /* Capture */ - /* For new chipset, - * use mod to get the DMA position just like old chipset - */ - mod_dma_pos = le32_to_cpu(*azx_dev->posbuf); - mod_dma_pos %= azx_dev->period_bytes; - - /* azx_dev->fifo_size can't get FIFO size of in stream. - * Get from base address + offset. - */ - fifo_size = readw(chip->remap_addr + VIA_IN_STREAM0_FIFO_SIZE_OFFSET); - - if (azx_dev->insufficient) { - /* Link position never gather than FIFO size */ - if (link_pos <= fifo_size) - return 0; - - azx_dev->insufficient = 0; - } - - if (link_pos <= fifo_size) - mini_pos = azx_dev->bufsize + link_pos - fifo_size; - else - mini_pos = link_pos - fifo_size; - - /* Find nearest previous boudary */ - mod_mini_pos = mini_pos % azx_dev->period_bytes; - mod_link_pos = link_pos % azx_dev->period_bytes; - if (mod_link_pos >= fifo_size) - bound_pos = link_pos - mod_link_pos; - else if (mod_dma_pos >= mod_mini_pos) - bound_pos = mini_pos - mod_mini_pos; - else { - bound_pos = mini_pos - mod_mini_pos + azx_dev->period_bytes; - if (bound_pos >= azx_dev->bufsize) - bound_pos = 0; - } + return azx_sd_readl(chip, azx_dev, SD_LPIB); +} +EXPORT_SYMBOL_GPL(azx_get_pos_lpib); - /* Calculate real DMA position we want */ - return bound_pos + mod_dma_pos; +unsigned int azx_get_pos_posbuf(struct azx *chip, struct azx_dev *azx_dev) +{ + return le32_to_cpu(*azx_dev->posbuf); } +EXPORT_SYMBOL_GPL(azx_get_pos_posbuf); unsigned int azx_get_position(struct azx *chip, - struct azx_dev *azx_dev, - bool with_check) + struct azx_dev *azx_dev) { struct snd_pcm_substream *substream = azx_dev->substream; - struct azx_pcm *apcm = snd_pcm_substream_chip(substream); unsigned int pos; int stream = substream->stream; - struct hda_pcm_stream *hinfo = apcm->hinfo[stream]; int delay = 0; - switch (chip->position_fix[stream]) { - case POS_FIX_LPIB: - /* read LPIB */ - pos = azx_sd_readl(chip, azx_dev, SD_LPIB); - break; - case POS_FIX_VIACOMBO: - pos = azx_via_get_position(chip, azx_dev); - break; - default: - /* use the position buffer */ - pos = le32_to_cpu(*azx_dev->posbuf); - if (with_check && chip->position_fix[stream] == POS_FIX_AUTO) { - if (!pos || pos == (u32)-1) { - dev_info(chip->card->dev, - "Invalid position buffer, using LPIB read method instead.\n"); - chip->position_fix[stream] = POS_FIX_LPIB; - pos = azx_sd_readl(chip, azx_dev, SD_LPIB); - } else - chip->position_fix[stream] = POS_FIX_POSBUF; - } - break; - } + if (chip->get_position[stream]) + pos = chip->get_position[stream](chip, azx_dev); + else /* use the position buffer as default */ + pos = azx_get_pos_posbuf(chip, azx_dev); if (pos >= azx_dev->bufsize) pos = 0; - /* calculate runtime delay from LPIB */ - if (substream->runtime && - chip->position_fix[stream] == POS_FIX_POSBUF && - (chip->driver_caps & AZX_DCAPS_COUNT_LPIB_DELAY)) { - unsigned int lpib_pos = azx_sd_readl(chip, azx_dev, SD_LPIB); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - delay = pos - lpib_pos; - else - delay = lpib_pos - pos; - if (delay < 0) { - if (delay >= azx_dev->delay_negative_threshold) - delay = 0; - else - delay += azx_dev->bufsize; - } - if (delay >= azx_dev->period_bytes) { - dev_info(chip->card->dev, - "Unstable LPIB (%d >= %d); disabling LPIB delay counting\n", - delay, azx_dev->period_bytes); - delay = 0; - chip->driver_caps &= ~AZX_DCAPS_COUNT_LPIB_DELAY; - } - delay = bytes_to_frames(substream->runtime, delay); - } - if (substream->runtime) { + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + struct hda_pcm_stream *hinfo = apcm->hinfo[stream]; + + if (chip->get_delay[stream]) + delay += chip->get_delay[stream](chip, azx_dev, pos); if (hinfo->ops.get_delay) delay += hinfo->ops.get_delay(hinfo, apcm->codec, substream); @@ -809,7 +726,7 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) struct azx *chip = apcm->chip; struct azx_dev *azx_dev = get_azx_dev(substream); return bytes_to_frames(substream->runtime, - azx_get_position(chip, azx_dev, false)); + azx_get_position(chip, azx_dev)); } static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, @@ -1059,10 +976,10 @@ static void azx_init_cmd_io(struct azx *chip) azx_writew(chip, CORBWP, 0); /* reset the corb hw read pointer */ - azx_writew(chip, CORBRP, ICH6_CORBRP_RST); + azx_writew(chip, CORBRP, AZX_CORBRP_RST); if (!(chip->driver_caps & AZX_DCAPS_CORBRP_SELF_CLEAR)) { for (timeout = 1000; timeout > 0; timeout--) { - if ((azx_readw(chip, CORBRP) & ICH6_CORBRP_RST) == ICH6_CORBRP_RST) + if ((azx_readw(chip, CORBRP) & AZX_CORBRP_RST) == AZX_CORBRP_RST) break; udelay(1); } @@ -1082,7 +999,7 @@ static void azx_init_cmd_io(struct azx *chip) } /* enable corb dma */ - azx_writeb(chip, CORBCTL, ICH6_CORBCTL_RUN); + azx_writeb(chip, CORBCTL, AZX_CORBCTL_RUN); /* RIRB set up */ chip->rirb.addr = chip->rb.addr + 2048; @@ -1095,14 +1012,14 @@ static void azx_init_cmd_io(struct azx *chip) /* set the rirb size to 256 entries (ULI requires explicitly) */ azx_writeb(chip, RIRBSIZE, 0x02); /* reset the rirb hw write pointer */ - azx_writew(chip, RIRBWP, ICH6_RIRBWP_RST); + azx_writew(chip, RIRBWP, AZX_RIRBWP_RST); /* set N=1, get RIRB response interrupt for new entry */ if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) azx_writew(chip, RINTCNT, 0xc0); else azx_writew(chip, RINTCNT, 1); /* enable rirb dma and response irq */ - azx_writeb(chip, RIRBCTL, ICH6_RBCTL_DMA_EN | ICH6_RBCTL_IRQ_EN); + azx_writeb(chip, RIRBCTL, AZX_RBCTL_DMA_EN | AZX_RBCTL_IRQ_EN); spin_unlock_irq(&chip->reg_lock); } EXPORT_SYMBOL_GPL(azx_init_cmd_io); @@ -1146,7 +1063,7 @@ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val) return -EIO; } wp++; - wp %= ICH6_MAX_CORB_ENTRIES; + wp %= AZX_MAX_CORB_ENTRIES; rp = azx_readw(chip, CORBRP); if (wp == rp) { @@ -1164,7 +1081,7 @@ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val) return 0; } -#define ICH6_RIRB_EX_UNSOL_EV (1<<4) +#define AZX_RIRB_EX_UNSOL_EV (1<<4) /* retrieve RIRB entry - called from interrupt handler */ static void azx_update_rirb(struct azx *chip) @@ -1185,7 +1102,7 @@ static void azx_update_rirb(struct azx *chip) while (chip->rirb.rp != wp) { chip->rirb.rp++; - chip->rirb.rp %= ICH6_MAX_RIRB_ENTRIES; + chip->rirb.rp %= AZX_MAX_RIRB_ENTRIES; rp = chip->rirb.rp << 1; /* an RIRB entry is 8-bytes */ res_ex = le32_to_cpu(chip->rirb.buf[rp + 1]); @@ -1196,8 +1113,7 @@ static void azx_update_rirb(struct azx *chip) res, res_ex, chip->rirb.rp, wp); snd_BUG(); - } - else if (res_ex & ICH6_RIRB_EX_UNSOL_EV) + } else if (res_ex & AZX_RIRB_EX_UNSOL_EV) snd_hda_queue_unsol_event(chip->bus, res, res_ex); else if (chip->rirb.cmds[addr]) { chip->rirb.res[addr] = res; @@ -1305,7 +1221,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, /* release CORB/RIRB */ azx_free_cmd_io(chip); /* disable unsolicited responses */ - azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_UNSOL); + azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~AZX_GCTL_UNSOL); return -1; } @@ -1326,7 +1242,7 @@ static int azx_single_wait_for_response(struct azx *chip, unsigned int addr) while (timeout--) { /* check IRV busy bit */ - if (azx_readw(chip, IRS) & ICH6_IRS_VALID) { + if (azx_readw(chip, IRS) & AZX_IRS_VALID) { /* reuse rirb.res as the response return value */ chip->rirb.res[addr] = azx_readl(chip, IR); return 0; @@ -1350,13 +1266,13 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val) bus->rirb_error = 0; while (timeout--) { /* check ICB busy bit */ - if (!((azx_readw(chip, IRS) & ICH6_IRS_BUSY))) { + if (!((azx_readw(chip, IRS) & AZX_IRS_BUSY))) { /* Clear IRV valid bit */ azx_writew(chip, IRS, azx_readw(chip, IRS) | - ICH6_IRS_VALID); + AZX_IRS_VALID); azx_writel(chip, IC, val); azx_writew(chip, IRS, azx_readw(chip, IRS) | - ICH6_IRS_BUSY); + AZX_IRS_BUSY); return azx_single_wait_for_response(chip, addr); } udelay(1); @@ -1585,10 +1501,10 @@ void azx_enter_link_reset(struct azx *chip) unsigned long timeout; /* reset controller */ - azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET); + azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~AZX_GCTL_RESET); timeout = jiffies + msecs_to_jiffies(100); - while ((azx_readb(chip, GCTL) & ICH6_GCTL_RESET) && + while ((azx_readb(chip, GCTL) & AZX_GCTL_RESET) && time_before(jiffies, timeout)) usleep_range(500, 1000); } @@ -1599,7 +1515,7 @@ static void azx_exit_link_reset(struct azx *chip) { unsigned long timeout; - azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); + azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | AZX_GCTL_RESET); timeout = jiffies + msecs_to_jiffies(100); while (!azx_readb(chip, GCTL) && @@ -1640,7 +1556,7 @@ static int azx_reset(struct azx *chip, bool full_reset) /* Accept unsolicited responses */ if (!chip->single_cmd) azx_writel(chip, GCTL, azx_readl(chip, GCTL) | - ICH6_GCTL_UNSOL); + AZX_GCTL_UNSOL); /* detect codecs */ if (!chip->codec_mask) { @@ -1657,7 +1573,7 @@ static void azx_int_enable(struct azx *chip) { /* enable controller CIE and GIE */ azx_writel(chip, INTCTL, azx_readl(chip, INTCTL) | - ICH6_INT_CTRL_EN | ICH6_INT_GLOBAL_EN); + AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN); } /* disable interrupts */ @@ -1678,7 +1594,7 @@ static void azx_int_disable(struct azx *chip) /* disable controller CIE and GIE */ azx_writel(chip, INTCTL, azx_readl(chip, INTCTL) & - ~(ICH6_INT_CTRL_EN | ICH6_INT_GLOBAL_EN)); + ~(AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN)); } /* clear interrupts */ @@ -1699,7 +1615,7 @@ static void azx_int_clear(struct azx *chip) azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); /* clear int status */ - azx_writel(chip, INTSTS, ICH6_INT_CTRL_EN | ICH6_INT_ALL_STREAM); + azx_writel(chip, INTSTS, AZX_INT_CTRL_EN | AZX_INT_ALL_STREAM); } /* @@ -2031,5 +1947,30 @@ int azx_init_stream(struct azx *chip) } EXPORT_SYMBOL_GPL(azx_init_stream); +/* + * reboot notifier for hang-up problem at power-down + */ +static int azx_halt(struct notifier_block *nb, unsigned long event, void *buf) +{ + struct azx *chip = container_of(nb, struct azx, reboot_notifier); + snd_hda_bus_reboot_notify(chip->bus); + azx_stop_chip(chip); + return NOTIFY_OK; +} + +void azx_notifier_register(struct azx *chip) +{ + chip->reboot_notifier.notifier_call = azx_halt; + register_reboot_notifier(&chip->reboot_notifier); +} +EXPORT_SYMBOL_GPL(azx_notifier_register); + +void azx_notifier_unregister(struct azx *chip) +{ + if (chip->reboot_notifier.notifier_call) + unregister_reboot_notifier(&chip->reboot_notifier); +} +EXPORT_SYMBOL_GPL(azx_notifier_unregister); + MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Common HDA driver funcitons"); diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index baf0e77330af..c90d10fd4d8f 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -25,9 +25,9 @@ static inline struct azx_dev *get_azx_dev(struct snd_pcm_substream *substream) { return substream->runtime->private_data; } -unsigned int azx_get_position(struct azx *chip, - struct azx_dev *azx_dev, - bool with_check); +unsigned int azx_get_position(struct azx *chip, struct azx_dev *azx_dev); +unsigned int azx_get_pos_lpib(struct azx *chip, struct azx_dev *azx_dev); +unsigned int azx_get_pos_posbuf(struct azx *chip, struct azx_dev *azx_dev); /* Stream control. */ void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev); @@ -50,4 +50,7 @@ int azx_codec_configure(struct azx *chip); int azx_mixer_create(struct azx *chip); int azx_init_stream(struct azx *chip); +void azx_notifier_register(struct azx *chip); +void azx_notifier_unregister(struct azx *chip); + #endif /* __SOUND_HDA_CONTROLLER_H */ diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 46690a7f48f6..e1cd34d9011d 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -167,7 +167,8 @@ static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid, (buf[byte] >> (lowbit)) & ((1 << (bits)) - 1); \ }) -static void hdmi_update_short_audio_desc(struct cea_sad *a, +static void hdmi_update_short_audio_desc(struct hda_codec *codec, + struct cea_sad *a, const unsigned char *buf) { int i; @@ -188,8 +189,7 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, a->format = GRAB_BITS(buf, 0, 3, 4); switch (a->format) { case AUDIO_CODING_TYPE_REF_STREAM_HEADER: - snd_printd(KERN_INFO - "HDMI: audio coding type 0 not expected\n"); + codec_info(codec, "HDMI: audio coding type 0 not expected\n"); break; case AUDIO_CODING_TYPE_LPCM: @@ -233,9 +233,9 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, a->format = GRAB_BITS(buf, 2, 3, 5); if (a->format == AUDIO_CODING_XTYPE_HE_REF_CT || a->format >= AUDIO_CODING_XTYPE_FIRST_RESERVED) { - snd_printd(KERN_INFO - "HDMI: audio coding xtype %d not expected\n", - a->format); + codec_info(codec, + "HDMI: audio coding xtype %d not expected\n", + a->format); a->format = 0; } else a->format += AUDIO_CODING_TYPE_HE_AAC - @@ -247,7 +247,7 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, /* * Be careful, ELD buf could be totally rubbish! */ -int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e, +int snd_hdmi_parse_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e, const unsigned char *buf, int size) { int mnl; @@ -256,8 +256,7 @@ int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e, e->eld_ver = GRAB_BITS(buf, 0, 3, 5); if (e->eld_ver != ELD_VER_CEA_861D && e->eld_ver != ELD_VER_PARTIAL) { - snd_printd(KERN_INFO "HDMI: Unknown ELD version %d\n", - e->eld_ver); + codec_info(codec, "HDMI: Unknown ELD version %d\n", e->eld_ver); goto out_fail; } @@ -280,20 +279,20 @@ int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e, e->product_id = get_unaligned_le16(buf + 18); if (mnl > ELD_MAX_MNL) { - snd_printd(KERN_INFO "HDMI: MNL is reserved value %d\n", mnl); + codec_info(codec, "HDMI: MNL is reserved value %d\n", mnl); goto out_fail; } else if (ELD_FIXED_BYTES + mnl > size) { - snd_printd(KERN_INFO "HDMI: out of range MNL %d\n", mnl); + codec_info(codec, "HDMI: out of range MNL %d\n", mnl); goto out_fail; } else strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl + 1); for (i = 0; i < e->sad_count; i++) { if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) { - snd_printd(KERN_INFO "HDMI: out of range SAD %d\n", i); + codec_info(codec, "HDMI: out of range SAD %d\n", i); goto out_fail; } - hdmi_update_short_audio_desc(e->sad + i, + hdmi_update_short_audio_desc(codec, e->sad + i, buf + ELD_FIXED_BYTES + mnl + 3 * i); } @@ -394,7 +393,8 @@ static void hdmi_print_pcm_rates(int pcm, char *buf, int buflen) #define SND_PRINT_RATES_ADVISED_BUFSIZE 80 -static void hdmi_show_short_audio_desc(struct cea_sad *a) +static void hdmi_show_short_audio_desc(struct hda_codec *codec, + struct cea_sad *a) { char buf[SND_PRINT_RATES_ADVISED_BUFSIZE]; char buf2[8 + SND_PRINT_BITS_ADVISED_BUFSIZE] = ", bits ="; @@ -412,12 +412,10 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a) else buf2[0] = '\0'; - _snd_printd(SND_PR_VERBOSE, "HDMI: supports coding type %s:" - " channels = %d, rates =%s%s\n", - cea_audio_coding_type_names[a->format], - a->channels, - buf, - buf2); + codec_dbg(codec, + "HDMI: supports coding type %s: channels = %d, rates =%s%s\n", + cea_audio_coding_type_names[a->format], + a->channels, buf, buf2); } void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen) @@ -432,22 +430,22 @@ void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen) buf[j] = '\0'; /* necessary when j == 0 */ } -void snd_hdmi_show_eld(struct parsed_hdmi_eld *e) +void snd_hdmi_show_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e) { int i; - _snd_printd(SND_PR_VERBOSE, "HDMI: detected monitor %s at connection type %s\n", + codec_dbg(codec, "HDMI: detected monitor %s at connection type %s\n", e->monitor_name, eld_connection_type_names[e->conn_type]); if (e->spk_alloc) { char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf)); - _snd_printd(SND_PR_VERBOSE, "HDMI: available speakers:%s\n", buf); + codec_dbg(codec, "HDMI: available speakers:%s\n", buf); } for (i = 0; i < e->sad_count; i++) - hdmi_show_short_audio_desc(e->sad + i); + hdmi_show_short_audio_desc(codec, e->sad + i); } #ifdef CONFIG_PROC_FS diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 589e47c5aeb3..b956449ddada 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -350,16 +350,16 @@ static void print_nid_path(struct hda_codec *codec, const char *pfx, struct nid_path *path) { char buf[40]; + char *pos = buf; int i; + *pos = 0; + for (i = 0; i < path->depth; i++) + pos += scnprintf(pos, sizeof(buf) - (pos - buf), "%s%02x", + pos != buf ? ":" : "", + path->path[i]); - buf[0] = 0; - for (i = 0; i < path->depth; i++) { - char tmp[4]; - sprintf(tmp, ":%02x", path->path[i]); - strlcat(buf, tmp, sizeof(buf)); - } - codec_dbg(codec, "%s path: depth=%d %s\n", pfx, path->depth, buf); + codec_dbg(codec, "%s path: depth=%d '%s'\n", pfx, path->depth, buf); } /* called recursively */ @@ -1700,9 +1700,11 @@ static int fill_and_eval_dacs(struct hda_codec *codec, #define DEBUG_BADNESS #ifdef DEBUG_BADNESS -#define debug_badness(fmt, args...) codec_dbg(codec, fmt, ##args) +#define debug_badness(fmt, ...) \ + codec_dbg(codec, fmt, ##__VA_ARGS__) #else -#define debug_badness(...) +#define debug_badness(fmt, ...) \ + do { if (0) codec_dbg(codec, fmt, ##__VA_ARGS__); } while (0) #endif #ifdef DEBUG_BADNESS @@ -3054,7 +3056,7 @@ static int parse_capture_source(struct hda_codec *codec, hda_nid_t pin, if (spec->hp_mic_pin == pin) spec->hp_mic_mux_idx = imux->num_items; spec->imux_pins[imux->num_items] = pin; - snd_hda_add_imux_item(imux, label, cfg_idx, NULL); + snd_hda_add_imux_item(codec, imux, label, cfg_idx, NULL); imux_added = true; if (spec->dyn_adc_switch) spec->dyn_adc_idx[imux_idx] = c; diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c index 8b4940ba33d6..d4d0375ac181 100644 --- a/sound/pci/hda/hda_i915.c +++ b/sound/pci/hda/hda_i915.c @@ -28,8 +28,8 @@ * Clock) to 24MHz BCLK: BCLK = CDCLK * M / N * The values will be lost when the display power well is disabled. */ -#define ICH6_REG_EM4 0x100c -#define ICH6_REG_EM5 0x1010 +#define AZX_REG_EM4 0x100c +#define AZX_REG_EM5 0x1010 static int (*get_power)(void); static int (*put_power)(void); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 83cd19017cf3..5db1948699d8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -44,7 +44,6 @@ #include <linux/slab.h> #include <linux/pci.h> #include <linux/mutex.h> -#include <linux/reboot.h> #include <linux/io.h> #include <linux/pm_runtime.h> #include <linux/clocksource.h> @@ -66,6 +65,52 @@ #include "hda_priv.h" #include "hda_i915.h" +/* position fix mode */ +enum { + POS_FIX_AUTO, + POS_FIX_LPIB, + POS_FIX_POSBUF, + POS_FIX_VIACOMBO, + POS_FIX_COMBO, +}; + +/* Defines for ATI HD Audio support in SB450 south bridge */ +#define ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR 0x42 +#define ATI_SB450_HDAUDIO_ENABLE_SNOOP 0x02 + +/* Defines for Nvidia HDA support */ +#define NVIDIA_HDA_TRANSREG_ADDR 0x4e +#define NVIDIA_HDA_ENABLE_COHBITS 0x0f +#define NVIDIA_HDA_ISTRM_COH 0x4d +#define NVIDIA_HDA_OSTRM_COH 0x4c +#define NVIDIA_HDA_ENABLE_COHBIT 0x01 + +/* Defines for Intel SCH HDA snoop control */ +#define INTEL_SCH_HDA_DEVC 0x78 +#define INTEL_SCH_HDA_DEVC_NOSNOOP (0x1<<11) + +/* Define IN stream 0 FIFO size offset in VIA controller */ +#define VIA_IN_STREAM0_FIFO_SIZE_OFFSET 0x90 +/* Define VIA HD Audio Device ID*/ +#define VIA_HDAC_DEVICE_ID 0x3288 + +/* max number of SDs */ +/* ICH, ATI and VIA have 4 playback and 4 capture */ +#define ICH6_NUM_CAPTURE 4 +#define ICH6_NUM_PLAYBACK 4 + +/* ULI has 6 playback and 5 capture */ +#define ULI_NUM_CAPTURE 5 +#define ULI_NUM_PLAYBACK 6 + +/* ATI HDMI may have up to 8 playbacks and 0 capture */ +#define ATIHDMI_NUM_CAPTURE 0 +#define ATIHDMI_NUM_PLAYBACK 8 + +/* TERA has 4 playback and 3 capture */ +#define TERA_NUM_CAPTURE 3 +#define TERA_NUM_PLAYBACK 4 + static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; @@ -290,8 +335,28 @@ static char *driver_short_names[] = { struct hda_intel { struct azx chip; -}; + /* for pending irqs */ + struct work_struct irq_pending_work; + + /* sync probing */ + struct completion probe_wait; + struct work_struct probe_work; + + /* card list (for power_save trigger) */ + struct list_head list; + + /* extra flags */ + unsigned int irq_pending_warned:1; + + /* VGA-switcheroo setup */ + unsigned int use_vga_switcheroo:1; + unsigned int vga_switcheroo_registered:1; + unsigned int init_failed:1; /* delayed init failed */ + + /* secondary power domain for hdmi audio under vga device */ + struct dev_pm_domain hdmi_pm_domain; +}; #ifdef CONFIG_X86 static void __mark_pages_wc(struct azx *chip, struct snd_dma_buffer *dmab, bool on) @@ -373,7 +438,7 @@ static void azx_init_pci(struct azx *chip) */ if (!(chip->driver_caps & AZX_DCAPS_NO_TCSEL)) { dev_dbg(chip->card->dev, "Clearing TCSEL\n"); - update_pci_byte(chip->pci, ICH6_PCIREG_TCSEL, 0x07, 0); + update_pci_byte(chip->pci, AZX_PCIREG_TCSEL, 0x07, 0); } /* For ATI SB450/600/700/800/900 and AMD Hudson azalia HD audio, @@ -421,11 +486,44 @@ static void azx_init_pci(struct azx *chip) } } +/* calculate runtime delay from LPIB */ +static int azx_get_delay_from_lpib(struct azx *chip, struct azx_dev *azx_dev, + unsigned int pos) +{ + struct snd_pcm_substream *substream = azx_dev->substream; + int stream = substream->stream; + unsigned int lpib_pos = azx_get_pos_lpib(chip, azx_dev); + int delay; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + delay = pos - lpib_pos; + else + delay = lpib_pos - pos; + if (delay < 0) { + if (delay >= azx_dev->delay_negative_threshold) + delay = 0; + else + delay += azx_dev->bufsize; + } + + if (delay >= azx_dev->period_bytes) { + dev_info(chip->card->dev, + "Unstable LPIB (%d >= %d); disabling LPIB delay counting\n", + delay, azx_dev->period_bytes); + delay = 0; + chip->driver_caps &= ~AZX_DCAPS_COUNT_LPIB_DELAY; + chip->get_delay[stream] = NULL; + } + + return bytes_to_frames(substream->runtime, delay); +} + static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev); /* called from IRQ */ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) { + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); int ok; ok = azx_position_ok(chip, azx_dev); @@ -435,7 +533,7 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) } else if (ok == 0 && chip->bus && chip->bus->workq) { /* bogus IRQ, process it later */ azx_dev->irq_pending = 1; - queue_work(chip->bus->workq, &chip->irq_pending_work); + queue_work(chip->bus->workq, &hda->irq_pending_work); } return 0; } @@ -451,6 +549,8 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) */ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) { + struct snd_pcm_substream *substream = azx_dev->substream; + int stream = substream->stream; u32 wallclk; unsigned int pos; @@ -458,7 +558,25 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) if (wallclk < (azx_dev->period_wallclk * 2) / 3) return -1; /* bogus (too early) interrupt */ - pos = azx_get_position(chip, azx_dev, true); + if (chip->get_position[stream]) + pos = chip->get_position[stream](chip, azx_dev); + else { /* use the position buffer as default */ + pos = azx_get_pos_posbuf(chip, azx_dev); + if (!pos || pos == (u32)-1) { + dev_info(chip->card->dev, + "Invalid position buffer, using LPIB read method instead.\n"); + chip->get_position[stream] = azx_get_pos_lpib; + pos = azx_get_pos_lpib(chip, azx_dev); + chip->get_delay[stream] = NULL; + } else { + chip->get_position[stream] = azx_get_pos_posbuf; + if (chip->driver_caps & AZX_DCAPS_COUNT_LPIB_DELAY) + chip->get_delay[stream] = azx_get_delay_from_lpib; + } + } + + if (pos >= azx_dev->bufsize) + pos = 0; if (WARN_ONCE(!azx_dev->period_bytes, "hda-intel: zero azx_dev->period_bytes")) @@ -476,14 +594,15 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) */ static void azx_irq_pending_work(struct work_struct *work) { - struct azx *chip = container_of(work, struct azx, irq_pending_work); + struct hda_intel *hda = container_of(work, struct hda_intel, irq_pending_work); + struct azx *chip = &hda->chip; int i, pending, ok; - if (!chip->irq_pending_warned) { + if (!hda->irq_pending_warned) { dev_info(chip->card->dev, "IRQ timing workaround is activated for card #%d. Suggest a bigger bdl_pos_adj.\n", chip->card->number); - chip->irq_pending_warned = 1; + hda->irq_pending_warned = 1; } for (;;) { @@ -541,27 +660,86 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect) return 0; } +/* get the current DMA position with correction on VIA chips */ +static unsigned int azx_via_get_position(struct azx *chip, + struct azx_dev *azx_dev) +{ + unsigned int link_pos, mini_pos, bound_pos; + unsigned int mod_link_pos, mod_dma_pos, mod_mini_pos; + unsigned int fifo_size; + + link_pos = azx_sd_readl(chip, azx_dev, SD_LPIB); + if (azx_dev->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* Playback, no problem using link position */ + return link_pos; + } + + /* Capture */ + /* For new chipset, + * use mod to get the DMA position just like old chipset + */ + mod_dma_pos = le32_to_cpu(*azx_dev->posbuf); + mod_dma_pos %= azx_dev->period_bytes; + + /* azx_dev->fifo_size can't get FIFO size of in stream. + * Get from base address + offset. + */ + fifo_size = readw(chip->remap_addr + VIA_IN_STREAM0_FIFO_SIZE_OFFSET); + + if (azx_dev->insufficient) { + /* Link position never gather than FIFO size */ + if (link_pos <= fifo_size) + return 0; + + azx_dev->insufficient = 0; + } + + if (link_pos <= fifo_size) + mini_pos = azx_dev->bufsize + link_pos - fifo_size; + else + mini_pos = link_pos - fifo_size; + + /* Find nearest previous boudary */ + mod_mini_pos = mini_pos % azx_dev->period_bytes; + mod_link_pos = link_pos % azx_dev->period_bytes; + if (mod_link_pos >= fifo_size) + bound_pos = link_pos - mod_link_pos; + else if (mod_dma_pos >= mod_mini_pos) + bound_pos = mini_pos - mod_mini_pos; + else { + bound_pos = mini_pos - mod_mini_pos + azx_dev->period_bytes; + if (bound_pos >= azx_dev->bufsize) + bound_pos = 0; + } + + /* Calculate real DMA position we want */ + return bound_pos + mod_dma_pos; +} + #ifdef CONFIG_PM static DEFINE_MUTEX(card_list_lock); static LIST_HEAD(card_list); static void azx_add_card_list(struct azx *chip) { + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); mutex_lock(&card_list_lock); - list_add(&chip->list, &card_list); + list_add(&hda->list, &card_list); mutex_unlock(&card_list_lock); } static void azx_del_card_list(struct azx *chip) { + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); mutex_lock(&card_list_lock); - list_del_init(&chip->list); + list_del_init(&hda->list); mutex_unlock(&card_list_lock); } /* trigger power-save check at writing parameter */ static int param_set_xint(const char *val, const struct kernel_param *kp) { + struct hda_intel *hda; struct azx *chip; struct hda_codec *c; int prev = power_save; @@ -571,7 +749,8 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) return ret; mutex_lock(&card_list_lock); - list_for_each_entry(chip, &card_list, list) { + list_for_each_entry(hda, &card_list, list) { + chip = &hda->chip; if (!chip->bus || chip->disabled) continue; list_for_each_entry(c, &chip->bus->codec_list, list) @@ -593,10 +772,16 @@ static int azx_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; + struct azx *chip; + struct hda_intel *hda; struct azx_pcm *p; - if (chip->disabled || chip->init_failed) + if (!card) + return 0; + + chip = card->private_data; + hda = container_of(chip, struct hda_intel, chip); + if (chip->disabled || hda->init_failed) return 0; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -626,9 +811,15 @@ static int azx_resume(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; + struct azx *chip; + struct hda_intel *hda; - if (chip->disabled || chip->init_failed) + if (!card) + return 0; + + chip = card->private_data; + hda = container_of(chip, struct hda_intel, chip); + if (chip->disabled || hda->init_failed) return 0; if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { @@ -663,9 +854,15 @@ static int azx_resume(struct device *dev) static int azx_runtime_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; + struct azx *chip; + struct hda_intel *hda; + + if (!card) + return 0; - if (chip->disabled || chip->init_failed) + chip = card->private_data; + hda = container_of(chip, struct hda_intel, chip); + if (chip->disabled || hda->init_failed) return 0; if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) @@ -687,12 +884,18 @@ static int azx_runtime_suspend(struct device *dev) static int azx_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; + struct azx *chip; + struct hda_intel *hda; struct hda_bus *bus; struct hda_codec *codec; int status; - if (chip->disabled || chip->init_failed) + if (!card) + return 0; + + chip = card->private_data; + hda = container_of(chip, struct hda_intel, chip); + if (chip->disabled || hda->init_failed) return 0; if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) @@ -727,9 +930,15 @@ static int azx_runtime_resume(struct device *dev) static int azx_runtime_idle(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; + struct azx *chip; + struct hda_intel *hda; + + if (!card) + return 0; - if (chip->disabled || chip->init_failed) + chip = card->private_data; + hda = container_of(chip, struct hda_intel, chip); + if (chip->disabled || hda->init_failed) return 0; if (!power_save_controller || @@ -753,29 +962,6 @@ static const struct dev_pm_ops azx_pm = { #endif /* CONFIG_PM */ -/* - * reboot notifier for hang-up problem at power-down - */ -static int azx_halt(struct notifier_block *nb, unsigned long event, void *buf) -{ - struct azx *chip = container_of(nb, struct azx, reboot_notifier); - snd_hda_bus_reboot_notify(chip->bus); - azx_stop_chip(chip); - return NOTIFY_OK; -} - -static void azx_notifier_register(struct azx *chip) -{ - chip->reboot_notifier.notifier_call = azx_halt; - register_reboot_notifier(&chip->reboot_notifier); -} - -static void azx_notifier_unregister(struct azx *chip) -{ - if (chip->reboot_notifier.notifier_call) - unregister_reboot_notifier(&chip->reboot_notifier); -} - static int azx_probe_continue(struct azx *chip); #ifdef SUPPORT_VGA_SWITCHEROO @@ -786,10 +972,11 @@ static void azx_vs_set_state(struct pci_dev *pci, { struct snd_card *card = pci_get_drvdata(pci); struct azx *chip = card->private_data; + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); bool disabled; - wait_for_completion(&chip->probe_wait); - if (chip->init_failed) + wait_for_completion(&hda->probe_wait); + if (hda->init_failed) return; disabled = (state == VGA_SWITCHEROO_OFF); @@ -803,7 +990,7 @@ static void azx_vs_set_state(struct pci_dev *pci, "Start delayed initialization\n"); if (azx_probe_continue(chip) < 0) { dev_err(chip->card->dev, "initialization error\n"); - chip->init_failed = true; + hda->init_failed = true; } } } else { @@ -833,9 +1020,10 @@ static bool azx_vs_can_switch(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct azx *chip = card->private_data; + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); - wait_for_completion(&chip->probe_wait); - if (chip->init_failed) + wait_for_completion(&hda->probe_wait); + if (hda->init_failed) return false; if (chip->disabled || !chip->bus) return true; @@ -847,11 +1035,12 @@ static bool azx_vs_can_switch(struct pci_dev *pci) static void init_vga_switcheroo(struct azx *chip) { + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct pci_dev *p = get_bound_vga(chip->pci); if (p) { dev_info(chip->card->dev, "Handle VGA-switcheroo audio client\n"); - chip->use_vga_switcheroo = 1; + hda->use_vga_switcheroo = 1; pci_dev_put(p); } } @@ -863,9 +1052,10 @@ 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); int err; - if (!chip->use_vga_switcheroo) + if (!hda->use_vga_switcheroo) return 0; /* FIXME: currently only handling DIS controller * is there any machine with two switchable HDMI audio controllers? @@ -875,11 +1065,11 @@ static int register_vga_switcheroo(struct azx *chip) chip->bus != NULL); if (err < 0) return err; - chip->vga_switcheroo_registered = 1; + hda->vga_switcheroo_registered = 1; /* register as an optimus hdmi audio power domain */ vga_switcheroo_init_domain_pm_optimus_hdmi_audio(chip->card->dev, - &chip->hdmi_pm_domain); + &hda->hdmi_pm_domain); return 0; } #else @@ -895,7 +1085,6 @@ static int azx_free(struct azx *chip) { struct pci_dev *pci = chip->pci; struct hda_intel *hda = container_of(chip, struct hda_intel, chip); - int i; if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) @@ -906,13 +1095,13 @@ static int azx_free(struct azx *chip) azx_notifier_unregister(chip); - chip->init_failed = 1; /* to be sure */ - complete_all(&chip->probe_wait); + hda->init_failed = 1; /* to be sure */ + complete_all(&hda->probe_wait); - if (use_vga_switcheroo(chip)) { + if (use_vga_switcheroo(hda)) { if (chip->disabled && chip->bus) snd_hda_unlock_devices(chip->bus); - if (chip->vga_switcheroo_registered) + if (hda->vga_switcheroo_registered) vga_switcheroo_unregister_client(chip->pci); } @@ -1048,6 +1237,30 @@ static int check_position_fix(struct azx *chip, int fix) return POS_FIX_AUTO; } +static void assign_position_fix(struct azx *chip, int fix) +{ + static azx_get_pos_callback_t callbacks[] = { + [POS_FIX_AUTO] = NULL, + [POS_FIX_LPIB] = azx_get_pos_lpib, + [POS_FIX_POSBUF] = azx_get_pos_posbuf, + [POS_FIX_VIACOMBO] = azx_via_get_position, + [POS_FIX_COMBO] = azx_get_pos_lpib, + }; + + chip->get_position[0] = chip->get_position[1] = callbacks[fix]; + + /* combo mode uses LPIB only for playback */ + if (fix == POS_FIX_COMBO) + chip->get_position[1] = NULL; + + if (fix == POS_FIX_POSBUF && + (chip->driver_caps & AZX_DCAPS_COUNT_LPIB_DELAY)) { + chip->get_delay[0] = chip->get_delay[1] = + azx_get_delay_from_lpib; + } + +} + /* * black-lists for probe_mask */ @@ -1173,7 +1386,8 @@ static void azx_check_snoop_available(struct azx *chip) static void azx_probe_work(struct work_struct *work) { - azx_probe_continue(container_of(work, struct azx, probe_work)); + struct hda_intel *hda = container_of(work, struct hda_intel, probe_work); + azx_probe_continue(&hda->chip); } /* @@ -1216,19 +1430,13 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, check_msi(chip); chip->dev_index = dev; chip->jackpoll_ms = jackpoll_ms; - INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work); INIT_LIST_HEAD(&chip->pcm_list); - INIT_LIST_HEAD(&chip->list); + INIT_WORK(&hda->irq_pending_work, azx_irq_pending_work); + INIT_LIST_HEAD(&hda->list); init_vga_switcheroo(chip); - init_completion(&chip->probe_wait); - - chip->position_fix[0] = chip->position_fix[1] = - check_position_fix(chip, position_fix[dev]); - /* combo mode uses LPIB for playback */ - if (chip->position_fix[0] == POS_FIX_COMBO) { - chip->position_fix[0] = POS_FIX_LPIB; - chip->position_fix[1] = POS_FIX_AUTO; - } + init_completion(&hda->probe_wait); + + assign_position_fix(chip, check_position_fix(chip, position_fix[dev])); check_probe_mask(chip, dev); @@ -1257,7 +1465,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, } /* continue probing in work context as may trigger request module */ - INIT_WORK(&chip->probe_work, azx_probe_work); + INIT_WORK(&hda->probe_work, azx_probe_work); *rchip = chip; @@ -1315,7 +1523,7 @@ static int azx_first_init(struct azx *chip) NULL); if (p_smbus) { if (p_smbus->revision < 0x30) - gcap &= ~ICH6_GCAP_64OK; + gcap &= ~AZX_GCAP_64OK; pci_dev_put(p_smbus); } } @@ -1323,7 +1531,7 @@ static int azx_first_init(struct azx *chip) /* disable 64bit DMA address on some devices */ if (chip->driver_caps & AZX_DCAPS_NO_64BIT) { dev_dbg(card->dev, "Disabling 64bit DMA\n"); - gcap &= ~ICH6_GCAP_64OK; + gcap &= ~AZX_GCAP_64OK; } /* disable buffer size rounding to 128-byte multiples if supported */ @@ -1339,7 +1547,7 @@ static int azx_first_init(struct azx *chip) } /* allow 64bit DMA address if supported by H/W */ - if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) + if ((gcap & AZX_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64)); else { pci_set_dma_mask(pci, DMA_BIT_MASK(32)); @@ -1583,6 +1791,7 @@ static int azx_probe(struct pci_dev *pci, { static int dev; struct snd_card *card; + struct hda_intel *hda; struct azx *chip; bool schedule_probe; int err; @@ -1606,6 +1815,7 @@ static int azx_probe(struct pci_dev *pci, if (err < 0) goto out_free; card->private_data = chip; + hda = container_of(chip, struct hda_intel, chip); pci_set_drvdata(pci, card); @@ -1642,11 +1852,11 @@ static int azx_probe(struct pci_dev *pci, #endif if (schedule_probe) - schedule_work(&chip->probe_work); + schedule_work(&hda->probe_work); dev++; if (chip->disabled) - complete_all(&chip->probe_wait); + complete_all(&hda->probe_wait); return 0; out_free: @@ -1662,6 +1872,7 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] = { static int azx_probe_continue(struct azx *chip) { + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct pci_dev *pci = chip->pci; int dev = chip->dev_index; int err; @@ -1735,13 +1946,13 @@ static int azx_probe_continue(struct azx *chip) power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); - if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) || chip->use_vga_switcheroo) + if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) || hda->use_vga_switcheroo) pm_runtime_put_noidle(&pci->dev); out_free: if (err < 0) - chip->init_failed = 1; - complete_all(&chip->probe_wait); + hda->init_failed = 1; + complete_all(&hda->probe_wait); return err; } @@ -1806,6 +2017,9 @@ static const struct pci_device_id azx_ids[] = { /* BayTrail */ { PCI_DEVICE(0x8086, 0x0f04), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, + /* Braswell */ + { PCI_DEVICE(0x8086, 0x2284), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 4e2d4863daa1..364bb413e02a 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -268,7 +268,8 @@ int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux, struct snd_ctl_elem_value *ucontrol, hda_nid_t nid, unsigned int *cur_val); -int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label, +int snd_hda_add_imux_item(struct hda_codec *codec, + struct hda_input_mux *imux, const char *label, int index, int *type_index_ret); /* @@ -437,6 +438,8 @@ struct snd_hda_pin_quirk { #endif +#define HDA_FIXUP_ID_NOT_SET -1 +#define HDA_FIXUP_ID_NO_FIXUP -2 /* fixup types */ enum { @@ -773,9 +776,9 @@ struct hdmi_eld { int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid); int snd_hdmi_get_eld(struct hda_codec *codec, hda_nid_t nid, unsigned char *buf, int *eld_size); -int snd_hdmi_parse_eld(struct parsed_hdmi_eld *e, +int snd_hdmi_parse_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e, const unsigned char *buf, int size); -void snd_hdmi_show_eld(struct parsed_hdmi_eld *e); +void snd_hdmi_show_eld(struct hda_codec *codec, struct parsed_hdmi_eld *e); void snd_hdmi_eld_update_pcm_info(struct parsed_hdmi_eld *e, struct hda_pcm_stream *hinfo); diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index e9d1a5762a55..949cd437eeb2 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -22,107 +22,87 @@ /* * registers */ -#define ICH6_REG_GCAP 0x00 -#define ICH6_GCAP_64OK (1 << 0) /* 64bit address support */ -#define ICH6_GCAP_NSDO (3 << 1) /* # of serial data out signals */ -#define ICH6_GCAP_BSS (31 << 3) /* # of bidirectional streams */ -#define ICH6_GCAP_ISS (15 << 8) /* # of input streams */ -#define ICH6_GCAP_OSS (15 << 12) /* # of output streams */ -#define ICH6_REG_VMIN 0x02 -#define ICH6_REG_VMAJ 0x03 -#define ICH6_REG_OUTPAY 0x04 -#define ICH6_REG_INPAY 0x06 -#define ICH6_REG_GCTL 0x08 -#define ICH6_GCTL_RESET (1 << 0) /* controller reset */ -#define ICH6_GCTL_FCNTRL (1 << 1) /* flush control */ -#define ICH6_GCTL_UNSOL (1 << 8) /* accept unsol. response enable */ -#define ICH6_REG_WAKEEN 0x0c -#define ICH6_REG_STATESTS 0x0e -#define ICH6_REG_GSTS 0x10 -#define ICH6_GSTS_FSTS (1 << 1) /* flush status */ -#define ICH6_REG_INTCTL 0x20 -#define ICH6_REG_INTSTS 0x24 -#define ICH6_REG_WALLCLK 0x30 /* 24Mhz source */ -#define ICH6_REG_OLD_SSYNC 0x34 /* SSYNC for old ICH */ -#define ICH6_REG_SSYNC 0x38 -#define ICH6_REG_CORBLBASE 0x40 -#define ICH6_REG_CORBUBASE 0x44 -#define ICH6_REG_CORBWP 0x48 -#define ICH6_REG_CORBRP 0x4a -#define ICH6_CORBRP_RST (1 << 15) /* read pointer reset */ -#define ICH6_REG_CORBCTL 0x4c -#define ICH6_CORBCTL_RUN (1 << 1) /* enable DMA */ -#define ICH6_CORBCTL_CMEIE (1 << 0) /* enable memory error irq */ -#define ICH6_REG_CORBSTS 0x4d -#define ICH6_CORBSTS_CMEI (1 << 0) /* memory error indication */ -#define ICH6_REG_CORBSIZE 0x4e - -#define ICH6_REG_RIRBLBASE 0x50 -#define ICH6_REG_RIRBUBASE 0x54 -#define ICH6_REG_RIRBWP 0x58 -#define ICH6_RIRBWP_RST (1 << 15) /* write pointer reset */ -#define ICH6_REG_RINTCNT 0x5a -#define ICH6_REG_RIRBCTL 0x5c -#define ICH6_RBCTL_IRQ_EN (1 << 0) /* enable IRQ */ -#define ICH6_RBCTL_DMA_EN (1 << 1) /* enable DMA */ -#define ICH6_RBCTL_OVERRUN_EN (1 << 2) /* enable overrun irq */ -#define ICH6_REG_RIRBSTS 0x5d -#define ICH6_RBSTS_IRQ (1 << 0) /* response irq */ -#define ICH6_RBSTS_OVERRUN (1 << 2) /* overrun irq */ -#define ICH6_REG_RIRBSIZE 0x5e - -#define ICH6_REG_IC 0x60 -#define ICH6_REG_IR 0x64 -#define ICH6_REG_IRS 0x68 -#define ICH6_IRS_VALID (1<<1) -#define ICH6_IRS_BUSY (1<<0) - -#define ICH6_REG_DPLBASE 0x70 -#define ICH6_REG_DPUBASE 0x74 -#define ICH6_DPLBASE_ENABLE 0x1 /* Enable position buffer */ +#define AZX_REG_GCAP 0x00 +#define AZX_GCAP_64OK (1 << 0) /* 64bit address support */ +#define AZX_GCAP_NSDO (3 << 1) /* # of serial data out signals */ +#define AZX_GCAP_BSS (31 << 3) /* # of bidirectional streams */ +#define AZX_GCAP_ISS (15 << 8) /* # of input streams */ +#define AZX_GCAP_OSS (15 << 12) /* # of output streams */ +#define AZX_REG_VMIN 0x02 +#define AZX_REG_VMAJ 0x03 +#define AZX_REG_OUTPAY 0x04 +#define AZX_REG_INPAY 0x06 +#define AZX_REG_GCTL 0x08 +#define AZX_GCTL_RESET (1 << 0) /* controller reset */ +#define AZX_GCTL_FCNTRL (1 << 1) /* flush control */ +#define AZX_GCTL_UNSOL (1 << 8) /* accept unsol. response enable */ +#define AZX_REG_WAKEEN 0x0c +#define AZX_REG_STATESTS 0x0e +#define AZX_REG_GSTS 0x10 +#define AZX_GSTS_FSTS (1 << 1) /* flush status */ +#define AZX_REG_INTCTL 0x20 +#define AZX_REG_INTSTS 0x24 +#define AZX_REG_WALLCLK 0x30 /* 24Mhz source */ +#define AZX_REG_OLD_SSYNC 0x34 /* SSYNC for old ICH */ +#define AZX_REG_SSYNC 0x38 +#define AZX_REG_CORBLBASE 0x40 +#define AZX_REG_CORBUBASE 0x44 +#define AZX_REG_CORBWP 0x48 +#define AZX_REG_CORBRP 0x4a +#define AZX_CORBRP_RST (1 << 15) /* read pointer reset */ +#define AZX_REG_CORBCTL 0x4c +#define AZX_CORBCTL_RUN (1 << 1) /* enable DMA */ +#define AZX_CORBCTL_CMEIE (1 << 0) /* enable memory error irq */ +#define AZX_REG_CORBSTS 0x4d +#define AZX_CORBSTS_CMEI (1 << 0) /* memory error indication */ +#define AZX_REG_CORBSIZE 0x4e + +#define AZX_REG_RIRBLBASE 0x50 +#define AZX_REG_RIRBUBASE 0x54 +#define AZX_REG_RIRBWP 0x58 +#define AZX_RIRBWP_RST (1 << 15) /* write pointer reset */ +#define AZX_REG_RINTCNT 0x5a +#define AZX_REG_RIRBCTL 0x5c +#define AZX_RBCTL_IRQ_EN (1 << 0) /* enable IRQ */ +#define AZX_RBCTL_DMA_EN (1 << 1) /* enable DMA */ +#define AZX_RBCTL_OVERRUN_EN (1 << 2) /* enable overrun irq */ +#define AZX_REG_RIRBSTS 0x5d +#define AZX_RBSTS_IRQ (1 << 0) /* response irq */ +#define AZX_RBSTS_OVERRUN (1 << 2) /* overrun irq */ +#define AZX_REG_RIRBSIZE 0x5e + +#define AZX_REG_IC 0x60 +#define AZX_REG_IR 0x64 +#define AZX_REG_IRS 0x68 +#define AZX_IRS_VALID (1<<1) +#define AZX_IRS_BUSY (1<<0) + +#define AZX_REG_DPLBASE 0x70 +#define AZX_REG_DPUBASE 0x74 +#define AZX_DPLBASE_ENABLE 0x1 /* Enable position buffer */ /* SD offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; /* stream register offsets from stream base */ -#define ICH6_REG_SD_CTL 0x00 -#define ICH6_REG_SD_STS 0x03 -#define ICH6_REG_SD_LPIB 0x04 -#define ICH6_REG_SD_CBL 0x08 -#define ICH6_REG_SD_LVI 0x0c -#define ICH6_REG_SD_FIFOW 0x0e -#define ICH6_REG_SD_FIFOSIZE 0x10 -#define ICH6_REG_SD_FORMAT 0x12 -#define ICH6_REG_SD_BDLPL 0x18 -#define ICH6_REG_SD_BDLPU 0x1c +#define AZX_REG_SD_CTL 0x00 +#define AZX_REG_SD_STS 0x03 +#define AZX_REG_SD_LPIB 0x04 +#define AZX_REG_SD_CBL 0x08 +#define AZX_REG_SD_LVI 0x0c +#define AZX_REG_SD_FIFOW 0x0e +#define AZX_REG_SD_FIFOSIZE 0x10 +#define AZX_REG_SD_FORMAT 0x12 +#define AZX_REG_SD_BDLPL 0x18 +#define AZX_REG_SD_BDLPU 0x1c /* PCI space */ -#define ICH6_PCIREG_TCSEL 0x44 +#define AZX_PCIREG_TCSEL 0x44 /* * other constants */ -/* max number of SDs */ -/* ICH, ATI and VIA have 4 playback and 4 capture */ -#define ICH6_NUM_CAPTURE 4 -#define ICH6_NUM_PLAYBACK 4 - -/* ULI has 6 playback and 5 capture */ -#define ULI_NUM_CAPTURE 5 -#define ULI_NUM_PLAYBACK 6 - -/* ATI HDMI may have up to 8 playbacks and 0 capture */ -#define ATIHDMI_NUM_CAPTURE 0 -#define ATIHDMI_NUM_PLAYBACK 8 - -/* TERA has 4 playback and 3 capture */ -#define TERA_NUM_CAPTURE 3 -#define TERA_NUM_PLAYBACK 4 - -/* this number is statically defined for simplicity */ -#define MAX_AZX_DEV 16 - /* max number of fragments - we may use more if allocating more pages for BDL */ #define BDL_SIZE 4096 #define AZX_MAX_BDL_ENTRIES (BDL_SIZE / 16) @@ -160,13 +140,13 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define SD_STS_FIFO_READY 0x20 /* FIFO ready */ /* INTCTL and INTSTS */ -#define ICH6_INT_ALL_STREAM 0xff /* all stream interrupts */ -#define ICH6_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */ -#define ICH6_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */ +#define AZX_INT_ALL_STREAM 0xff /* all stream interrupts */ +#define AZX_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */ +#define AZX_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */ /* below are so far hardcoded - should read registers in future */ -#define ICH6_MAX_CORB_ENTRIES 256 -#define ICH6_MAX_RIRB_ENTRIES 256 +#define AZX_MAX_CORB_ENTRIES 256 +#define AZX_MAX_RIRB_ENTRIES 256 /* driver quirks (capabilities) */ /* bits 0-7 are used for indicating driver type */ @@ -192,35 +172,6 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ -/* position fix mode */ -enum { - POS_FIX_AUTO, - POS_FIX_LPIB, - POS_FIX_POSBUF, - POS_FIX_VIACOMBO, - POS_FIX_COMBO, -}; - -/* Defines for ATI HD Audio support in SB450 south bridge */ -#define ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR 0x42 -#define ATI_SB450_HDAUDIO_ENABLE_SNOOP 0x02 - -/* Defines for Nvidia HDA support */ -#define NVIDIA_HDA_TRANSREG_ADDR 0x4e -#define NVIDIA_HDA_ENABLE_COHBITS 0x0f -#define NVIDIA_HDA_ISTRM_COH 0x4d -#define NVIDIA_HDA_OSTRM_COH 0x4c -#define NVIDIA_HDA_ENABLE_COHBIT 0x01 - -/* Defines for Intel SCH HDA snoop control */ -#define INTEL_SCH_HDA_DEVC 0x78 -#define INTEL_SCH_HDA_DEVC_NOSNOOP (0x1<<11) - -/* Define IN stream 0 FIFO size offset in VIA controller */ -#define VIA_IN_STREAM0_FIFO_SIZE_OFFSET 0x90 -/* Define VIA HD Audio Device ID*/ -#define VIA_HDAC_DEVICE_ID 0x3288 - /* HD Audio class code */ #define PCI_CLASS_MULTIMEDIA_HD_AUDIO 0x0403 @@ -325,6 +276,9 @@ struct azx_pcm { struct list_head list; }; +typedef unsigned int (*azx_get_pos_callback_t)(struct azx *, struct azx_dev *); +typedef int (*azx_get_delay_callback_t)(struct azx *, struct azx_dev *, unsigned int pos); + struct azx { struct snd_card *card; struct pci_dev *pci; @@ -343,6 +297,10 @@ struct azx { /* Register interaction. */ const struct hda_controller_ops *ops; + /* position adjustment callbacks */ + azx_get_pos_callback_t get_position[2]; + azx_get_delay_callback_t get_delay[2]; + /* pci resources */ unsigned long addr; void __iomem *remap_addr; @@ -351,7 +309,6 @@ struct azx { /* locks */ spinlock_t reg_lock; struct mutex open_mutex; /* Prevents concurrent open/close operations */ - struct completion probe_wait; /* streams (x num_streams) */ struct azx_dev *azx_dev; @@ -378,7 +335,6 @@ struct azx { #endif /* flags */ - int position_fix[2]; /* for both playback/capture streams */ const int *bdl_pos_adj; int poll_count; unsigned int running:1; @@ -386,46 +342,23 @@ struct azx { unsigned int single_cmd:1; unsigned int polling_mode:1; unsigned int msi:1; - unsigned int irq_pending_warned:1; unsigned int probing:1; /* codec probing phase */ unsigned int snoop:1; unsigned int align_buffer_size:1; unsigned int region_requested:1; - - /* VGA-switcheroo setup */ - unsigned int use_vga_switcheroo:1; - unsigned int vga_switcheroo_registered:1; - unsigned int init_failed:1; /* delayed init failed */ unsigned int disabled:1; /* disabled by VGA-switcher */ /* for debugging */ unsigned int last_cmd[AZX_MAX_CODECS]; - /* for pending irqs */ - struct work_struct irq_pending_work; - - struct work_struct probe_work; - /* reboot notifier (for mysterious hangup problem at power-down) */ struct notifier_block reboot_notifier; - /* card list (for power_save trigger) */ - struct list_head list; - #ifdef CONFIG_SND_HDA_DSP_LOADER struct azx_dev saved_azx_dev; #endif - - /* secondary power domain for hdmi audio under vga device */ - struct dev_pm_domain hdmi_pm_domain; }; -#ifdef CONFIG_SND_VERBOSE_PRINTK -#define SFX /* nop */ -#else -#define SFX "hda-intel " -#endif - #ifdef CONFIG_X86 #define azx_snoop(chip) ((chip)->snoop) #else @@ -437,29 +370,29 @@ struct azx { */ #define azx_writel(chip, reg, value) \ - ((chip)->ops->reg_writel(value, (chip)->remap_addr + ICH6_REG_##reg)) + ((chip)->ops->reg_writel(value, (chip)->remap_addr + AZX_REG_##reg)) #define azx_readl(chip, reg) \ - ((chip)->ops->reg_readl((chip)->remap_addr + ICH6_REG_##reg)) + ((chip)->ops->reg_readl((chip)->remap_addr + AZX_REG_##reg)) #define azx_writew(chip, reg, value) \ - ((chip)->ops->reg_writew(value, (chip)->remap_addr + ICH6_REG_##reg)) + ((chip)->ops->reg_writew(value, (chip)->remap_addr + AZX_REG_##reg)) #define azx_readw(chip, reg) \ - ((chip)->ops->reg_readw((chip)->remap_addr + ICH6_REG_##reg)) + ((chip)->ops->reg_readw((chip)->remap_addr + AZX_REG_##reg)) #define azx_writeb(chip, reg, value) \ - ((chip)->ops->reg_writeb(value, (chip)->remap_addr + ICH6_REG_##reg)) + ((chip)->ops->reg_writeb(value, (chip)->remap_addr + AZX_REG_##reg)) #define azx_readb(chip, reg) \ - ((chip)->ops->reg_readb((chip)->remap_addr + ICH6_REG_##reg)) + ((chip)->ops->reg_readb((chip)->remap_addr + AZX_REG_##reg)) #define azx_sd_writel(chip, dev, reg, value) \ - ((chip)->ops->reg_writel(value, (dev)->sd_addr + ICH6_REG_##reg)) + ((chip)->ops->reg_writel(value, (dev)->sd_addr + AZX_REG_##reg)) #define azx_sd_readl(chip, dev, reg) \ - ((chip)->ops->reg_readl((dev)->sd_addr + ICH6_REG_##reg)) + ((chip)->ops->reg_readl((dev)->sd_addr + AZX_REG_##reg)) #define azx_sd_writew(chip, dev, reg, value) \ - ((chip)->ops->reg_writew(value, (dev)->sd_addr + ICH6_REG_##reg)) + ((chip)->ops->reg_writew(value, (dev)->sd_addr + AZX_REG_##reg)) #define azx_sd_readw(chip, dev, reg) \ - ((chip)->ops->reg_readw((dev)->sd_addr + ICH6_REG_##reg)) + ((chip)->ops->reg_readw((dev)->sd_addr + AZX_REG_##reg)) #define azx_sd_writeb(chip, dev, reg, value) \ - ((chip)->ops->reg_writeb(value, (dev)->sd_addr + ICH6_REG_##reg)) + ((chip)->ops->reg_writeb(value, (dev)->sd_addr + AZX_REG_##reg)) #define azx_sd_readb(chip, dev, reg) \ - ((chip)->ops->reg_readb((dev)->sd_addr + ICH6_REG_##reg)) + ((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg)) #endif /* __SOUND_HDA_PRIV_H */ diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 358414da6418..227990bc02e3 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -29,7 +29,6 @@ #include <linux/moduleparam.h> #include <linux/mutex.h> #include <linux/of_device.h> -#include <linux/reboot.h> #include <linux/slab.h> #include <linux/time.h> @@ -272,13 +271,9 @@ static int hda_tegra_resume(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); - int status; hda_tegra_enable_clocks(hda); - /* Read STATESTS before controller reset */ - status = azx_readw(chip, STATESTS); - hda_tegra_init(hda); azx_init_chip(chip, 1); @@ -295,30 +290,6 @@ static const struct dev_pm_ops hda_tegra_pm = { }; /* - * reboot notifier for hang-up problem at power-down - */ -static int hda_tegra_halt(struct notifier_block *nb, unsigned long event, - void *buf) -{ - struct azx *chip = container_of(nb, struct azx, reboot_notifier); - snd_hda_bus_reboot_notify(chip->bus); - azx_stop_chip(chip); - return NOTIFY_OK; -} - -static void hda_tegra_notifier_register(struct azx *chip) -{ - chip->reboot_notifier.notifier_call = hda_tegra_halt; - register_reboot_notifier(&chip->reboot_notifier); -} - -static void hda_tegra_notifier_unregister(struct azx *chip) -{ - if (chip->reboot_notifier.notifier_call) - unregister_reboot_notifier(&chip->reboot_notifier); -} - -/* * destructor */ static int hda_tegra_dev_free(struct snd_device *device) @@ -326,7 +297,7 @@ static int hda_tegra_dev_free(struct snd_device *device) int i; struct azx *chip = device->device_data; - hda_tegra_notifier_unregister(chip); + azx_notifier_unregister(chip); if (chip->initialized) { for (i = 0; i < chip->num_streams; i++) @@ -478,10 +449,7 @@ static int hda_tegra_create(struct snd_card *card, chip->driver_type = driver_caps & 0xff; chip->dev_index = 0; INIT_LIST_HEAD(&chip->pcm_list); - INIT_LIST_HEAD(&chip->list); - chip->position_fix[0] = POS_FIX_AUTO; - chip->position_fix[1] = POS_FIX_AUTO; chip->codec_probe_mask = -1; chip->single_cmd = false; @@ -559,7 +527,7 @@ static int hda_tegra_probe(struct platform_device *pdev) chip->running = 1; power_down_all_codecs(chip); - hda_tegra_notifier_register(chip); + azx_notifier_register(chip); return 0; diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 092f2bd030bd..4f3aba78f720 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2046,14 +2046,14 @@ enum dma_state { DMA_STATE_RUN = 1 }; -static int dma_convert_to_hda_format( +static int dma_convert_to_hda_format(struct hda_codec *codec, unsigned int sample_rate, unsigned short channels, unsigned short *hda_format) { unsigned int format_val; - format_val = snd_hda_calc_stream_format( + format_val = snd_hda_calc_stream_format(codec, sample_rate, channels, SNDRV_PCM_FORMAT_S32_LE, @@ -2452,7 +2452,7 @@ static int dspxfr_image(struct hda_codec *codec, } dma_engine->codec = codec; - dma_convert_to_hda_format(sample_rate, channels, &hda_format); + dma_convert_to_hda_format(codec, sample_rate, channels, &hda_format); dma_engine->m_converter_format = hda_format; dma_engine->buf_size = (ovly ? DSP_DMA_WRITE_BUFLEN_OVLY : DSP_DMA_WRITE_BUFLEN_INIT) * 2; diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 387f0b551889..3db724eaa53c 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -657,8 +657,10 @@ static void cs4208_fixup_mac(struct hda_codec *codec, { if (action != HDA_FIXUP_ACT_PRE_PROBE) return; + + codec->fixup_id = HDA_FIXUP_ID_NOT_SET; snd_hda_pick_fixup(codec, NULL, cs4208_mac_fixup_tbl, cs4208_fixups); - if (codec->fixup_id < 0 || codec->fixup_id == CS4208_MAC_AUTO) + if (codec->fixup_id == HDA_FIXUP_ID_NOT_SET) codec->fixup_id = CS4208_GPIO0; /* default fixup */ snd_hda_apply_fixup(codec, action); } diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 061ea5965dd5..ed3d133ffbb6 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -31,550 +31,11 @@ #include "hda_jack.h" #include "hda_generic.h" -#undef ENABLE_CMI_STATIC_QUIRKS - -#ifdef ENABLE_CMI_STATIC_QUIRKS -#define NUM_PINS 11 - - -/* board config type */ -enum { - CMI_MINIMAL, /* back 3-jack */ - CMI_MIN_FP, /* back 3-jack + front-panel 2-jack */ - CMI_FULL, /* back 6-jack + front-panel 2-jack */ - CMI_FULL_DIG, /* back 6-jack + front-panel 2-jack + digital I/O */ - CMI_ALLOUT, /* back 5-jack + front-panel 2-jack + digital out */ - CMI_AUTO, /* let driver guess it */ - CMI_MODELS -}; -#endif /* ENABLE_CMI_STATIC_QUIRKS */ - struct cmi_spec { struct hda_gen_spec gen; - -#ifdef ENABLE_CMI_STATIC_QUIRKS - /* below are only for static models */ - - int board_config; - unsigned int no_line_in: 1; /* no line-in (5-jack) */ - unsigned int front_panel: 1; /* has front-panel 2-jack */ - - /* playback */ - struct hda_multi_out multiout; - hda_nid_t dac_nids[AUTO_CFG_MAX_OUTS]; /* NID for each DAC */ - int num_dacs; - - /* capture */ - const hda_nid_t *adc_nids; - hda_nid_t dig_in_nid; - - /* capture source */ - const struct hda_input_mux *input_mux; - unsigned int cur_mux[2]; - - /* channel mode */ - int num_channel_modes; - const struct hda_channel_mode *channel_modes; - - struct hda_pcm pcm_rec[2]; /* PCM information */ - - /* pin default configuration */ - hda_nid_t pin_nid[NUM_PINS]; - unsigned int def_conf[NUM_PINS]; - unsigned int pin_def_confs; - - /* multichannel pins */ - struct hda_verb multi_init[9]; /* 2 verbs for each pin + terminator */ -#endif /* ENABLE_CMI_STATIC_QUIRKS */ -}; - -#ifdef ENABLE_CMI_STATIC_QUIRKS -/* - * input MUX - */ -static int cmi_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct cmi_spec *spec = codec->spec; - return snd_hda_input_mux_info(spec->input_mux, uinfo); -} - -static int cmi_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct cmi_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - - ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx]; - return 0; -} - -static int cmi_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct cmi_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - spec->adc_nids[adc_idx], &spec->cur_mux[adc_idx]); -} - -/* - * shared line-in, mic for surrounds - */ - -/* 3-stack / 2 channel */ -static const struct hda_verb cmi9880_ch2_init[] = { - /* set line-in PIN for input */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - /* set mic PIN for input, also enable vref */ - { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - /* route front PCM (DAC1) to HP */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, - {} -}; - -/* 3-stack / 6 channel */ -static const struct hda_verb cmi9880_ch6_init[] = { - /* set line-in PIN for output */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - /* set mic PIN for output */ - { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - /* route front PCM (DAC1) to HP */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, - {} -}; - -/* 3-stack+front / 8 channel */ -static const struct hda_verb cmi9880_ch8_init[] = { - /* set line-in PIN for output */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - /* set mic PIN for output */ - { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - /* route rear-surround PCM (DAC4) to HP */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x03 }, - {} -}; - -static const struct hda_channel_mode cmi9880_channel_modes[3] = { - { 2, cmi9880_ch2_init }, - { 6, cmi9880_ch6_init }, - { 8, cmi9880_ch8_init }, -}; - -static int cmi_ch_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct cmi_spec *spec = codec->spec; - return snd_hda_ch_mode_info(codec, uinfo, spec->channel_modes, - spec->num_channel_modes); -} - -static int cmi_ch_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct cmi_spec *spec = codec->spec; - return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_modes, - spec->num_channel_modes, spec->multiout.max_channels); -} - -static int cmi_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct cmi_spec *spec = codec->spec; - return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_modes, - spec->num_channel_modes, &spec->multiout.max_channels); -} - -/* - */ -static const struct snd_kcontrol_new cmi9880_basic_mixer[] = { - /* CMI9880 has no playback volumes! */ - HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), /* front */ - HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Side Playback Switch", 0x06, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = cmi_mux_enum_info, - .get = cmi_mux_enum_get, - .put = cmi_mux_enum_put, - }, - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x23, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x23, 0, HDA_OUTPUT), - { } /* end */ }; /* - * shared I/O pins - */ -static const struct snd_kcontrol_new cmi9880_ch_mode_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = cmi_ch_mode_info, - .get = cmi_ch_mode_get, - .put = cmi_ch_mode_put, - }, - { } /* end */ -}; - -/* AUD-in selections: - * 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x1f 0x20 - */ -static const struct hda_input_mux cmi9880_basic_mux = { - .num_items = 4, - .items = { - { "Front Mic", 0x5 }, - { "Rear Mic", 0x2 }, - { "Line", 0x1 }, - { "CD", 0x7 }, - } -}; - -static const struct hda_input_mux cmi9880_no_line_mux = { - .num_items = 3, - .items = { - { "Front Mic", 0x5 }, - { "Rear Mic", 0x2 }, - { "CD", 0x7 }, - } -}; - -/* front, rear, clfe, rear_surr */ -static const hda_nid_t cmi9880_dac_nids[4] = { - 0x03, 0x04, 0x05, 0x06 -}; -/* ADC0, ADC1 */ -static const hda_nid_t cmi9880_adc_nids[2] = { - 0x08, 0x09 -}; - -#define CMI_DIG_OUT_NID 0x07 -#define CMI_DIG_IN_NID 0x0a - -/* - */ -static const struct hda_verb cmi9880_basic_init[] = { - /* port-D for line out (rear panel) */ - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - /* port-E for HP out (front panel) */ - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - /* route front PCM to HP */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, - /* port-A for surround (rear panel) */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - /* port-G for CLFE (rear panel) */ - { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - { 0x1f, AC_VERB_SET_CONNECT_SEL, 0x02 }, - /* port-H for side (rear panel) */ - { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - { 0x20, AC_VERB_SET_CONNECT_SEL, 0x01 }, - /* port-C for line-in (rear panel) */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - /* port-B for mic-in (rear panel) with vref */ - { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - /* port-F for mic-in (front panel) with vref */ - { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - /* CD-in */ - { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - /* route front mic to ADC1/2 */ - { 0x08, AC_VERB_SET_CONNECT_SEL, 0x05 }, - { 0x09, AC_VERB_SET_CONNECT_SEL, 0x05 }, - {} /* terminator */ -}; - -static const struct hda_verb cmi9880_allout_init[] = { - /* port-D for line out (rear panel) */ - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - /* port-E for HP out (front panel) */ - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - /* route front PCM to HP */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, - /* port-A for side (rear panel) */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - /* port-G for CLFE (rear panel) */ - { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - { 0x1f, AC_VERB_SET_CONNECT_SEL, 0x02 }, - /* port-H for side (rear panel) */ - { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - { 0x20, AC_VERB_SET_CONNECT_SEL, 0x01 }, - /* port-C for surround (rear panel) */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - /* port-B for mic-in (rear panel) with vref */ - { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - /* port-F for mic-in (front panel) with vref */ - { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - /* CD-in */ - { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - /* route front mic to ADC1/2 */ - { 0x08, AC_VERB_SET_CONNECT_SEL, 0x05 }, - { 0x09, AC_VERB_SET_CONNECT_SEL, 0x05 }, - {} /* terminator */ -}; - -/* - */ -static int cmi9880_build_controls(struct hda_codec *codec) -{ - struct cmi_spec *spec = codec->spec; - struct snd_kcontrol *kctl; - int i, err; - - err = snd_hda_add_new_ctls(codec, cmi9880_basic_mixer); - if (err < 0) - return err; - if (spec->channel_modes) { - err = snd_hda_add_new_ctls(codec, cmi9880_ch_mode_mixer); - if (err < 0) - return err; - } - if (spec->multiout.dig_out_nid) { - err = snd_hda_create_spdif_out_ctls(codec, - spec->multiout.dig_out_nid, - spec->multiout.dig_out_nid); - if (err < 0) - return err; - err = snd_hda_create_spdif_share_sw(codec, - &spec->multiout); - if (err < 0) - return err; - spec->multiout.share_spdif = 1; - } - if (spec->dig_in_nid) { - err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); - if (err < 0) - return err; - } - - /* assign Capture Source enums to NID */ - kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); - for (i = 0; kctl && i < kctl->count; i++) { - err = snd_hda_add_nid(codec, kctl, i, spec->adc_nids[i]); - if (err < 0) - return err; - } - return 0; -} - -static int cmi9880_init(struct hda_codec *codec) -{ - struct cmi_spec *spec = codec->spec; - if (spec->board_config == CMI_ALLOUT) - snd_hda_sequence_write(codec, cmi9880_allout_init); - else - snd_hda_sequence_write(codec, cmi9880_basic_init); - if (spec->board_config == CMI_AUTO) - snd_hda_sequence_write(codec, spec->multi_init); - return 0; -} - -/* - * Analog playback callbacks - */ -static int cmi9880_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct cmi_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, - hinfo); -} - -static int cmi9880_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct cmi_spec *spec = codec->spec; - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, - format, substream); -} - -static int cmi9880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct cmi_spec *spec = codec->spec; - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - -/* - * Digital out - */ -static int cmi9880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct cmi_spec *spec = codec->spec; - return snd_hda_multi_out_dig_open(codec, &spec->multiout); -} - -static int cmi9880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct cmi_spec *spec = codec->spec; - return snd_hda_multi_out_dig_close(codec, &spec->multiout); -} - -static int cmi9880_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct cmi_spec *spec = codec->spec; - return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, - format, substream); -} - -/* - * Analog capture - */ -static int cmi9880_capture_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct cmi_spec *spec = codec->spec; - - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - stream_tag, 0, format); - return 0; -} - -static int cmi9880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct cmi_spec *spec = codec->spec; - - snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); - return 0; -} - - -/* - */ -static const struct hda_pcm_stream cmi9880_pcm_analog_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 8, - .nid = 0x03, /* NID to query formats and rates */ - .ops = { - .open = cmi9880_playback_pcm_open, - .prepare = cmi9880_playback_pcm_prepare, - .cleanup = cmi9880_playback_pcm_cleanup - }, -}; - -static const struct hda_pcm_stream cmi9880_pcm_analog_capture = { - .substreams = 2, - .channels_min = 2, - .channels_max = 2, - .nid = 0x08, /* NID to query formats and rates */ - .ops = { - .prepare = cmi9880_capture_pcm_prepare, - .cleanup = cmi9880_capture_pcm_cleanup - }, -}; - -static const struct hda_pcm_stream cmi9880_pcm_digital_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - /* NID is set in cmi9880_build_pcms */ - .ops = { - .open = cmi9880_dig_playback_pcm_open, - .close = cmi9880_dig_playback_pcm_close, - .prepare = cmi9880_dig_playback_pcm_prepare - }, -}; - -static const struct hda_pcm_stream cmi9880_pcm_digital_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - /* NID is set in cmi9880_build_pcms */ -}; - -static int cmi9880_build_pcms(struct hda_codec *codec) -{ - struct cmi_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; - - codec->num_pcms = 1; - codec->pcm_info = info; - - info->name = "CMI9880"; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cmi9880_pcm_analog_playback; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = cmi9880_pcm_analog_capture; - - if (spec->multiout.dig_out_nid || spec->dig_in_nid) { - codec->num_pcms++; - info++; - info->name = "CMI9880 Digital"; - info->pcm_type = HDA_PCM_TYPE_SPDIF; - if (spec->multiout.dig_out_nid) { - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cmi9880_pcm_digital_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; - } - if (spec->dig_in_nid) { - info->stream[SNDRV_PCM_STREAM_CAPTURE] = cmi9880_pcm_digital_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; - } - } - - return 0; -} - -static void cmi9880_free(struct hda_codec *codec) -{ - kfree(codec->spec); -} - -/* - */ - -static const char * const cmi9880_models[CMI_MODELS] = { - [CMI_MINIMAL] = "minimal", - [CMI_MIN_FP] = "min_fp", - [CMI_FULL] = "full", - [CMI_FULL_DIG] = "full_dig", - [CMI_ALLOUT] = "allout", - [CMI_AUTO] = "auto", -}; - -static const struct snd_pci_quirk cmi9880_cfg_tbl[] = { - SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", CMI_FULL_DIG), - SND_PCI_QUIRK(0x1854, 0x002b, "LG LS75", CMI_MINIMAL), - SND_PCI_QUIRK(0x1854, 0x0032, "LG", CMI_FULL_DIG), - {} /* terminator */ -}; - -static const struct hda_codec_ops cmi9880_patch_ops = { - .build_controls = cmi9880_build_controls, - .build_pcms = cmi9880_build_pcms, - .init = cmi9880_init, - .free = cmi9880_free, -}; -#endif /* ENABLE_CMI_STATIC_QUIRKS */ - -/* * stuff for auto-parser */ static const struct hda_codec_ops cmi_auto_patch_ops = { @@ -585,12 +46,18 @@ static const struct hda_codec_ops cmi_auto_patch_ops = { .unsol_event = snd_hda_jack_unsol_event, }; -static int cmi_parse_auto_config(struct hda_codec *codec) +static int patch_cmi9880(struct hda_codec *codec) { - struct cmi_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->gen.autocfg; + struct cmi_spec *spec; + struct auto_pin_cfg *cfg; int err; + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + cfg = &spec->gen.autocfg; snd_hda_gen_spec_init(&spec->gen); err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0); @@ -608,79 +75,6 @@ static int cmi_parse_auto_config(struct hda_codec *codec) return err; } - -static int patch_cmi9880(struct hda_codec *codec) -{ - struct cmi_spec *spec; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; -#ifdef ENABLE_CMI_STATIC_QUIRKS - spec->board_config = snd_hda_check_board_config(codec, CMI_MODELS, - cmi9880_models, - cmi9880_cfg_tbl); - if (spec->board_config < 0) { - codec_dbg(codec, "%s: BIOS auto-probing.\n", - codec->chip_name); - spec->board_config = CMI_AUTO; /* try everything */ - } - - if (spec->board_config == CMI_AUTO) - return cmi_parse_auto_config(codec); - - /* copy default DAC NIDs */ - memcpy(spec->dac_nids, cmi9880_dac_nids, sizeof(spec->dac_nids)); - spec->num_dacs = 4; - - switch (spec->board_config) { - case CMI_MINIMAL: - case CMI_MIN_FP: - spec->channel_modes = cmi9880_channel_modes; - if (spec->board_config == CMI_MINIMAL) - spec->num_channel_modes = 2; - else { - spec->front_panel = 1; - spec->num_channel_modes = 3; - } - spec->multiout.max_channels = cmi9880_channel_modes[0].channels; - spec->input_mux = &cmi9880_basic_mux; - break; - case CMI_FULL: - case CMI_FULL_DIG: - spec->front_panel = 1; - spec->multiout.max_channels = 8; - spec->input_mux = &cmi9880_basic_mux; - if (spec->board_config == CMI_FULL_DIG) { - spec->multiout.dig_out_nid = CMI_DIG_OUT_NID; - spec->dig_in_nid = CMI_DIG_IN_NID; - } - break; - case CMI_ALLOUT: - default: - spec->front_panel = 1; - spec->multiout.max_channels = 8; - spec->no_line_in = 1; - spec->input_mux = &cmi9880_no_line_mux; - spec->multiout.dig_out_nid = CMI_DIG_OUT_NID; - break; - } - - spec->multiout.num_dacs = spec->num_dacs; - spec->multiout.dac_nids = spec->dac_nids; - - spec->adc_nids = cmi9880_adc_nids; - - codec->patch_ops = cmi9880_patch_ops; - - return 0; -#else - return cmi_parse_auto_config(codec); -#endif -} - /* * patch entries */ diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1dc7e974f3b1..7627a69ca6d7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -34,27 +34,6 @@ #include "hda_jack.h" #include "hda_generic.h" -#undef ENABLE_CXT_STATIC_QUIRKS - -#define CXT_PIN_DIR_IN 0x00 -#define CXT_PIN_DIR_OUT 0x01 -#define CXT_PIN_DIR_INOUT 0x02 -#define CXT_PIN_DIR_IN_NOMICBIAS 0x03 -#define CXT_PIN_DIR_INOUT_NOMICBIAS 0x04 - -#define CONEXANT_HP_EVENT 0x37 -#define CONEXANT_MIC_EVENT 0x38 -#define CONEXANT_LINE_EVENT 0x39 - -/* Conexant 5051 specific */ - -#define CXT5051_SPDIF_OUT 0x12 -#define CXT5051_PORTB_EVENT 0x38 -#define CXT5051_PORTC_EVENT 0x39 - -#define AUTO_MIC_PORTB (1 << 1) -#define AUTO_MIC_PORTC (1 << 2) - struct conexant_spec { struct hda_gen_spec gen; @@ -72,64 +51,6 @@ struct conexant_spec { bool dc_enable; unsigned int dc_input_bias; /* offset into olpc_xo_dc_bias */ struct nid_path *dc_mode_path; - -#ifdef ENABLE_CXT_STATIC_QUIRKS - const struct snd_kcontrol_new *mixers[5]; - int num_mixers; - hda_nid_t vmaster_nid; - - const struct hda_verb *init_verbs[5]; /* initialization verbs - * don't forget NULL - * termination! - */ - unsigned int num_init_verbs; - - /* playback */ - struct hda_multi_out multiout; /* playback set-up - * max_channels, dacs must be set - * dig_out_nid and hp_nid are optional - */ - unsigned int cur_eapd; - unsigned int hp_present; - unsigned int line_present; - unsigned int auto_mic; - - /* capture */ - unsigned int num_adc_nids; - const hda_nid_t *adc_nids; - hda_nid_t dig_in_nid; /* digital-in NID; optional */ - - unsigned int cur_adc_idx; - hda_nid_t cur_adc; - unsigned int cur_adc_stream_tag; - unsigned int cur_adc_format; - - const struct hda_pcm_stream *capture_stream; - - /* capture source */ - const struct hda_input_mux *input_mux; - const hda_nid_t *capsrc_nids; - unsigned int cur_mux[3]; - - /* channel model */ - const struct hda_channel_mode *channel_mode; - int num_channel_mode; - - /* PCM information */ - struct hda_pcm pcm_rec[2]; /* used in build_pcms() */ - - unsigned int spdif_route; - - unsigned int port_d_mode; - unsigned int dell_automute:1; - unsigned int dell_vostro:1; - unsigned int ideapad:1; - unsigned int thinkpad:1; - unsigned int hp_laptop:1; - unsigned int asus:1; - - unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */ -#endif /* ENABLE_CXT_STATIC_QUIRKS */ }; @@ -173,2533 +94,6 @@ static int add_beep_ctls(struct hda_codec *codec) #define add_beep_ctls(codec) 0 #endif - -#ifdef ENABLE_CXT_STATIC_QUIRKS -static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, - hinfo); -} - -static int conexant_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, - stream_tag, - format, substream); -} - -static int conexant_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - -/* - * Digital out - */ -static int conexant_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_dig_open(codec, &spec->multiout); -} - -static int conexant_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_dig_close(codec, &spec->multiout); -} - -static int conexant_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, - stream_tag, - format, substream); -} - -/* - * Analog capture - */ -static int conexant_capture_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - stream_tag, 0, format); - return 0; -} - -static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); - return 0; -} - - - -static const struct hda_pcm_stream conexant_pcm_analog_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .open = conexant_playback_pcm_open, - .prepare = conexant_playback_pcm_prepare, - .cleanup = conexant_playback_pcm_cleanup - }, -}; - -static const struct hda_pcm_stream conexant_pcm_analog_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .prepare = conexant_capture_pcm_prepare, - .cleanup = conexant_capture_pcm_cleanup - }, -}; - - -static const struct hda_pcm_stream conexant_pcm_digital_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .open = conexant_dig_playback_pcm_open, - .close = conexant_dig_playback_pcm_close, - .prepare = conexant_dig_playback_pcm_prepare - }, -}; - -static const struct hda_pcm_stream conexant_pcm_digital_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - /* NID is set in alc_build_pcms */ -}; - -static int cx5051_capture_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - spec->cur_adc = spec->adc_nids[spec->cur_adc_idx]; - spec->cur_adc_stream_tag = stream_tag; - spec->cur_adc_format = format; - snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format); - return 0; -} - -static int cx5051_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct conexant_spec *spec = codec->spec; - snd_hda_codec_cleanup_stream(codec, spec->cur_adc); - spec->cur_adc = 0; - return 0; -} - -static const struct hda_pcm_stream cx5051_pcm_analog_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .prepare = cx5051_capture_pcm_prepare, - .cleanup = cx5051_capture_pcm_cleanup - }, -}; - -static int conexant_build_pcms(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; - - codec->num_pcms = 1; - codec->pcm_info = info; - - info->name = "CONEXANT Analog"; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = conexant_pcm_analog_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = - spec->multiout.max_channels; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = - spec->multiout.dac_nids[0]; - if (spec->capture_stream) - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *spec->capture_stream; - else { - if (codec->vendor_id == 0x14f15051) - info->stream[SNDRV_PCM_STREAM_CAPTURE] = - cx5051_pcm_analog_capture; - else { - info->stream[SNDRV_PCM_STREAM_CAPTURE] = - conexant_pcm_analog_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = - spec->num_adc_nids; - } - } - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; - - if (spec->multiout.dig_out_nid) { - info++; - codec->num_pcms++; - info->name = "Conexant Digital"; - info->pcm_type = HDA_PCM_TYPE_SPDIF; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = - conexant_pcm_digital_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = - spec->multiout.dig_out_nid; - if (spec->dig_in_nid) { - info->stream[SNDRV_PCM_STREAM_CAPTURE] = - conexant_pcm_digital_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = - spec->dig_in_nid; - } - } - - return 0; -} - -static int conexant_mux_enum_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - - return snd_hda_input_mux_info(spec->input_mux, uinfo); -} - -static int conexant_mux_enum_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - - ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx]; - return 0; -} - -static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - spec->capsrc_nids[adc_idx], - &spec->cur_mux[adc_idx]); -} - -static void conexant_set_power(struct hda_codec *codec, hda_nid_t fg, - unsigned int power_state) -{ - if (power_state == AC_PWRST_D3) - msleep(100); - snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, - power_state); - /* partial workaround for "azx_get_response timeout" */ - if (power_state == AC_PWRST_D0) - msleep(10); - snd_hda_codec_set_power_to_all(codec, fg, power_state); -} - -static int conexant_init(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_init_verbs; i++) - snd_hda_sequence_write(codec, spec->init_verbs[i]); - return 0; -} - -static void conexant_free(struct hda_codec *codec) -{ - kfree(codec->spec); -} - -static const struct snd_kcontrol_new cxt_capture_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, - {} -}; - -static const char * const slave_pfxs[] = { - "Headphone", "Speaker", "Bass Speaker", "Front", "Surround", "CLFE", - NULL -}; - -static int conexant_build_controls(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int i; - int err; - - for (i = 0; i < spec->num_mixers; i++) { - err = snd_hda_add_new_ctls(codec, spec->mixers[i]); - if (err < 0) - return err; - } - if (spec->multiout.dig_out_nid) { - err = snd_hda_create_spdif_out_ctls(codec, - spec->multiout.dig_out_nid, - spec->multiout.dig_out_nid); - if (err < 0) - return err; - err = snd_hda_create_spdif_share_sw(codec, - &spec->multiout); - if (err < 0) - return err; - spec->multiout.share_spdif = 1; - } - if (spec->dig_in_nid) { - err = snd_hda_create_spdif_in_ctls(codec,spec->dig_in_nid); - if (err < 0) - return err; - } - - /* if we have no master control, let's create it */ - if (spec->vmaster_nid && - !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { - unsigned int vmaster_tlv[4]; - snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, - HDA_OUTPUT, vmaster_tlv); - err = snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, slave_pfxs, - "Playback Volume"); - if (err < 0) - return err; - } - if (spec->vmaster_nid && - !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { - err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, slave_pfxs, - "Playback Switch"); - if (err < 0) - return err; - } - - if (spec->input_mux) { - err = snd_hda_add_new_ctls(codec, cxt_capture_mixers); - if (err < 0) - return err; - } - - err = add_beep_ctls(codec); - if (err < 0) - return err; - - return 0; -} - -static const struct hda_codec_ops conexant_patch_ops = { - .build_controls = conexant_build_controls, - .build_pcms = conexant_build_pcms, - .init = conexant_init, - .free = conexant_free, - .set_power_state = conexant_set_power, -}; - -static int patch_conexant_auto(struct hda_codec *codec); -/* - * EAPD control - * the private value = nid | (invert << 8) - */ - -#define cxt_eapd_info snd_ctl_boolean_mono_info - -static int cxt_eapd_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - int invert = (kcontrol->private_value >> 8) & 1; - if (invert) - ucontrol->value.integer.value[0] = !spec->cur_eapd; - else - ucontrol->value.integer.value[0] = spec->cur_eapd; - return 0; - -} - -static int cxt_eapd_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - int invert = (kcontrol->private_value >> 8) & 1; - hda_nid_t nid = kcontrol->private_value & 0xff; - unsigned int eapd; - - eapd = !!ucontrol->value.integer.value[0]; - if (invert) - eapd = !eapd; - if (eapd == spec->cur_eapd) - return 0; - - spec->cur_eapd = eapd; - snd_hda_codec_write_cache(codec, nid, - 0, AC_VERB_SET_EAPD_BTLENABLE, - eapd ? 0x02 : 0x00); - return 1; -} - -/* controls for test mode */ -#ifdef CONFIG_SND_DEBUG - -#define CXT_EAPD_SWITCH(xname, nid, mask) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .info = cxt_eapd_info, \ - .get = cxt_eapd_get, \ - .put = cxt_eapd_put, \ - .private_value = nid | (mask<<16) } - - - -static int conexant_ch_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode, - spec->num_channel_mode); -} - -static int conexant_ch_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - spec->multiout.max_channels); -} - -static int conexant_ch_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - &spec->multiout.max_channels); - return err; -} - -#define CXT_PIN_MODE(xname, nid, dir) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .info = conexant_ch_mode_info, \ - .get = conexant_ch_mode_get, \ - .put = conexant_ch_mode_put, \ - .private_value = nid | (dir<<16) } - -#endif /* CONFIG_SND_DEBUG */ - -/* Conexant 5045 specific */ - -static const hda_nid_t cxt5045_dac_nids[1] = { 0x19 }; -static const hda_nid_t cxt5045_adc_nids[1] = { 0x1a }; -static const hda_nid_t cxt5045_capsrc_nids[1] = { 0x1a }; -#define CXT5045_SPDIF_OUT 0x18 - -static const struct hda_channel_mode cxt5045_modes[1] = { - { 2, NULL }, -}; - -static const struct hda_input_mux cxt5045_capture_source = { - .num_items = 2, - .items = { - { "Internal Mic", 0x1 }, - { "Mic", 0x2 }, - } -}; - -static const struct hda_input_mux cxt5045_capture_source_benq = { - .num_items = 4, - .items = { - { "Internal Mic", 0x1 }, - { "Mic", 0x2 }, - { "Line", 0x3 }, - { "Mixer", 0x0 }, - } -}; - -/* turn on/off EAPD (+ mute HP) as a master switch */ -static int cxt5045_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - if (!cxt_eapd_put(kcontrol, ucontrol)) - return 0; - - /* toggle internal speakers mute depending of presence of - * the headphone jack - */ - bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - - bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, 0x11, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - return 1; -} - -/* bind volumes of both NID 0x10 and 0x11 */ -static const struct hda_bind_ctls cxt5045_hp_bind_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x10, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -/* toggle input of built-in and mic jack appropriately */ -static void cxt5045_hp_automic(struct hda_codec *codec) -{ - static const struct hda_verb mic_jack_on[] = { - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {} - }; - static const struct hda_verb mic_jack_off[] = { - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {} - }; - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x12); - if (present) - snd_hda_sequence_write(codec, mic_jack_on); - else - snd_hda_sequence_write(codec, mic_jack_off); -} - - -/* mute internal speaker if HP is plugged */ -static void cxt5045_hp_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - spec->hp_present = snd_hda_jack_detect(codec, 0x11); - - bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} - -/* unsolicited event for HP jack sensing */ -static void cxt5045_hp_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - res >>= 26; - switch (res) { - case CONEXANT_HP_EVENT: - cxt5045_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: - cxt5045_hp_automic(codec); - break; - - } -} - -static const struct snd_kcontrol_new cxt5045_mixers[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x17, 0x2, HDA_INPUT), - HDA_BIND_VOL("Master Playback Volume", &cxt5045_hp_bind_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5045_hp_master_sw_put, - .private_value = 0x10, - }, - - {} -}; - -static const struct snd_kcontrol_new cxt5045_benq_mixers[] = { - HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT), - - {} -}; - -static const struct hda_verb cxt5045_init_verbs[] = { - /* Line in, Mic */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, - /* HP, Amp */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x10, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x11, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Record selector: Internal mic */ - {0x1a, AC_VERB_SET_CONNECT_SEL,0x1}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, - /* SPDIF route: PCM */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - { 0x13, AC_VERB_SET_CONNECT_SEL, 0x0 }, - /* EAPD */ - {0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2 }, /* default on */ - { } /* end */ -}; - -static const struct hda_verb cxt5045_benq_init_verbs[] = { - /* Internal Mic, Mic */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, - /* Line In,HP, Amp */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x10, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x11, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Record selector: Internal mic */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, - /* SPDIF route: PCM */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* EAPD */ - {0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - { } /* end */ -}; - -static const struct hda_verb cxt5045_hp_sense_init_verbs[] = { - /* pin sensing on HP jack */ - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5045_mic_sense_init_verbs[] = { - /* pin sensing on HP jack */ - {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -#ifdef CONFIG_SND_DEBUG -/* Test configuration for debugging, modelled after the ALC260 test - * configuration. - */ -static const struct hda_input_mux cxt5045_test_capture_source = { - .num_items = 5, - .items = { - { "MIXER", 0x0 }, - { "MIC1 pin", 0x1 }, - { "LINE1 pin", 0x2 }, - { "HP-OUT pin", 0x3 }, - { "CD pin", 0x4 }, - }, -}; - -static const struct snd_kcontrol_new cxt5045_test_mixer[] = { - - /* Output controls */ - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("HP-OUT Playback Volume", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("HP-OUT Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("LINE1 Playback Switch", 0x12, 0x0, HDA_OUTPUT), - - /* Modes for retasking pin widgets */ - CXT_PIN_MODE("HP-OUT pin mode", 0x11, CXT_PIN_DIR_INOUT), - CXT_PIN_MODE("LINE1 pin mode", 0x12, CXT_PIN_DIR_INOUT), - - /* EAPD Switch Control */ - CXT_EAPD_SWITCH("External Amplifier", 0x10, 0x0), - - /* Loopback mixer controls */ - - HDA_CODEC_VOLUME("PCM Volume", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("PCM Switch", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("MIC1 pin Volume", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("MIC1 pin Switch", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("LINE1 pin Volume", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("LINE1 pin Switch", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("HP-OUT pin Volume", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("HP-OUT pin Switch", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("CD pin Volume", 0x17, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD pin Switch", 0x17, 0x4, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Input Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put, - }, - /* Audio input controls */ - HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_verb cxt5045_test_init_verbs[] = { - /* Set connections */ - { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 }, - { 0x11, AC_VERB_SET_CONNECT_SEL, 0x0 }, - { 0x12, AC_VERB_SET_CONNECT_SEL, 0x0 }, - /* Enable retasking pins as output, initially without power amp */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* Disable digital (SPDIF) pins initially, but users can enable - * them via a mixer switch. In the case of SPDIF-out, this initverb - * payload also sets the generation to 0, output to be in "consumer" - * PCM format, copyright asserted, no pre-emphasis and no validity - * control. - */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Unmute retasking pin widget output buffers since the default - * state appears to be output. As the pin mode is changed by the - * user the pin mode control will take care of enabling the pin's - * input/output buffers as needed. - */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Mute capture amp left and right */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - - /* Set ADC connection select to match default mixer setting (mic1 - * pin) - */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x17, AC_VERB_SET_CONNECT_SEL, 0x01}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Mic1 pin */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* Line pin */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* HP pin */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - - { } -}; -#endif - - -/* initialize jack-sensing, too */ -static int cxt5045_init(struct hda_codec *codec) -{ - conexant_init(codec); - cxt5045_hp_automute(codec); - return 0; -} - - -enum { - CXT5045_LAPTOP_HPSENSE, - CXT5045_LAPTOP_MICSENSE, - CXT5045_LAPTOP_HPMICSENSE, - CXT5045_BENQ, -#ifdef CONFIG_SND_DEBUG - CXT5045_TEST, -#endif - CXT5045_AUTO, - CXT5045_MODELS -}; - -static const char * const cxt5045_models[CXT5045_MODELS] = { - [CXT5045_LAPTOP_HPSENSE] = "laptop-hpsense", - [CXT5045_LAPTOP_MICSENSE] = "laptop-micsense", - [CXT5045_LAPTOP_HPMICSENSE] = "laptop-hpmicsense", - [CXT5045_BENQ] = "benq", -#ifdef CONFIG_SND_DEBUG - [CXT5045_TEST] = "test", -#endif - [CXT5045_AUTO] = "auto", -}; - -static const struct snd_pci_quirk cxt5045_cfg_tbl[] = { - SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), - SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), - SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505", - CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK_MASK(0x1631, 0xff00, 0xc100, "Packard Bell", - CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP_HPSENSE), - {} -}; - -static int patch_cxt5045(struct hda_codec *codec) -{ - struct conexant_spec *spec; - int board_config; - - board_config = snd_hda_check_board_config(codec, CXT5045_MODELS, - cxt5045_models, - cxt5045_cfg_tbl); - if (board_config < 0) - board_config = CXT5045_AUTO; /* model=auto as default */ - if (board_config == CXT5045_AUTO) - return patch_conexant_auto(codec); - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; - codec->spec = spec; - codec->single_adc_amp = 1; - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids); - spec->multiout.dac_nids = cxt5045_dac_nids; - spec->multiout.dig_out_nid = CXT5045_SPDIF_OUT; - spec->num_adc_nids = 1; - spec->adc_nids = cxt5045_adc_nids; - spec->capsrc_nids = cxt5045_capsrc_nids; - spec->input_mux = &cxt5045_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = cxt5045_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = cxt5045_init_verbs; - spec->spdif_route = 0; - spec->num_channel_mode = ARRAY_SIZE(cxt5045_modes); - spec->channel_mode = cxt5045_modes; - - set_beep_amp(spec, 0x16, 0, 1); - - codec->patch_ops = conexant_patch_ops; - - switch (board_config) { - case CXT5045_LAPTOP_HPSENSE: - codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; - spec->input_mux = &cxt5045_capture_source; - spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5045_hp_sense_init_verbs; - spec->mixers[0] = cxt5045_mixers; - codec->patch_ops.init = cxt5045_init; - break; - case CXT5045_LAPTOP_MICSENSE: - codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; - spec->input_mux = &cxt5045_capture_source; - spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5045_mic_sense_init_verbs; - spec->mixers[0] = cxt5045_mixers; - codec->patch_ops.init = cxt5045_init; - break; - default: - case CXT5045_LAPTOP_HPMICSENSE: - codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; - spec->input_mux = &cxt5045_capture_source; - spec->num_init_verbs = 3; - spec->init_verbs[1] = cxt5045_hp_sense_init_verbs; - spec->init_verbs[2] = cxt5045_mic_sense_init_verbs; - spec->mixers[0] = cxt5045_mixers; - codec->patch_ops.init = cxt5045_init; - break; - case CXT5045_BENQ: - codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; - spec->input_mux = &cxt5045_capture_source_benq; - spec->num_init_verbs = 1; - spec->init_verbs[0] = cxt5045_benq_init_verbs; - spec->mixers[0] = cxt5045_mixers; - spec->mixers[1] = cxt5045_benq_mixers; - spec->num_mixers = 2; - codec->patch_ops.init = cxt5045_init; - break; -#ifdef CONFIG_SND_DEBUG - case CXT5045_TEST: - spec->input_mux = &cxt5045_test_capture_source; - spec->mixers[0] = cxt5045_test_mixer; - spec->init_verbs[0] = cxt5045_test_init_verbs; - break; - -#endif - } - - switch (codec->subsystem_id >> 16) { - case 0x103c: - case 0x1631: - case 0x1734: - case 0x17aa: - /* HP, Packard Bell, Fujitsu-Siemens & Lenovo laptops have - * really bad sound over 0dB on NID 0x17. Fix max PCM level to - * 0 dB (originally it has 0x2b steps with 0dB offset 0x14) - */ - snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT, - (0x14 << AC_AMPCAP_OFFSET_SHIFT) | - (0x14 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - } - - if (spec->beep_amp) - snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); - - return 0; -} - - -/* Conexant 5047 specific */ -#define CXT5047_SPDIF_OUT 0x11 - -static const hda_nid_t cxt5047_dac_nids[1] = { 0x10 }; /* 0x1c */ -static const hda_nid_t cxt5047_adc_nids[1] = { 0x12 }; -static const hda_nid_t cxt5047_capsrc_nids[1] = { 0x1a }; - -static const struct hda_channel_mode cxt5047_modes[1] = { - { 2, NULL }, -}; - -static const struct hda_input_mux cxt5047_toshiba_capture_source = { - .num_items = 2, - .items = { - { "ExtMic", 0x2 }, - { "Line-In", 0x1 }, - } -}; - -/* turn on/off EAPD (+ mute HP) as a master switch */ -static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - if (!cxt_eapd_put(kcontrol, ucontrol)) - return 0; - - /* toggle internal speakers mute depending of presence of - * the headphone jack - */ - bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE; - /* NOTE: Conexat codec needs the index for *OUTPUT* amp of - * pin widgets unlike other codecs. In this case, we need to - * set index 0x01 for the volume from the mixer amp 0x19. - */ - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, - HDA_AMP_MUTE, bits); - bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - return 1; -} - -/* mute internal speaker if HP is plugged */ -static void cxt5047_hp_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - spec->hp_present = snd_hda_jack_detect(codec, 0x13); - - bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; - /* See the note in cxt5047_hp_master_sw_put */ - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, - HDA_AMP_MUTE, bits); -} - -/* toggle input of built-in and mic jack appropriately */ -static void cxt5047_hp_automic(struct hda_codec *codec) -{ - static const struct hda_verb mic_jack_on[] = { - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {} - }; - static const struct hda_verb mic_jack_off[] = { - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {} - }; - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x15); - if (present) - snd_hda_sequence_write(codec, mic_jack_on); - else - snd_hda_sequence_write(codec, mic_jack_off); -} - -/* unsolicited event for HP jack sensing */ -static void cxt5047_hp_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5047_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: - cxt5047_hp_automic(codec); - break; - } -} - -static const struct snd_kcontrol_new cxt5047_base_mixers[] = { - HDA_CODEC_VOLUME("Mic Playback Volume", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5047_hp_master_sw_put, - .private_value = 0x13, - }, - - {} -}; - -static const struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = { - /* See the note in cxt5047_hp_master_sw_put */ - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x01, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5047_hp_only_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x13, 0x00, HDA_OUTPUT), - { } /* end */ -}; - -static const struct hda_verb cxt5047_init_verbs[] = { - /* Line in, Mic, Built-in Mic */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 }, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 }, - /* HP, Speaker */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, /* mixer(0x19) */ - {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mixer(0x19) */ - /* Record selector: Mic */ - {0x12, AC_VERB_SET_CONNECT_SEL,0x03}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, - {0x1A, AC_VERB_SET_CONNECT_SEL,0x02}, - {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_OUTPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x00}, - {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_OUTPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x03}, - /* SPDIF route: PCM */ - { 0x18, AC_VERB_SET_CONNECT_SEL, 0x0 }, - /* Enable unsolicited events */ - {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -/* configuration for Toshiba Laptops */ -static const struct hda_verb cxt5047_toshiba_init_verbs[] = { - {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0}, /* default off */ - {} -}; - -/* Test configuration for debugging, modelled after the ALC260 test - * configuration. - */ -#ifdef CONFIG_SND_DEBUG -static const struct hda_input_mux cxt5047_test_capture_source = { - .num_items = 4, - .items = { - { "LINE1 pin", 0x0 }, - { "MIC1 pin", 0x1 }, - { "MIC2 pin", 0x2 }, - { "CD pin", 0x3 }, - }, -}; - -static const struct snd_kcontrol_new cxt5047_test_mixer[] = { - - /* Output only controls */ - HDA_CODEC_VOLUME("OutAmp-1 Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("OutAmp-1 Switch", 0x10,0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("OutAmp-2 Volume", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("OutAmp-2 Switch", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("HeadPhone Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("HeadPhone Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line1-Out Playback Volume", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line1-Out Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line2-Out Playback Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line2-Out Playback Switch", 0x15, 0x0, HDA_OUTPUT), - - /* Modes for retasking pin widgets */ - CXT_PIN_MODE("LINE1 pin mode", 0x14, CXT_PIN_DIR_INOUT), - CXT_PIN_MODE("MIC1 pin mode", 0x15, CXT_PIN_DIR_INOUT), - - /* EAPD Switch Control */ - CXT_EAPD_SWITCH("External Amplifier", 0x13, 0x0), - - /* Loopback mixer controls */ - HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x12, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("MIC1 Playback Switch", 0x12, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x12, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("MIC2 Playback Switch", 0x12, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("LINE Playback Volume", 0x12, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("LINE Playback Switch", 0x12, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x12, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x12, 0x04, HDA_INPUT), - - HDA_CODEC_VOLUME("Capture-1 Volume", 0x19, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture-1 Switch", 0x19, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture-2 Volume", 0x19, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Capture-2 Switch", 0x19, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture-3 Volume", 0x19, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Capture-3 Switch", 0x19, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Capture-4 Volume", 0x19, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Capture-4 Switch", 0x19, 0x3, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Input Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put, - }, - HDA_CODEC_VOLUME("Mic Boost Volume", 0x1a, 0x0, HDA_OUTPUT), - - { } /* end */ -}; - -static const struct hda_verb cxt5047_test_init_verbs[] = { - /* Enable retasking pins as output, initially without power amp */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* Disable digital (SPDIF) pins initially, but users can enable - * them via a mixer switch. In the case of SPDIF-out, this initverb - * payload also sets the generation to 0, output to be in "consumer" - * PCM format, copyright asserted, no pre-emphasis and no validity - * control. - */ - {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Ensure mic1, mic2, line1 pin widgets take input from the - * OUT1 sum bus when acting as an output. - */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0}, - - /* Start with output sum widgets muted and their output gains at min */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - - /* Unmute retasking pin widget output buffers since the default - * state appears to be output. As the pin mode is changed by the - * user the pin mode control will take care of enabling the pin's - * input/output buffers as needed. - */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Mute capture amp left and right */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - - /* Set ADC connection select to match default mixer setting (mic1 - * pin) - */ - {0x12, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ - - { } -}; -#endif - - -/* initialize jack-sensing, too */ -static int cxt5047_hp_init(struct hda_codec *codec) -{ - conexant_init(codec); - cxt5047_hp_automute(codec); - return 0; -} - - -enum { - CXT5047_LAPTOP, /* Laptops w/o EAPD support */ - CXT5047_LAPTOP_HP, /* Some HP laptops */ - CXT5047_LAPTOP_EAPD, /* Laptops with EAPD support */ -#ifdef CONFIG_SND_DEBUG - CXT5047_TEST, -#endif - CXT5047_AUTO, - CXT5047_MODELS -}; - -static const char * const cxt5047_models[CXT5047_MODELS] = { - [CXT5047_LAPTOP] = "laptop", - [CXT5047_LAPTOP_HP] = "laptop-hp", - [CXT5047_LAPTOP_EAPD] = "laptop-eapd", -#ifdef CONFIG_SND_DEBUG - [CXT5047_TEST] = "test", -#endif - [CXT5047_AUTO] = "auto", -}; - -static const struct snd_pci_quirk cxt5047_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP), - SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series", - CXT5047_LAPTOP), - SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD), - {} -}; - -static int patch_cxt5047(struct hda_codec *codec) -{ - struct conexant_spec *spec; - int board_config; - - board_config = snd_hda_check_board_config(codec, CXT5047_MODELS, - cxt5047_models, - cxt5047_cfg_tbl); - if (board_config < 0) - board_config = CXT5047_AUTO; /* model=auto as default */ - if (board_config == CXT5047_AUTO) - return patch_conexant_auto(codec); - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; - codec->spec = spec; - codec->pin_amp_workaround = 1; - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(cxt5047_dac_nids); - spec->multiout.dac_nids = cxt5047_dac_nids; - spec->multiout.dig_out_nid = CXT5047_SPDIF_OUT; - spec->num_adc_nids = 1; - spec->adc_nids = cxt5047_adc_nids; - spec->capsrc_nids = cxt5047_capsrc_nids; - spec->num_mixers = 1; - spec->mixers[0] = cxt5047_base_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = cxt5047_init_verbs; - spec->spdif_route = 0; - spec->num_channel_mode = ARRAY_SIZE(cxt5047_modes), - spec->channel_mode = cxt5047_modes, - - codec->patch_ops = conexant_patch_ops; - - switch (board_config) { - case CXT5047_LAPTOP: - spec->num_mixers = 2; - spec->mixers[1] = cxt5047_hp_spk_mixers; - codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; - break; - case CXT5047_LAPTOP_HP: - spec->num_mixers = 2; - spec->mixers[1] = cxt5047_hp_only_mixers; - codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; - codec->patch_ops.init = cxt5047_hp_init; - break; - case CXT5047_LAPTOP_EAPD: - spec->input_mux = &cxt5047_toshiba_capture_source; - spec->num_mixers = 2; - spec->mixers[1] = cxt5047_hp_spk_mixers; - spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5047_toshiba_init_verbs; - codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; - break; -#ifdef CONFIG_SND_DEBUG - case CXT5047_TEST: - spec->input_mux = &cxt5047_test_capture_source; - spec->mixers[0] = cxt5047_test_mixer; - spec->init_verbs[0] = cxt5047_test_init_verbs; - codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; -#endif - } - spec->vmaster_nid = 0x13; - - switch (codec->subsystem_id >> 16) { - case 0x103c: - /* HP laptops have really bad sound over 0 dB on NID 0x10. - * Fix max PCM level to 0 dB (originally it has 0x1e steps - * with 0 dB offset 0x17) - */ - snd_hda_override_amp_caps(codec, 0x10, HDA_INPUT, - (0x17 << AC_AMPCAP_OFFSET_SHIFT) | - (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - } - - return 0; -} - -/* Conexant 5051 specific */ -static const hda_nid_t cxt5051_dac_nids[1] = { 0x10 }; -static const hda_nid_t cxt5051_adc_nids[2] = { 0x14, 0x15 }; - -static const struct hda_channel_mode cxt5051_modes[1] = { - { 2, NULL }, -}; - -static void cxt5051_update_speaker(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int pinctl; - /* headphone pin */ - pinctl = (spec->hp_present && spec->cur_eapd) ? PIN_HP : 0; - snd_hda_set_pin_ctl(codec, 0x16, pinctl); - /* speaker pin */ - pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; - snd_hda_set_pin_ctl(codec, 0x1a, pinctl); - /* on ideapad there is an additional speaker (subwoofer) to mute */ - if (spec->ideapad) - snd_hda_set_pin_ctl(codec, 0x1b, pinctl); -} - -/* turn on/off EAPD (+ mute HP) as a master switch */ -static int cxt5051_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - - if (!cxt_eapd_put(kcontrol, ucontrol)) - return 0; - cxt5051_update_speaker(codec); - return 1; -} - -/* toggle input of built-in and mic jack appropriately */ -static void cxt5051_portb_automic(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int present; - - if (!(spec->auto_mic & AUTO_MIC_PORTB)) - return; - present = snd_hda_jack_detect(codec, 0x17); - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_CONNECT_SEL, - present ? 0x01 : 0x00); -} - -/* switch the current ADC according to the jack state */ -static void cxt5051_portc_automic(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int present; - hda_nid_t new_adc; - - if (!(spec->auto_mic & AUTO_MIC_PORTC)) - return; - present = snd_hda_jack_detect(codec, 0x18); - if (present) - spec->cur_adc_idx = 1; - else - spec->cur_adc_idx = 0; - new_adc = spec->adc_nids[spec->cur_adc_idx]; - if (spec->cur_adc && spec->cur_adc != new_adc) { - /* stream is running, let's swap the current ADC */ - __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1); - spec->cur_adc = new_adc; - snd_hda_codec_setup_stream(codec, new_adc, - spec->cur_adc_stream_tag, 0, - spec->cur_adc_format); - } -} - -/* mute internal speaker if HP is plugged */ -static void cxt5051_hp_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - - spec->hp_present = snd_hda_jack_detect(codec, 0x16); - cxt5051_update_speaker(codec); -} - -/* unsolicited event for HP jack sensing */ -static void cxt5051_hp_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5051_hp_automute(codec); - break; - case CXT5051_PORTB_EVENT: - cxt5051_portb_automic(codec); - break; - case CXT5051_PORTC_EVENT: - cxt5051_portc_automic(codec); - break; - } -} - -static const struct snd_kcontrol_new cxt5051_playback_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5051_hp_master_sw_put, - .private_value = 0x1a, - }, - {} -}; - -static const struct snd_kcontrol_new cxt5051_capture_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Dock Mic Volume", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Dock Mic Switch", 0x15, 0x00, HDA_INPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5051_hp_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Volume", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Mic Switch", 0x15, 0x00, HDA_INPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x14, 0x00, HDA_INPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5051_f700_mixers[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x14, 0x01, HDA_INPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5051_toshiba_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), - {} -}; - -static const struct hda_verb cxt5051_init_verbs[] = { - /* Line in, Mic */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - /* SPK */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP, Amp */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Record selector: Internal mic */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, - /* SPDIF route: PCM */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* EAPD */ - {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5051_hp_dv6736_init_verbs[] = { - /* Line in, Mic */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, - /* SPK */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP, Amp */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Record selector: Internal mic */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* SPDIF route: PCM */ - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* EAPD */ - {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5051_f700_init_verbs[] = { - /* Line in, Mic */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, - /* SPK */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP, Amp */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Record selector: Internal mic */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* SPDIF route: PCM */ - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* EAPD */ - {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, - { } /* end */ -}; - -static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid, - unsigned int event) -{ - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | event); -} - -static const struct hda_verb cxt5051_ideapad_init_verbs[] = { - /* Subwoofer */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - { } /* end */ -}; - -/* initialize jack-sensing, too */ -static int cxt5051_init(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - - conexant_init(codec); - - if (spec->auto_mic & AUTO_MIC_PORTB) - cxt5051_init_mic_port(codec, 0x17, CXT5051_PORTB_EVENT); - if (spec->auto_mic & AUTO_MIC_PORTC) - cxt5051_init_mic_port(codec, 0x18, CXT5051_PORTC_EVENT); - - if (codec->patch_ops.unsol_event) { - cxt5051_hp_automute(codec); - cxt5051_portb_automic(codec); - cxt5051_portc_automic(codec); - } - return 0; -} - - -enum { - CXT5051_LAPTOP, /* Laptops w/ EAPD support */ - CXT5051_HP, /* no docking */ - CXT5051_HP_DV6736, /* HP without mic switch */ - CXT5051_F700, /* HP Compaq Presario F700 */ - CXT5051_TOSHIBA, /* Toshiba M300 & co */ - CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */ - CXT5051_AUTO, /* auto-parser */ - CXT5051_MODELS -}; - -static const char *const cxt5051_models[CXT5051_MODELS] = { - [CXT5051_LAPTOP] = "laptop", - [CXT5051_HP] = "hp", - [CXT5051_HP_DV6736] = "hp-dv6736", - [CXT5051_F700] = "hp-700", - [CXT5051_TOSHIBA] = "toshiba", - [CXT5051_IDEAPAD] = "ideapad", - [CXT5051_AUTO] = "auto", -}; - -static const struct snd_pci_quirk cxt5051_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), - SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP), - SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700), - SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba M30x", CXT5051_TOSHIBA), - SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", - CXT5051_LAPTOP), - SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), - SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD), - {} -}; - -static int patch_cxt5051(struct hda_codec *codec) -{ - struct conexant_spec *spec; - int board_config; - - board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, - cxt5051_models, - cxt5051_cfg_tbl); - if (board_config < 0) - board_config = CXT5051_AUTO; /* model=auto as default */ - if (board_config == CXT5051_AUTO) - return patch_conexant_auto(codec); - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; - codec->spec = spec; - codec->pin_amp_workaround = 1; - - codec->patch_ops = conexant_patch_ops; - codec->patch_ops.init = cxt5051_init; - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(cxt5051_dac_nids); - spec->multiout.dac_nids = cxt5051_dac_nids; - spec->multiout.dig_out_nid = CXT5051_SPDIF_OUT; - spec->num_adc_nids = 1; /* not 2; via auto-mic switch */ - spec->adc_nids = cxt5051_adc_nids; - spec->num_mixers = 2; - spec->mixers[0] = cxt5051_capture_mixers; - spec->mixers[1] = cxt5051_playback_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = cxt5051_init_verbs; - spec->spdif_route = 0; - spec->num_channel_mode = ARRAY_SIZE(cxt5051_modes); - spec->channel_mode = cxt5051_modes; - spec->cur_adc = 0; - spec->cur_adc_idx = 0; - - set_beep_amp(spec, 0x13, 0, HDA_OUTPUT); - - codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; - - spec->auto_mic = AUTO_MIC_PORTB | AUTO_MIC_PORTC; - switch (board_config) { - case CXT5051_HP: - spec->mixers[0] = cxt5051_hp_mixers; - break; - case CXT5051_HP_DV6736: - spec->init_verbs[0] = cxt5051_hp_dv6736_init_verbs; - spec->mixers[0] = cxt5051_hp_dv6736_mixers; - spec->auto_mic = 0; - break; - case CXT5051_F700: - spec->init_verbs[0] = cxt5051_f700_init_verbs; - spec->mixers[0] = cxt5051_f700_mixers; - spec->auto_mic = 0; - break; - case CXT5051_TOSHIBA: - spec->mixers[0] = cxt5051_toshiba_mixers; - spec->auto_mic = AUTO_MIC_PORTB; - break; - case CXT5051_IDEAPAD: - spec->init_verbs[spec->num_init_verbs++] = - cxt5051_ideapad_init_verbs; - spec->ideapad = 1; - break; - } - - if (spec->beep_amp) - snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); - - return 0; -} - -/* Conexant 5066 specific */ - -static const hda_nid_t cxt5066_dac_nids[1] = { 0x10 }; -static const hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 }; -static const hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 }; -static const hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 }; - -static const struct hda_channel_mode cxt5066_modes[1] = { - { 2, NULL }, -}; - -#define HP_PRESENT_PORT_A (1 << 0) -#define HP_PRESENT_PORT_D (1 << 1) -#define hp_port_a_present(spec) ((spec)->hp_present & HP_PRESENT_PORT_A) -#define hp_port_d_present(spec) ((spec)->hp_present & HP_PRESENT_PORT_D) - -static void cxt5066_update_speaker(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int pinctl; - - codec_dbg(codec, - "CXT5066: update speaker, hp_present=%d, cur_eapd=%d\n", - spec->hp_present, spec->cur_eapd); - - /* Port A (HP) */ - pinctl = (hp_port_a_present(spec) && spec->cur_eapd) ? PIN_HP : 0; - snd_hda_set_pin_ctl(codec, 0x19, pinctl); - - /* Port D (HP/LO) */ - pinctl = spec->cur_eapd ? spec->port_d_mode : 0; - if (spec->dell_automute || spec->thinkpad) { - /* Mute if Port A is connected */ - if (hp_port_a_present(spec)) - pinctl = 0; - } else { - /* Thinkpad/Dell doesn't give pin-D status */ - if (!hp_port_d_present(spec)) - pinctl = 0; - } - snd_hda_set_pin_ctl(codec, 0x1c, pinctl); - - /* CLASS_D AMP */ - pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; - snd_hda_set_pin_ctl(codec, 0x1f, pinctl); -} - -/* turn on/off EAPD (+ mute HP) as a master switch */ -static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - - if (!cxt_eapd_put(kcontrol, ucontrol)) - return 0; - - cxt5066_update_speaker(codec); - return 1; -} - -/* toggle input of built-in digital mic and mic jack appropriately */ -static void cxt5066_vostro_automic(struct hda_codec *codec) -{ - unsigned int present; - - struct hda_verb ext_mic_present[] = { - /* enable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - - /* switch to external mic input */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0}, - - /* disable internal digital mic */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static const struct hda_verb ext_mic_absent[] = { - /* enable internal mic, port C */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - /* switch to internal mic input */ - {0x14, AC_VERB_SET_CONNECT_SEL, 2}, - - /* disable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - - present = snd_hda_jack_detect(codec, 0x1a); - if (present) { - codec_dbg(codec, "CXT5066: external microphone detected\n"); - snd_hda_sequence_write(codec, ext_mic_present); - } else { - codec_dbg(codec, "CXT5066: external microphone absent\n"); - snd_hda_sequence_write(codec, ext_mic_absent); - } -} - -/* toggle input of built-in digital mic and mic jack appropriately */ -static void cxt5066_ideapad_automic(struct hda_codec *codec) -{ - unsigned int present; - - struct hda_verb ext_mic_present[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static const struct hda_verb ext_mic_absent[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 2}, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - - present = snd_hda_jack_detect(codec, 0x1b); - if (present) { - codec_dbg(codec, "CXT5066: external microphone detected\n"); - snd_hda_sequence_write(codec, ext_mic_present); - } else { - codec_dbg(codec, "CXT5066: external microphone absent\n"); - snd_hda_sequence_write(codec, ext_mic_absent); - } -} - - -/* toggle input of built-in digital mic and mic jack appropriately */ -static void cxt5066_asus_automic(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x1b); - codec_dbg(codec, "CXT5066: external microphone present=%d\n", present); - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, - present ? 1 : 0); -} - - -/* toggle input of built-in digital mic and mic jack appropriately */ -static void cxt5066_hp_laptop_automic(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x1b); - codec_dbg(codec, "CXT5066: external microphone present=%d\n", present); - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, - present ? 1 : 3); -} - - -/* toggle input of built-in digital mic and mic jack appropriately - order is: external mic -> dock mic -> interal mic */ -static void cxt5066_thinkpad_automic(struct hda_codec *codec) -{ - unsigned int ext_present, dock_present; - - static const struct hda_verb ext_mic_present[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0}, - {0x17, AC_VERB_SET_CONNECT_SEL, 1}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static const struct hda_verb dock_mic_present[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0}, - {0x17, AC_VERB_SET_CONNECT_SEL, 0}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static const struct hda_verb ext_mic_absent[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 2}, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - - ext_present = snd_hda_jack_detect(codec, 0x1b); - dock_present = snd_hda_jack_detect(codec, 0x1a); - if (ext_present) { - codec_dbg(codec, "CXT5066: external microphone detected\n"); - snd_hda_sequence_write(codec, ext_mic_present); - } else if (dock_present) { - codec_dbg(codec, "CXT5066: dock microphone detected\n"); - snd_hda_sequence_write(codec, dock_mic_present); - } else { - codec_dbg(codec, "CXT5066: external microphone absent\n"); - snd_hda_sequence_write(codec, ext_mic_absent); - } -} - -/* mute internal speaker if HP is plugged */ -static void cxt5066_hp_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int portA, portD; - - /* Port A */ - portA = snd_hda_jack_detect(codec, 0x19); - - /* Port D */ - portD = snd_hda_jack_detect(codec, 0x1c); - - spec->hp_present = portA ? HP_PRESENT_PORT_A : 0; - spec->hp_present |= portD ? HP_PRESENT_PORT_D : 0; - codec_dbg(codec, "CXT5066: hp automute portA=%x portD=%x present=%d\n", - portA, portD, spec->hp_present); - cxt5066_update_speaker(codec); -} - -/* Dispatch the right mic autoswitch function */ -static void cxt5066_automic(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - - if (spec->dell_vostro) - cxt5066_vostro_automic(codec); - else if (spec->ideapad) - cxt5066_ideapad_automic(codec); - else if (spec->thinkpad) - cxt5066_thinkpad_automic(codec); - else if (spec->hp_laptop) - cxt5066_hp_laptop_automic(codec); - else if (spec->asus) - cxt5066_asus_automic(codec); -} - -/* unsolicited event for jack sensing */ -static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) -{ - codec_dbg(codec, "CXT5066: unsol event %x (%x)\n", res, res >> 26); - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5066_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: - cxt5066_automic(codec); - break; - } -} - - -static const struct hda_input_mux cxt5066_analog_mic_boost = { - .num_items = 5, - .items = { - { "0dB", 0 }, - { "10dB", 1 }, - { "20dB", 2 }, - { "30dB", 3 }, - { "40dB", 4 }, - }, -}; - -static void cxt5066_set_mic_boost(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - snd_hda_codec_write_cache(codec, 0x17, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | - cxt5066_analog_mic_boost.items[spec->mic_boost].index); - if (spec->ideapad || spec->thinkpad) { - /* adjust the internal mic as well...it is not through 0x17 */ - snd_hda_codec_write_cache(codec, 0x23, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_INPUT | - cxt5066_analog_mic_boost. - items[spec->mic_boost].index); - } -} - -static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - return snd_hda_input_mux_info(&cxt5066_analog_mic_boost, uinfo); -} - -static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - ucontrol->value.enumerated.item[0] = spec->mic_boost; - return 0; -} - -static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct conexant_spec *spec = codec->spec; - const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; - unsigned int idx; - idx = ucontrol->value.enumerated.item[0]; - if (idx >= imux->num_items) - idx = imux->num_items - 1; - - spec->mic_boost = idx; - cxt5066_set_mic_boost(codec); - return 1; -} - -static void conexant_check_dig_outs(struct hda_codec *codec, - const hda_nid_t *dig_pins, - int num_pins) -{ - struct conexant_spec *spec = codec->spec; - hda_nid_t *nid_loc = &spec->multiout.dig_out_nid; - int i; - - for (i = 0; i < num_pins; i++, dig_pins++) { - unsigned int cfg = snd_hda_codec_get_pincfg(codec, *dig_pins); - if (get_defcfg_connect(cfg) == AC_JACK_PORT_NONE) - continue; - if (snd_hda_get_connections(codec, *dig_pins, nid_loc, 1) != 1) - continue; - } -} - -static const struct hda_input_mux cxt5066_capture_source = { - .num_items = 4, - .items = { - { "Mic B", 0 }, - { "Mic C", 1 }, - { "Mic E", 2 }, - { "Mic F", 3 }, - }, -}; - -static const struct hda_bind_ctls cxt5066_bind_capture_vol_others = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT), - 0 - }, -}; - -static const struct hda_bind_ctls cxt5066_bind_capture_sw_others = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT), - 0 - }, -}; - -static const struct snd_kcontrol_new cxt5066_mixer_master[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), - {} -}; - -static const struct snd_kcontrol_new cxt5066_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5066_hp_master_sw_put, - .private_value = 0x1d, - }, - - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Mic Boost Capture Enum", - .info = cxt5066_mic_boost_mux_enum_info, - .get = cxt5066_mic_boost_mux_enum_get, - .put = cxt5066_mic_boost_mux_enum_put, - }, - - HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others), - HDA_BIND_SW("Capture Switch", &cxt5066_bind_capture_sw_others), - {} -}; - -static const struct snd_kcontrol_new cxt5066_vostro_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Internal Mic Boost Capture Enum", - .info = cxt5066_mic_boost_mux_enum_info, - .get = cxt5066_mic_boost_mux_enum_get, - .put = cxt5066_mic_boost_mux_enum_put, - .private_value = 0x23 | 0x100, - }, - {} -}; - -static const struct hda_verb cxt5066_init_verbs[] = { - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ - {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ - - /* Speakers */ - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* HP, Amp */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - - /* no digital microphone support yet */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Audio input selector */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, - - /* SPDIF route: PCM */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, - - {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* EAPD */ - {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - - /* not handling these yet */ - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, - { } /* end */ -}; - -static const struct hda_verb cxt5066_init_verbs_vostro[] = { - /* Port A: headphones */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* Port B: external microphone */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port C: unused */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port D: unused */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port E: unused, but has primary EAPD */ - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - - /* Port F: unused */ - {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port G: internal speakers */ - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* DAC2: unused */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - - /* Digital microphone port */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - /* Audio input selectors */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - - /* Disable SPDIF */ - {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* enable unsolicited events for Port A and B */ - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5066_init_verbs_ideapad[] = { - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ - {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ - - /* Speakers */ - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* HP, Amp */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x14, AC_VERB_SET_CONNECT_SEL, 2}, /* default to internal mic */ - - /* Audio input selector */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x2}, - {0x17, AC_VERB_SET_CONNECT_SEL, 1}, /* route ext mic */ - - /* SPDIF route: PCM */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, - - {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* internal microphone */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* enable internal mic */ - - /* EAPD */ - {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5066_init_verbs_thinkpad[] = { - {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ - - /* Port G: internal speakers */ - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* Port A: HP, Amp */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* Port B: Mic Dock */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port C: Mic */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - - /* Port D: HP Dock, Amp */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ - - /* DAC1 */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x14, AC_VERB_SET_CONNECT_SEL, 2}, /* default to internal mic */ - - /* Audio input selector */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x2}, - {0x17, AC_VERB_SET_CONNECT_SEL, 1}, /* route ext mic */ - - /* SPDIF route: PCM */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, - - {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* internal microphone */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* enable internal mic */ - - /* EAPD */ - {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - - /* enable unsolicited events for Port A, B, C and D */ - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -static const struct hda_verb cxt5066_init_verbs_portd_lo[] = { - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - { } /* end */ -}; - - -static const struct hda_verb cxt5066_init_verbs_hp_laptop[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - { } /* end */ -}; - -/* initialize jack-sensing, too */ -static int cxt5066_init(struct hda_codec *codec) -{ - codec_dbg(codec, "CXT5066: init\n"); - conexant_init(codec); - if (codec->patch_ops.unsol_event) { - cxt5066_hp_automute(codec); - cxt5066_automic(codec); - } - cxt5066_set_mic_boost(codec); - return 0; -} - -enum { - CXT5066_LAPTOP, /* Laptops w/ EAPD support */ - CXT5066_DELL_LAPTOP, /* Dell Laptop */ - CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */ - CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ - CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ - CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ - CXT5066_HP_LAPTOP, /* HP Laptop */ - CXT5066_AUTO, /* BIOS auto-parser */ - CXT5066_MODELS -}; - -static const char * const cxt5066_models[CXT5066_MODELS] = { - [CXT5066_LAPTOP] = "laptop", - [CXT5066_DELL_LAPTOP] = "dell-laptop", - [CXT5066_DELL_VOSTRO] = "dell-vostro", - [CXT5066_IDEAPAD] = "ideapad", - [CXT5066_THINKPAD] = "thinkpad", - [CXT5066_ASUS] = "asus", - [CXT5066_HP_LAPTOP] = "hp-laptop", - [CXT5066_AUTO] = "auto", -}; - -static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { - SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), - SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), - SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), - SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS), - SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), - SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS), - SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", - CXT5066_LAPTOP), - SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), - SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), - SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), - {} -}; - -static int patch_cxt5066(struct hda_codec *codec) -{ - struct conexant_spec *spec; - int board_config; - - board_config = snd_hda_check_board_config(codec, CXT5066_MODELS, - cxt5066_models, cxt5066_cfg_tbl); - if (board_config < 0) - board_config = CXT5066_AUTO; /* model=auto as default */ - if (board_config == CXT5066_AUTO) - return patch_conexant_auto(codec); - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; - codec->spec = spec; - - codec->patch_ops = conexant_patch_ops; - codec->patch_ops.init = conexant_init; - - spec->dell_automute = 0; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids); - spec->multiout.dac_nids = cxt5066_dac_nids; - conexant_check_dig_outs(codec, cxt5066_digout_pin_nids, - ARRAY_SIZE(cxt5066_digout_pin_nids)); - spec->num_adc_nids = 1; - spec->adc_nids = cxt5066_adc_nids; - spec->capsrc_nids = cxt5066_capsrc_nids; - spec->input_mux = &cxt5066_capture_source; - - spec->port_d_mode = PIN_HP; - - spec->num_init_verbs = 1; - spec->init_verbs[0] = cxt5066_init_verbs; - spec->num_channel_mode = ARRAY_SIZE(cxt5066_modes); - spec->channel_mode = cxt5066_modes; - spec->cur_adc = 0; - spec->cur_adc_idx = 0; - - set_beep_amp(spec, 0x13, 0, HDA_OUTPUT); - - switch (board_config) { - default: - case CXT5066_LAPTOP: - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - break; - case CXT5066_DELL_LAPTOP: - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - - spec->port_d_mode = PIN_OUT; - spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_portd_lo; - spec->num_init_verbs++; - spec->dell_automute = 1; - break; - case CXT5066_ASUS: - case CXT5066_HP_LAPTOP: - codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_unsol_event; - spec->init_verbs[spec->num_init_verbs] = - cxt5066_init_verbs_hp_laptop; - spec->num_init_verbs++; - spec->hp_laptop = board_config == CXT5066_HP_LAPTOP; - spec->asus = board_config == CXT5066_ASUS; - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - /* no S/PDIF out */ - if (board_config == CXT5066_HP_LAPTOP) - spec->multiout.dig_out_nid = 0; - /* input source automatically selected */ - spec->input_mux = NULL; - spec->port_d_mode = 0; - spec->mic_boost = 3; /* default 30dB gain */ - break; - - case CXT5066_DELL_VOSTRO: - codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_unsol_event; - spec->init_verbs[0] = cxt5066_init_verbs_vostro; - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers; - spec->port_d_mode = 0; - spec->dell_vostro = 1; - spec->mic_boost = 3; /* default 30dB gain */ - - /* no S/PDIF out */ - spec->multiout.dig_out_nid = 0; - - /* input source automatically selected */ - spec->input_mux = NULL; - break; - case CXT5066_IDEAPAD: - codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_unsol_event; - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - spec->init_verbs[0] = cxt5066_init_verbs_ideapad; - spec->port_d_mode = 0; - spec->ideapad = 1; - spec->mic_boost = 2; /* default 20dB gain */ - - /* no S/PDIF out */ - spec->multiout.dig_out_nid = 0; - - /* input source automatically selected */ - spec->input_mux = NULL; - break; - case CXT5066_THINKPAD: - codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_unsol_event; - spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; - spec->mixers[spec->num_mixers++] = cxt5066_mixers; - spec->init_verbs[0] = cxt5066_init_verbs_thinkpad; - spec->thinkpad = 1; - spec->port_d_mode = PIN_OUT; - spec->mic_boost = 2; /* default 20dB gain */ - - /* no S/PDIF out */ - spec->multiout.dig_out_nid = 0; - - /* input source automatically selected */ - spec->input_mux = NULL; - break; - } - - if (spec->beep_amp) - snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); - - return 0; -} - -#endif /* ENABLE_CXT_STATIC_QUIRKS */ - - /* * Automatic parser for CX20641 & co */ @@ -3487,35 +881,28 @@ static int patch_conexant_auto(struct hda_codec *codec) return err; } -#ifndef ENABLE_CXT_STATIC_QUIRKS -#define patch_cxt5045 patch_conexant_auto -#define patch_cxt5047 patch_conexant_auto -#define patch_cxt5051 patch_conexant_auto -#define patch_cxt5066 patch_conexant_auto -#endif - /* */ static const struct hda_codec_preset snd_hda_preset_conexant[] = { { .id = 0x14f15045, .name = "CX20549 (Venice)", - .patch = patch_cxt5045 }, + .patch = patch_conexant_auto }, { .id = 0x14f15047, .name = "CX20551 (Waikiki)", - .patch = patch_cxt5047 }, + .patch = patch_conexant_auto }, { .id = 0x14f15051, .name = "CX20561 (Hermosa)", - .patch = patch_cxt5051 }, + .patch = patch_conexant_auto }, { .id = 0x14f15066, .name = "CX20582 (Pebble)", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f15067, .name = "CX20583 (Pebble HSF)", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f15068, .name = "CX20584", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f15069, .name = "CX20585", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f1506c, .name = "CX20588", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f1506e, .name = "CX20590", - .patch = patch_cxt5066 }, + .patch = patch_conexant_auto }, { .id = 0x14f15097, .name = "CX20631", .patch = patch_conexant_auto }, { .id = 0x14f15098, .name = "CX20632", diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ba4ca52072ff..36badba2dcec 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -648,7 +648,8 @@ static int get_channel_allocation_order(int ca) * * TODO: it could select the wrong CA from multiple candidates. */ -static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels) +static int hdmi_channel_allocation(struct hda_codec *codec, + struct hdmi_eld *eld, int channels) { int i; int ca = 0; @@ -694,7 +695,7 @@ static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels) } snd_print_channel_allocation(eld->info.spk_alloc, buf, sizeof(buf)); - snd_printdd("HDMI: select CA 0x%x for %d-channel allocation: %s\n", + codec_dbg(codec, "HDMI: select CA 0x%x for %d-channel allocation: %s\n", ca, channels, buf); return ca; @@ -1131,7 +1132,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (!non_pcm && per_pin->chmap_set) ca = hdmi_manual_channel_allocation(channels, per_pin->chmap); else - ca = hdmi_channel_allocation(eld, channels); + ca = hdmi_channel_allocation(codec, eld, channels); if (ca < 0) ca = 0; @@ -1557,13 +1558,13 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) eld->eld_valid = false; else { memset(&eld->info, 0, sizeof(struct parsed_hdmi_eld)); - if (snd_hdmi_parse_eld(&eld->info, eld->eld_buffer, + if (snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer, eld->eld_size) < 0) eld->eld_valid = false; } if (eld->eld_valid) { - snd_hdmi_show_eld(&eld->info); + snd_hdmi_show_eld(codec, &eld->info); update_eld = true; } else if (repoll) { @@ -3355,6 +3356,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x80862808, .name = "Broadwell HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862882, .name = "Valleyview2 HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862883, .name = "Braswell HDMI", .patch = patch_generic_hdmi }, { .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, {} /* terminator */ }; @@ -3414,6 +3416,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862807"); MODULE_ALIAS("snd-hda-codec-id:80862808"); MODULE_ALIAS("snd-hda-codec-id:80862880"); MODULE_ALIAS("snd-hda-codec-id:80862882"); +MODULE_ALIAS("snd-hda-codec-id:80862883"); MODULE_ALIAS("snd-hda-codec-id:808629fb"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b60824e90408..654c8f16d150 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -101,6 +101,7 @@ struct alc_spec { /* 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() */ @@ -3402,7 +3403,8 @@ static unsigned int led_power_filter(struct hda_codec *codec, { struct alc_spec *spec = codec->spec; - if (power_state != AC_PWRST_D3 || nid != spec->mute_led_nid) + if (power_state != AC_PWRST_D3 || nid == 0 || + (nid != spec->mute_led_nid && nid != spec->cap_mute_led_nid)) return power_state; /* Set pin ctl again, it might have just been set to 0 */ @@ -3520,6 +3522,68 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, } } +/* 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) +{ + struct alc_spec *spec = codec->spec; + unsigned int pinval, enable, disable; + + 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; + else + pinval = enable; + + if (spec->cap_mute_led_nid) + 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 }, + {} + }; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gen.vmaster_mute.hook = alc269_fixup_hp_gpio_mute_hook; + spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook; + spec->gpio_led = 0; + spec->cap_mute_led_nid = 0x18; + snd_hda_add_verbs(codec, gpio_init); + codec->power_filter = led_power_filter; + } +} + +static void alc269_fixup_hp_line1_mic1_led(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->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; + } +} + static void alc_headset_mode_unplugged(struct hda_codec *codec) { int val; @@ -4231,6 +4295,9 @@ static void alc290_fixup_mono_speakers(struct hda_codec *codec, /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" +/* for dell wmi mic mute led */ +#include "dell_wmi_helper.c" + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -4255,6 +4322,8 @@ enum { ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC269_FIXUP_HP_MUTE_LED_MIC2, ALC269_FIXUP_HP_GPIO_LED, + ALC269_FIXUP_HP_GPIO_MIC1_LED, + ALC269_FIXUP_HP_LINE1_MIC1_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, ALC269_FIXUP_NO_SHUTUP, @@ -4292,6 +4361,8 @@ enum { ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, + ALC283_FIXUP_BXBT2807_MIC, + ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, }; static const struct hda_fixup alc269_fixups[] = { @@ -4447,6 +4518,14 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_gpio_led, }, + [ALC269_FIXUP_HP_GPIO_MIC1_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_hp_gpio_mic1_led, + }, + [ALC269_FIXUP_HP_LINE1_MIC1_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_hp_line1_mic1_led, + }, [ALC269_FIXUP_INV_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, @@ -4718,6 +4797,20 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, + [ALC283_FIXUP_BXBT2807_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x04a110f0 }, + { }, + }, + }, + [ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_dell_wmi, + .chained_before = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE + }, + }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -4727,7 +4820,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), - SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), @@ -4761,10 +4853,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), + SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED), SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -4782,6 +4876,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), /* ALC282 */ + SND_PCI_QUIRK(0x103c, 0x21f8, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x21f9, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x220d, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x220e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x220f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), @@ -4790,6 +4886,20 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2212, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2213, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2214, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2234, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2235, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2236, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2237, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2238, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2239, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2246, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2247, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2248, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2249, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x224a, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x224b, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x224c, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x224d, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2266, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2267, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2268, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), @@ -4814,13 +4924,43 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x22ce, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22cf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22d0, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22da, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x22db, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x22dc, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x22fb, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x8004, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), /* ALC290 */ + SND_PCI_QUIRK(0x103c, 0x221b, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x221c, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x221d, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2220, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2222, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2223, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2224, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2225, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2246, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2247, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2248, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2249, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2253, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2254, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2255, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2256, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2257, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2258, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2259, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x225a, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2260, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2261, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2262, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2263, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2264, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2265, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2273, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2277, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2278, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x227d, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x227e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x227f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), @@ -4843,7 +4983,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), - SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -4864,9 +5003,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9099, "Sony VAIO S13", ALC275_FIXUP_SONY_DISABLE_AAMIX), - SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), + SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_BXBT2807_MIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), @@ -4891,7 +5030,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), - SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", ALC269_FIXUP_THINKPAD_ACPI), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 @@ -4945,6 +5083,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { {} }; +static const struct snd_pci_quirk alc269_fixup_vendor_tbl[] = { + SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), + SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", ALC269_FIXUP_THINKPAD_ACPI), + {} +}; + static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_AMIC, .name = "laptop-amic"}, {.id = ALC269_FIXUP_DMIC, .name = "laptop-dmic"}, @@ -5040,6 +5186,17 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1d, 0x40700001}, {0x1e, 0x411111f0}, {0x21, 0x02211040}), + SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP 15 Touchsmart", ALC269_FIXUP_HP_MUTE_LED_MIC1, + {0x12, 0x99a30130}, + {0x14, 0x90170110}, + {0x17, 0x40000000}, + {0x18, 0x411111f0}, + {0x19, 0x03a11020}, + {0x1a, 0x411111f0}, + {0x1b, 0x411111f0}, + {0x1d, 0x40f41905}, + {0x1e, 0x411111f0}, + {0x21, 0x0321101f}), SND_HDA_PIN_QUIRK(0x10ec0283, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60130}, {0x14, 0x90170110}, @@ -5162,6 +5319,8 @@ static int patch_alc269(struct hda_codec *codec) 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); @@ -5858,6 +6017,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05fe, "Dell XPS 15", ALC668_FIXUP_DELL_XPS13), SND_PCI_QUIRK(0x1028, 0x060a, "Dell XPS 13", ALC668_FIXUP_DELL_XPS13), SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3744ea4e843d..ea823e1100da 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -84,6 +84,7 @@ enum { STAC_DELL_EQ, STAC_ALIENWARE_M17X, STAC_92HD89XX_HP_FRONT_JACK, + STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK, STAC_92HD73XX_MODELS }; @@ -103,6 +104,7 @@ enum { STAC_92HD83XXX_HP, STAC_HP_ENVY_BASS, STAC_HP_BNB13_EQ, + STAC_HP_ENVY_TS_BASS, STAC_92HD83XXX_MODELS }; @@ -1017,7 +1019,7 @@ static int stac_create_spdif_mux_ctls(struct hda_codec *codec) for (i = 0; i < num_cons; i++) { if (snd_BUG_ON(!labels[i])) return -EINVAL; - snd_hda_add_imux_item(&spec->spdif_mux, labels[i], i, NULL); + snd_hda_add_imux_item(codec, &spec->spdif_mux, labels[i], i, NULL); } kctl = snd_hda_gen_add_kctl(&spec->gen, NULL, &stac_smux_mixer); @@ -1809,6 +1811,11 @@ static const struct hda_pintbl stac92hd89xx_hp_front_jack_pin_configs[] = { {} }; +static const struct hda_pintbl stac92hd89xx_hp_z1_g2_right_mic_jack_pin_configs[] = { + { 0x0e, 0x400000f0 }, + {} +}; + static void stac92hd73xx_fixup_ref(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -1931,6 +1938,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = { [STAC_92HD89XX_HP_FRONT_JACK] = { .type = HDA_FIXUP_PINS, .v.pins = stac92hd89xx_hp_front_jack_pin_configs, + }, + [STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK] = { + .type = HDA_FIXUP_PINS, + .v.pins = stac92hd89xx_hp_z1_g2_right_mic_jack_pin_configs, } }; @@ -1991,6 +2002,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = { "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, "Alienware M17x R3", STAC_DELL_EQ), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1927, + "HP Z1 G2", STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17, "unknown HP", STAC_92HD89XX_HP_FRONT_JACK), {} /* terminator */ @@ -2668,6 +2681,13 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = { .chained = true, .chain_id = STAC_92HD83XXX_HP_MIC_LED, }, + [STAC_HP_ENVY_TS_BASS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x10, 0x92170111 }, + {} + }, + }, }; static const struct hda_model_fixup stac92hd83xxx_models[] = { @@ -2684,6 +2704,7 @@ static const struct hda_model_fixup stac92hd83xxx_models[] = { { .id = STAC_92HD83XXX_HEADSET_JACK, .name = "headset-jack" }, { .id = STAC_HP_ENVY_BASS, .name = "hp-envy-bass" }, { .id = STAC_HP_BNB13_EQ, .name = "hp-bnb13-eq" }, + { .id = STAC_HP_ENVY_TS_BASS, .name = "hp-envy-ts-bass" }, {} }; @@ -2739,6 +2760,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP bNB13", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x190A, "HP bNB13", STAC_HP_BNB13_EQ), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x190e, + "HP ENVY TS", STAC_HP_ENVY_TS_BASS), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1940, "HP bNB13", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1941, @@ -3438,9 +3461,11 @@ static void stac922x_fixup_intel_mac_auto(struct hda_codec *codec, { if (action != HDA_FIXUP_ACT_PRE_PROBE) return; + + codec->fixup_id = HDA_FIXUP_ID_NOT_SET; snd_hda_pick_fixup(codec, NULL, stac922x_intel_mac_fixup_tbl, stac922x_fixups); - if (codec->fixup_id != STAC_INTEL_MAC_AUTO) + if (codec->fixup_id != HDA_FIXUP_ID_NOT_SET) snd_hda_apply_fixup(codec, action); } diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index b209fc30b334..58f8f2ae758d 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -41,14 +41,17 @@ #define ICEREG(ice, x) ((ice)->port + ICE1712_REG_##x) #define ICE1712_REG_CONTROL 0x00 /* byte */ -#define ICE1712_RESET 0x80 /* reset whole chip */ -#define ICE1712_SERR_LEVEL 0x04 /* SERR# level otherwise edge */ +#define ICE1712_RESET 0x80 /* soft reset whole chip */ +#define ICE1712_SERR_ASSERT_DS_DMA 0x40 /* disabled SERR# assertion for the DS DMA Ch-C irq otherwise enabled */ +#define ICE1712_DOS_VOL 0x10 /* DOS WT/FM volume control */ +#define ICE1712_SERR_LEVEL 0x08 /* SERR# level otherwise edge */ +#define ICE1712_SERR_ASSERT_SB 0x02 /* disabled SERR# assertion for SB irq otherwise enabled */ #define ICE1712_NATIVE 0x01 /* native mode otherwise SB */ #define ICE1712_REG_IRQMASK 0x01 /* byte */ -#define ICE1712_IRQ_MPU1 0x80 -#define ICE1712_IRQ_TIMER 0x40 -#define ICE1712_IRQ_MPU2 0x20 -#define ICE1712_IRQ_PROPCM 0x10 +#define ICE1712_IRQ_MPU1 0x80 /* MIDI irq mask */ +#define ICE1712_IRQ_TIMER 0x40 /* Timer mask */ +#define ICE1712_IRQ_MPU2 0x20 /* Secondary MIDI irq mask */ +#define ICE1712_IRQ_PROPCM 0x10 /* professional multi-track */ #define ICE1712_IRQ_FM 0x08 /* FM/MIDI - legacy */ #define ICE1712_IRQ_PBKDS 0x04 /* playback DS channels */ #define ICE1712_IRQ_CONCAP 0x02 /* consumer capture */ diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 71f4bdcc4055..84f67450924e 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -151,13 +151,11 @@ static int send_msg( struct mixart_mgr *mgr, { u32 headptr, tailptr; u32 msg_frame_address; - int err, i; + int i; if (snd_BUG_ON(msg->size % 4)) return -EINVAL; - err = 0; - /* get message frame address */ tailptr = readl_be(MIXART_MEM(mgr, MSG_INBOUND_FREE_TAIL)); headptr = readl_be(MIXART_MEM(mgr, MSG_INBOUND_FREE_HEAD)); diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 64b9fda5f04a..dbbbacfd535e 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -53,6 +53,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, + { OXYGEN_PCI_SUBID(0x1043, 0x85f4) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index c8c7f2c9b355..e02605931669 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -100,8 +100,8 @@ */ /* - * Xonar Essence ST (Deluxe)/STX - * ----------------------------- + * Xonar Essence ST (Deluxe)/STX (II) + * ---------------------------------- * * CMI8788: * @@ -1138,6 +1138,14 @@ int get_xonar_pcm179x_model(struct oxygen *chip, chip->model.resume = xonar_stx_resume; chip->model.set_dac_params = set_pcm1796_params; break; + case 0x85f4: + chip->model = model_xonar_st; + /* TODO: daughterboard support */ + chip->model.shortname = "Xonar STX II"; + chip->model.init = xonar_stx_init; + chip->model.resume = xonar_stx_resume; + chip->model.set_dac_params = set_pcm1796_params; + break; default: return -EINVAL; } diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index b4a8278241b1..f0315c3f7de4 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -941,7 +941,7 @@ setmixer(struct cmdif *cif, short num, unsigned short rval, unsigned short lval) union cmdret rptr = CMDRET_ZERO; int i = 0; - snd_printdd("sent mixer %d: 0x%d 0x%d\n", num, rval, lval); + snd_printdd("sent mixer %d: 0x%x 0x%x\n", num, rval, lval); do { SEND_SDGV(cif, num, num, rval, lval); SEND_RDGV(cif, num, num, &rptr); @@ -1080,7 +1080,7 @@ getmixer(struct cmdif *cif, short num, unsigned short *rval, return -EIO; *rval = rptr.retwords[0]; *lval = rptr.retwords[1]; - snd_printdd("got mixer %d: 0x%d 0x%d\n", num, *rval, *lval); + snd_printdd("got mixer %d: 0x%x 0x%x\n", num, *rval, *lval); return 0; } diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 1272c18a2544..da875dced2ef 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3880,14 +3880,12 @@ void snd_trident_free_voice(struct snd_trident * trident, struct snd_trident_voi { unsigned long flags; void (*private_free)(struct snd_trident_voice *); - void *private_data; if (voice == NULL || !voice->use) return; snd_trident_clear_voices(trident, voice->number, voice->number); spin_lock_irqsave(&trident->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/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index 3102a579660b..04c474658e3c 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -139,12 +139,11 @@ static inline void *offset_ptr(struct snd_trident *trident, int offset) static struct snd_util_memblk * search_empty(struct snd_util_memhdr *hdr, int size) { - struct snd_util_memblk *blk, *prev; + struct snd_util_memblk *blk; int page, psize; struct list_head *p; psize = get_aligned_page(size + ALIGN_PAGE_SIZE -1); - prev = NULL; page = 0; list_for_each(p, &hdr->block) { blk = list_entry(p, struct snd_util_memblk, list); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 0060b31cc3f3..0e9623368ab0 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -47,6 +47,7 @@ source "sound/soc/kirkwood/Kconfig" source "sound/soc/intel/Kconfig" source "sound/soc/mxs/Kconfig" source "sound/soc/pxa/Kconfig" +source "sound/soc/rockchip/Kconfig" source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 5f1df02984f8..534714a1ca44 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -24,6 +24,7 @@ obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += kirkwood/ obj-$(CONFIG_SND_SOC) += pxa/ +obj-$(CONFIG_SND_SOC) += rockchip/ obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index de433cfd044c..f403f399808a 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -347,6 +347,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, u32 tfmr, rfmr, tcmr, rcmr; int start_event; int ret; + int fslen, fslen_ext; /* * Currently, there is only one set of dma params for @@ -388,18 +389,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, } /* - * The SSC only supports up to 16-bit samples in I2S format, due - * to the size of the Frame Mode Register FSLEN field. - */ - if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S - && bits > 16) { - printk(KERN_WARNING - "atmel_ssc_dai: sample size %d " - "is too large for I2S\n", bits); - return -EINVAL; - } - - /* * Compute SSC register settings. */ switch (ssc_p->daifmt @@ -413,6 +402,17 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, * from the MCK divider, and the BCLK signal * is output on the SSC TK line. */ + + if (bits > 16 && !ssc->pdata->has_fslen_ext) { + dev_err(dai->dev, + "sample size %d is too large for SSC device\n", + bits); + return -EINVAL; + } + + fslen_ext = (bits - 1) / 16; + fslen = (bits - 1) % 16; + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | SSC_BF(RCMR_STTDLY, START_DELAY) | SSC_BF(RCMR_START, SSC_START_FALLING_RF) @@ -420,9 +420,10 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, SSC_CKS_DIV); - rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + rfmr = SSC_BF(RFMR_FSLEN_EXT, fslen_ext) + | SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) - | SSC_BF(RFMR_FSLEN, (bits - 1)) + | SSC_BF(RFMR_FSLEN, fslen) | SSC_BF(RFMR_DATNB, (channels - 1)) | SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_LOOP, 0) @@ -435,10 +436,11 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | SSC_BF(TCMR_CKS, SSC_CKS_DIV); - tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + tfmr = SSC_BF(TFMR_FSLEN_EXT, fslen_ext) + | SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(TFMR_FSDEN, 0) | SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) - | SSC_BF(TFMR_FSLEN, (bits - 1)) + | SSC_BF(TFMR_FSLEN, fslen) | SSC_BF(TFMR_DATNB, (channels - 1)) | SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATDEF, 0) diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index b4e36901a40b..4052268ce462 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -18,10 +18,6 @@ #include "../codecs/wm8904.h" #include "atmel_ssc_dai.h" -#define MCLK_RATE 32768 - -static struct clk *mclk; - static const struct snd_soc_dapm_widget atmel_asoc_wm8904_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Mic", NULL), @@ -61,26 +57,6 @@ static struct snd_soc_ops atmel_asoc_wm8904_ops = { .hw_params = atmel_asoc_wm8904_hw_params, }; -static int atmel_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - switch (level) { - case SND_SOC_BIAS_PREPARE: - clk_prepare_enable(mclk); - break; - case SND_SOC_BIAS_OFF: - clk_disable_unprepare(mclk); - break; - default: - break; - } - } - - return 0; -}; - static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = { .name = "WM8904", .stream_name = "WM8904 PCM", @@ -94,7 +70,6 @@ static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = { static struct snd_soc_card atmel_asoc_wm8904_card = { .name = "atmel_asoc_wm8904", .owner = THIS_MODULE, - .set_bias_level = atmel_set_bias_level, .dai_link = &atmel_asoc_wm8904_dailink, .num_links = 1, .dapm_widgets = atmel_asoc_wm8904_dapm_widgets, @@ -153,7 +128,6 @@ static int atmel_asoc_wm8904_probe(struct platform_device *pdev) { struct snd_soc_card *card = &atmel_asoc_wm8904_card; struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; - struct clk *clk_src; int id, ret; card->dev = &pdev->dev; @@ -170,30 +144,6 @@ static int atmel_asoc_wm8904_probe(struct platform_device *pdev) return ret; } - mclk = clk_get(NULL, "pck0"); - if (IS_ERR(mclk)) { - dev_err(&pdev->dev, "failed to get pck0\n"); - ret = PTR_ERR(mclk); - goto err_set_audio; - } - - clk_src = clk_get(NULL, "clk32k"); - if (IS_ERR(clk_src)) { - dev_err(&pdev->dev, "failed to get clk32k\n"); - ret = PTR_ERR(clk_src); - goto err_set_audio; - } - - ret = clk_set_parent(mclk, clk_src); - clk_put(clk_src); - if (ret != 0) { - dev_err(&pdev->dev, "failed to set MCLK parent\n"); - goto err_set_audio; - } - - dev_info(&pdev->dev, "setting pck0 to %dHz\n", MCLK_RATE); - clk_set_rate(mclk, MCLK_RATE); - ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed\n"); diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index a3881c4381c9..bcf591373a7a 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -290,19 +290,19 @@ static int bf5xx_pcm_silence(struct snd_pcm_substream *substream, unsigned int sample_size = runtime->sample_bits / 8; void *buf = runtime->dma_area; struct bf5xx_i2s_pcm_data *dma_data; - unsigned int offset, size; + unsigned int offset, samples; dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if (dma_data->tdm_mode) { offset = pos * 8 * sample_size; - size = count * 8 * sample_size; + samples = count * 8; } else { offset = frames_to_bytes(runtime, pos); - size = frames_to_bytes(runtime, count); + samples = count * runtime->channels; } - snd_pcm_format_set_silence(runtime->format, buf + offset, size); + snd_pcm_format_set_silence(runtime->format, buf + offset, samples); return 0; } diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 3c4b10ff48c1..922006dd0583 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -945,11 +945,11 @@ static int pm860x_pcm_hw_params(struct snd_pcm_substream *substream, unsigned char inf = 0, mask = 0; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: inf &= ~PCM_INF2_18WL; break; - case SNDRV_PCM_FORMAT_S18_3LE: + case 18: inf |= PCM_INF2_18WL; break; default: @@ -1044,11 +1044,11 @@ static int pm860x_i2s_hw_params(struct snd_pcm_substream *substream, unsigned char inf; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: inf = 0; break; - case SNDRV_PCM_FORMAT_S18_3LE: + case 18: inf = PCM_INF2_18WL; break; default: diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0b9571c858f8..8838838e25ed 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -47,6 +47,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT select SND_SOC_CS42L73 if I2C + select SND_SOC_CS4265 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CS42XX8_I2C if I2C @@ -74,10 +75,12 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RT286 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_RT5645 if I2C select SND_SOC_RT5651 if I2C + select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE @@ -91,6 +94,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS + select SND_SOC_TAS2552 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23_I2C if I2C select SND_SOC_TLV320AIC23_SPI if SPI_MASTER @@ -338,6 +342,11 @@ config SND_SOC_CS42L73 tristate "Cirrus Logic CS42L73 CODEC" depends on I2C +config SND_SOC_CS4265 + tristate "Cirrus Logic CS4265 CODEC" + depends on I2C + select REGMAP_I2C + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate "Cirrus Logic CS4270 CODEC" @@ -445,9 +454,16 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5640=y default y if SND_SOC_RT5645=y default y if SND_SOC_RT5651=y + default y if SND_SOC_RT5670=y + default y if SND_SOC_RT5677=y default m if SND_SOC_RT5640=m default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m + default m if SND_SOC_RT5670=m + default m if SND_SOC_RT5677=m + +config SND_SOC_RT286 + tristate config SND_SOC_RT5631 tristate @@ -461,6 +477,9 @@ config SND_SOC_RT5645 config SND_SOC_RT5651 tristate +config SND_SOC_RT5670 + tristate + config SND_SOC_RT5677 tristate @@ -521,6 +540,10 @@ config SND_SOC_STA529 config SND_SOC_STAC9766 tristate +config SND_SOC_TAS2552 + tristate "Texas Instruments TAS2552 Mono Audio amplifier" + depends on I2C + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C @@ -541,7 +564,9 @@ config SND_SOC_TLV320AIC26 depends on SPI config SND_SOC_TLV320AIC31XX - tristate + tristate "Texas Instruments TLV320AIC31xx CODECs" + depends on I2C + select REGMAP_I2C config SND_SOC_TLV320AIC32X4 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1bd6e1cf6f82..20afe0f0c5be 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -37,6 +37,7 @@ snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l56-objs := cs42l56.o snd-soc-cs42l73-objs := cs42l73.o +snd-soc-cs4265-objs := cs4265.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cs42xx8-objs := cs42xx8.o @@ -68,10 +69,12 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rl6231-objs := rl6231.o +snd-soc-rt286-objs := rt286.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o +snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -162,6 +165,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o +snd-soc-tas2552-objs := tas2552.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o @@ -204,6 +208,7 @@ obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L56) += snd-soc-cs42l56.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o +obj-$(CONFIG_SND_SOC_CS4265) += snd-soc-cs4265.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o @@ -235,10 +240,12 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o +obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o @@ -255,6 +262,7 @@ obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o +obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 8d9ba4ba4bfe..e889e1b84192 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -89,8 +89,8 @@ static int ac97_soc_probe(struct snd_soc_codec *codec) int ret; /* add codec as bus device for standard ac97 */ - ret = snd_ac97_bus(codec->card->snd_card, 0, soc_ac97_ops, NULL, - &ac97_bus); + ret = snd_ac97_bus(codec->component.card->snd_card, 0, soc_ac97_ops, + NULL, &ac97_bus); if (ret < 0) return ret; diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index d71c59cf7bdd..370b742117ef 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -230,8 +230,10 @@ static int adau1701_reg_read(void *context, unsigned int reg, *value = 0; - for (i = 0; i < size; i++) - *value |= recv_buf[i] << (i * 8); + for (i = 0; i < size; i++) { + *value <<= 8; + *value |= recv_buf[i]; + } return 0; } diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 2961fae9670a..0b659704e60c 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -359,14 +359,14 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, if (adau->dai_fmt != SND_SOC_DAIFMT_RIGHT_J) return 0; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: val = ADAU17X1_SERIAL_PORT1_DELAY16; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: val = ADAU17X1_SERIAL_PORT1_DELAY8; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: val = ADAU17X1_SERIAL_PORT1_DELAY0; break; default: diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index fd55da7cb9d4..70ab35744aba 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -968,7 +968,7 @@ int adau1977_probe(struct device *dev, struct regmap *regmap, if (adau1977->dvdd_reg) power_off_mask = ~0; else - power_off_mask = ~ADAU1977_BLOCK_POWER_SAI_LDO_EN; + power_off_mask = (unsigned int)~ADAU1977_BLOCK_POWER_SAI_LDO_EN; ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI, power_off_mask, 0x00); diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 3ba4c0f11418..041712592e29 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -547,7 +547,7 @@ static const struct ak4642_drvdata ak4648_drvdata = { .extended_frequencies = 1, }; -static struct of_device_id ak4642_of_match[]; +static const struct of_device_id ak4642_of_match[]; static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -593,7 +593,7 @@ static int ak4642_i2c_remove(struct i2c_client *client) return 0; } -static struct of_device_id ak4642_of_match[] = { +static const struct of_device_id ak4642_of_match[] = { { .compatible = "asahi-kasei,ak4642", .data = &ak4642_drvdata}, { .compatible = "asahi-kasei,ak4643", .data = &ak4643_drvdata}, { .compatible = "asahi-kasei,ak4648", .data = &ak4648_drvdata}, diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 72e953b2cb41..8107a1cac876 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -14,12 +14,18 @@ #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/of_device.h> +#include <linux/regulator/consumer.h> #include <sound/soc.h> #include <sound/pcm.h> #include <sound/initval.h> +static const char * const supply_names[] = { + "va", "vd" +}; + struct ak5386_priv { int reset_gpio; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = { @@ -32,7 +38,42 @@ static const struct snd_soc_dapm_route ak5386_dapm_routes[] = { { "Capture", NULL, "AINR" }, }; +static int ak5386_soc_probe(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); +} + +static int ak5386_soc_remove(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; +} + +#ifdef CONFIG_PM +static int ak5386_soc_suspend(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; +} + +static int ak5386_soc_resume(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); +} +#else +#define ak5386_soc_suspend NULL +#define ak5386_soc_resume NULL +#endif /* CONFIG_PM */ + static struct snd_soc_codec_driver soc_codec_ak5386 = { + .probe = ak5386_soc_probe, + .remove = ak5386_soc_remove, + .suspend = ak5386_soc_suspend, + .resume = ak5386_soc_resume, .dapm_widgets = ak5386_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak5386_dapm_widgets), .dapm_routes = ak5386_dapm_routes, @@ -122,6 +163,7 @@ static int ak5386_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct ak5386_priv *priv; + int ret, i; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -130,6 +172,14 @@ static int ak5386_probe(struct platform_device *pdev) priv->reset_gpio = -EINVAL; dev_set_drvdata(dev, priv); + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) + return ret; + if (of_match_device(of_match_ptr(ak5386_dt_ids), dev)) priv->reset_gpio = of_get_named_gpio(dev->of_node, "reset-gpio", 0); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 29e198f57d4c..2f2e91ac690f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -243,6 +243,31 @@ int arizona_init_spk(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(arizona_init_spk); +static const struct snd_soc_dapm_route arizona_mono_routes[] = { + { "OUT1R", NULL, "OUT1L" }, + { "OUT2R", NULL, "OUT2L" }, + { "OUT3R", NULL, "OUT3L" }, + { "OUT4R", NULL, "OUT4L" }, + { "OUT5R", NULL, "OUT5L" }, + { "OUT6R", NULL, "OUT6L" }, +}; + +int arizona_init_mono(struct snd_soc_codec *codec) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int i; + + for (i = 0; i < ARIZONA_MAX_OUTPUT; ++i) { + if (arizona->pdata.out_mono[i]) + snd_soc_dapm_add_routes(&codec->dapm, + &arizona_mono_routes[i], 1); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_mono); + int arizona_init_gpio(struct snd_soc_codec *codec) { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); @@ -1127,6 +1152,31 @@ static int arizona_startup(struct snd_pcm_substream *substream, constraint); } +static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec, + unsigned int rate) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + struct reg_default dac_comp[] = { + { 0x80, 0x3 }, + { ARIZONA_DAC_COMP_1, 0 }, + { ARIZONA_DAC_COMP_2, 0 }, + { 0x80, 0x0 }, + }; + + mutex_lock(&codec->mutex); + + dac_comp[1].def = arizona->dac_comp_coeff; + if (rate >= 176400) + dac_comp[2].def = arizona->dac_comp_enabled; + + mutex_unlock(&codec->mutex); + + regmap_multi_reg_write(arizona->regmap, + dac_comp, + ARRAY_SIZE(dac_comp)); +} + static int arizona_hw_params_rate(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1153,6 +1203,15 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream, switch (dai_priv->clk) { case ARIZONA_CLK_SYSCLK: + switch (priv->arizona->type) { + case WM5102: + arizona_wm5102_set_dac_comp(codec, + params_rate(params)); + break; + default: + break; + } + snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, ARIZONA_SAMPLE_RATE_1_MASK, sr_val); if (base) @@ -1175,6 +1234,27 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream, return 0; } +static bool arizona_aif_cfg_changed(struct snd_soc_codec *codec, + int base, int bclk, int lrclk, int frame) +{ + int val; + + val = snd_soc_read(codec, base + ARIZONA_AIF_BCLK_CTRL); + if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK)) + return true; + + val = snd_soc_read(codec, base + ARIZONA_AIF_TX_BCLK_RATE); + if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK)) + return true; + + val = snd_soc_read(codec, base + ARIZONA_AIF_FRAME_CTRL_1); + if (frame != (val & (ARIZONA_AIF1TX_WL_MASK | + ARIZONA_AIF1TX_SLOT_LEN_MASK))) + return true; + + return false; +} + static int arizona_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1185,26 +1265,40 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, int base = dai->driver->base; const int *rates; int i, ret, val; + int channels = params_channels(params); int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1]; + int tdm_width = arizona->tdm_width[dai->id - 1]; + int tdm_slots = arizona->tdm_slots[dai->id - 1]; int bclk, lrclk, wl, frame, bclk_target; + bool reconfig; + unsigned int aif_tx_state, aif_rx_state; if (params_rate(params) % 8000) rates = &arizona_44k1_bclk_rates[0]; else rates = &arizona_48k_bclk_rates[0]; - bclk_target = snd_soc_params_to_bclk(params); - if (chan_limit && chan_limit < params_channels(params)) { + if (tdm_slots) { + arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", + tdm_slots, tdm_width); + bclk_target = tdm_slots * tdm_width * params_rate(params); + channels = tdm_slots; + } else { + bclk_target = snd_soc_params_to_bclk(params); + } + + if (chan_limit && chan_limit < channels) { arizona_aif_dbg(dai, "Limiting to %d channels\n", chan_limit); - bclk_target /= params_channels(params); + bclk_target /= channels; bclk_target *= chan_limit; } - /* Force stereo for I2S mode */ + /* Force multiple of 2 channels for I2S mode */ val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); - if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) { + if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) { arizona_aif_dbg(dai, "Forcing stereo mode\n"); - bclk_target *= 2; + bclk_target /= channels; + bclk_target *= channels + 1; } for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { @@ -1228,28 +1322,56 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, wl = snd_pcm_format_width(params_format(params)); frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame); + + if (reconfig) { + /* Save AIF TX/RX state */ + aif_tx_state = snd_soc_read(codec, + base + ARIZONA_AIF_TX_ENABLES); + aif_rx_state = snd_soc_read(codec, + base + ARIZONA_AIF_RX_ENABLES); + /* Disable AIF TX/RX before reconfiguring it */ + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_TX_ENABLES, 0xff, 0x0); + regmap_update_bits(arizona->regmap, + base + ARIZONA_AIF_RX_ENABLES, 0xff, 0x0); + } + ret = arizona_hw_params_rate(substream, params, dai); if (ret != 0) - return ret; + goto restore_aif; - regmap_update_bits_async(arizona->regmap, - base + ARIZONA_AIF_BCLK_CTRL, - ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); - regmap_update_bits_async(arizona->regmap, - base + ARIZONA_AIF_TX_BCLK_RATE, - ARIZONA_AIF1TX_BCPF_MASK, lrclk); - regmap_update_bits_async(arizona->regmap, - base + ARIZONA_AIF_RX_BCLK_RATE, - ARIZONA_AIF1RX_BCPF_MASK, lrclk); - regmap_update_bits_async(arizona->regmap, - base + ARIZONA_AIF_FRAME_CTRL_1, - ARIZONA_AIF1TX_WL_MASK | - ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); - regmap_update_bits(arizona->regmap, base + ARIZONA_AIF_FRAME_CTRL_2, - ARIZONA_AIF1RX_WL_MASK | - ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); + if (reconfig) { + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_TX_BCLK_RATE, + ARIZONA_AIF1TX_BCPF_MASK, lrclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_RX_BCLK_RATE, + ARIZONA_AIF1RX_BCPF_MASK, lrclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_FRAME_CTRL_1, + ARIZONA_AIF1TX_WL_MASK | + ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); + regmap_update_bits(arizona->regmap, + base + ARIZONA_AIF_FRAME_CTRL_2, + ARIZONA_AIF1RX_WL_MASK | + ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); + } - return 0; +restore_aif: + if (reconfig) { + /* Restore AIF TX/RX state */ + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_TX_ENABLES, + 0xff, aif_tx_state); + regmap_update_bits(arizona->regmap, + base + ARIZONA_AIF_RX_ENABLES, + 0xff, aif_rx_state); + } + return ret; } static const char *arizona_dai_clk_str(int clk_id) @@ -1324,9 +1446,63 @@ static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) ARIZONA_AIF1_TRI, reg); } +static void arizona_set_channels_to_mask(struct snd_soc_dai *dai, + unsigned int base, + int channels, unsigned int mask) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int slot, i; + + for (i = 0; i < channels; ++i) { + slot = ffs(mask) - 1; + if (slot < 0) + return; + + regmap_write(arizona->regmap, base + i, slot); + + mask &= ~(1 << slot); + } + + if (mask) + arizona_aif_warn(dai, "Too many channels in TDM mask\n"); +} + +static int arizona_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int base = dai->driver->base; + int rx_max_chan = dai->driver->playback.channels_max; + int tx_max_chan = dai->driver->capture.channels_max; + + /* Only support TDM for the physical AIFs */ + if (dai->id > ARIZONA_MAX_AIF) + return -ENOTSUPP; + + if (slots == 0) { + tx_mask = (1 << tx_max_chan) - 1; + rx_mask = (1 << rx_max_chan) - 1; + } + + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_3, + tx_max_chan, tx_mask); + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_11, + rx_max_chan, rx_mask); + + arizona->tdm_width[dai->id - 1] = slot_width; + arizona->tdm_slots[dai->id - 1] = slots; + + return 0; +} + const struct snd_soc_dai_ops arizona_dai_ops = { .startup = arizona_startup, .set_fmt = arizona_set_fmt, + .set_tdm_slot = arizona_set_tdm_slot, .hw_params = arizona_hw_params, .set_sysclk = arizona_dai_set_sysclk, .set_tristate = arizona_set_tristate, @@ -1400,6 +1576,12 @@ static int arizona_validate_fll(struct arizona_fll *fll, { unsigned int Fvco_min; + if (fll->fout && Fout != fll->fout) { + arizona_fll_err(fll, + "Can't change output on active FLL\n"); + return -EINVAL; + } + if (Fref / ARIZONA_FLL_MAX_REFDIV > ARIZONA_FLL_MAX_FREF) { arizona_fll_err(fll, "Can't scale %dMHz in to <=13.5MHz\n", @@ -1478,6 +1660,10 @@ static int arizona_calc_fratio(struct arizona_fll *fll, while (div <= ARIZONA_FLL_MAX_REFDIV) { for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; ratio++) { + if ((ARIZONA_FLL_VCO_CORNER / 2) / + (fll->vco_mult * ratio) < Fref) + break; + if (target % (ratio * Fref)) { cfg->refdiv = refdiv; cfg->fratio = ratio - 1; @@ -1485,11 +1671,7 @@ static int arizona_calc_fratio(struct arizona_fll *fll, } } - for (ratio = init_ratio - 1; ratio >= 0; ratio--) { - if (ARIZONA_FLL_VCO_CORNER / (fll->vco_mult * ratio) < - Fref) - break; - + for (ratio = init_ratio - 1; ratio > 0; ratio--) { if (target % (ratio * Fref)) { cfg->refdiv = refdiv; cfg->fratio = ratio - 1; @@ -1616,7 +1798,7 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base, ARIZONA_FLL1_CTRL_UPD | cfg->n); } -static bool arizona_is_enabled_fll(struct arizona_fll *fll) +static int arizona_is_enabled_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; unsigned int reg; @@ -1632,13 +1814,26 @@ static bool arizona_is_enabled_fll(struct arizona_fll *fll) return reg & ARIZONA_FLL1_ENA; } -static void arizona_enable_fll(struct arizona_fll *fll) +static int arizona_enable_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; int ret; bool use_sync = false; + int already_enabled = arizona_is_enabled_fll(fll); struct arizona_fll_cfg cfg; + if (already_enabled < 0) + return already_enabled; + + if (already_enabled) { + /* Facilitate smooth refclk across the transition */ + regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x7, + ARIZONA_FLL1_GAIN_MASK, 0); + regmap_update_bits_async(fll->arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, + ARIZONA_FLL1_FREERUN); + } + /* * If we have both REFCLK and SYNCCLK then enable both, * otherwise apply the SYNCCLK settings to REFCLK. @@ -1666,7 +1861,7 @@ static void arizona_enable_fll(struct arizona_fll *fll) ARIZONA_FLL1_SYNC_ENA, 0); } else { arizona_fll_err(fll, "No clocks provided\n"); - return; + return -EINVAL; } /* @@ -1681,25 +1876,29 @@ static void arizona_enable_fll(struct arizona_fll *fll) ARIZONA_FLL1_SYNC_BW, ARIZONA_FLL1_SYNC_BW); - if (!arizona_is_enabled_fll(fll)) + if (!already_enabled) pm_runtime_get(arizona->dev); /* Clear any pending completions */ try_wait_for_completion(&fll->ok); regmap_update_bits_async(arizona->regmap, fll->base + 1, - ARIZONA_FLL1_FREERUN, 0); - regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); if (use_sync) regmap_update_bits_async(arizona->regmap, fll->base + 0x11, ARIZONA_FLL1_SYNC_ENA, ARIZONA_FLL1_SYNC_ENA); + if (already_enabled) + regmap_update_bits_async(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, 0); + ret = wait_for_completion_timeout(&fll->ok, msecs_to_jiffies(250)); if (ret == 0) arizona_fll_warn(fll, "Timed out waiting for lock\n"); + + return 0; } static void arizona_disable_fll(struct arizona_fll *fll) @@ -1713,6 +1912,8 @@ static void arizona_disable_fll(struct arizona_fll *fll) ARIZONA_FLL1_ENA, 0, &change); regmap_update_bits(arizona->regmap, fll->base + 0x11, ARIZONA_FLL1_SYNC_ENA, 0); + regmap_update_bits_async(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, 0); if (change) pm_runtime_put_autosuspend(arizona->dev); @@ -1721,7 +1922,7 @@ static void arizona_disable_fll(struct arizona_fll *fll) int arizona_set_fll_refclk(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { - int ret; + int ret = 0; if (fll->ref_src == source && fll->ref_freq == Fref) return 0; @@ -1736,17 +1937,17 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, fll->ref_freq = Fref; if (fll->fout && Fref > 0) { - arizona_enable_fll(fll); + ret = arizona_enable_fll(fll); } - return 0; + return ret; } EXPORT_SYMBOL_GPL(arizona_set_fll_refclk); int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { - int ret; + int ret = 0; if (fll->sync_src == source && fll->sync_freq == Fref && fll->fout == Fout) @@ -1768,13 +1969,12 @@ int arizona_set_fll(struct arizona_fll *fll, int source, fll->sync_freq = Fref; fll->fout = Fout; - if (Fout) { - arizona_enable_fll(fll); - } else { + if (Fout) + ret = arizona_enable_fll(fll); + else arizona_disable_fll(fll); - } - return 0; + return ret; } EXPORT_SYMBOL_GPL(arizona_set_fll); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 05ae17f5bca3..942cfb197b6d 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -249,6 +249,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source, extern int arizona_init_spk(struct snd_soc_codec *codec); extern int arizona_init_gpio(struct snd_soc_codec *codec); +extern int arizona_init_mono(struct snd_soc_codec *codec); extern int arizona_init_dai(struct arizona_priv *priv, int dai); diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c new file mode 100644 index 000000000000..a20b30ca52c0 --- /dev/null +++ b/sound/soc/codecs/cs4265.c @@ -0,0 +1,682 @@ +/* + * cs4265.c -- CS4265 ALSA SoC audio driver + * + * Copyright 2014 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan@cirrus.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/kernel.h> +#include <linux/gpio/consumer.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs4265.h" + +struct cs4265_private { + struct device *dev; + struct regmap *regmap; + struct gpio_desc *reset_gpio; + u8 format; + u32 sysclk; +}; + +static const struct reg_default cs4265_reg_defaults[] = { + { CS4265_PWRCTL, 0x0F }, + { CS4265_DAC_CTL, 0x08 }, + { CS4265_ADC_CTL, 0x00 }, + { CS4265_MCLK_FREQ, 0x00 }, + { CS4265_SIG_SEL, 0x40 }, + { CS4265_CHB_PGA_CTL, 0x00 }, + { CS4265_CHA_PGA_CTL, 0x00 }, + { CS4265_ADC_CTL2, 0x19 }, + { CS4265_DAC_CHA_VOL, 0x00 }, + { CS4265_DAC_CHB_VOL, 0x00 }, + { CS4265_DAC_CTL2, 0xC0 }, + { CS4265_SPDIF_CTL1, 0x00 }, + { CS4265_SPDIF_CTL2, 0x00 }, + { CS4265_INT_MASK, 0x00 }, + { CS4265_STATUS_MODE_MSB, 0x00 }, + { CS4265_STATUS_MODE_LSB, 0x00 }, +}; + +static bool cs4265_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4265_PWRCTL: + case CS4265_DAC_CTL: + case CS4265_ADC_CTL: + case CS4265_MCLK_FREQ: + case CS4265_SIG_SEL: + case CS4265_CHB_PGA_CTL: + case CS4265_CHA_PGA_CTL: + case CS4265_ADC_CTL2: + case CS4265_DAC_CHA_VOL: + case CS4265_DAC_CHB_VOL: + case CS4265_DAC_CTL2: + case CS4265_SPDIF_CTL1: + case CS4265_SPDIF_CTL2: + case CS4265_INT_MASK: + case CS4265_STATUS_MODE_MSB: + case CS4265_STATUS_MODE_LSB: + return true; + default: + return false; + } +} + +static bool cs4265_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4265_INT_STATUS: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(pga_tlv, -1200, 50, 0); + +static DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 0); + +static const char * const digital_input_mux_text[] = { + "SDIN1", "SDIN2" +}; + +static SOC_ENUM_SINGLE_DECL(digital_input_mux_enum, CS4265_SIG_SEL, 7, + digital_input_mux_text); + +static const struct snd_kcontrol_new digital_input_mux = + SOC_DAPM_ENUM("Digital Input Mux", digital_input_mux_enum); + +static const char * const mic_linein_text[] = { + "MIC", "LINEIN" +}; + +static SOC_ENUM_SINGLE_DECL(mic_linein_enum, CS4265_ADC_CTL2, 0, + mic_linein_text); + +static const char * const cam_mode_text[] = { + "One Byte", "Two Byte" +}; + +static SOC_ENUM_SINGLE_DECL(cam_mode_enum, CS4265_SPDIF_CTL1, 5, + cam_mode_text); + +static const char * const cam_mono_stereo_text[] = { + "Stereo", "Mono" +}; + +static SOC_ENUM_SINGLE_DECL(spdif_mono_stereo_enum, CS4265_SPDIF_CTL2, 2, + cam_mono_stereo_text); + +static const char * const mono_select_text[] = { + "Channel A", "Channel B" +}; + +static SOC_ENUM_SINGLE_DECL(spdif_mono_select_enum, CS4265_SPDIF_CTL2, 0, + mono_select_text); + +static const struct snd_kcontrol_new mic_linein_mux = + SOC_DAPM_ENUM("ADC Input Capture Mux", mic_linein_enum); + +static const struct snd_kcontrol_new loopback_ctl = + SOC_DAPM_SINGLE("Switch", CS4265_SIG_SEL, 1, 1, 0); + +static const struct snd_kcontrol_new spdif_switch = + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 0, 0); + +static const struct snd_kcontrol_new dac_switch = + SOC_DAPM_SINGLE("Switch", CS4265_PWRCTL, 1, 1, 0); + +static const struct snd_kcontrol_new cs4265_snd_controls[] = { + + SOC_DOUBLE_R_SX_TLV("PGA Volume", CS4265_CHA_PGA_CTL, + CS4265_CHB_PGA_CTL, 0, 0x28, 0x30, pga_tlv), + SOC_DOUBLE_R_TLV("DAC Volume", CS4265_DAC_CHA_VOL, + CS4265_DAC_CHB_VOL, 0, 0xFF, 1, dac_tlv), + SOC_SINGLE("De-emp 44.1kHz Switch", CS4265_DAC_CTL, 1, + 1, 0), + SOC_SINGLE("DAC INV Switch", CS4265_DAC_CTL2, 5, + 1, 0), + SOC_SINGLE("DAC Zero Cross Switch", CS4265_DAC_CTL2, 6, + 1, 0), + SOC_SINGLE("DAC Soft Ramp Switch", CS4265_DAC_CTL2, 7, + 1, 0), + SOC_SINGLE("ADC HPF Switch", CS4265_ADC_CTL, 1, + 1, 0), + SOC_SINGLE("ADC Zero Cross Switch", CS4265_ADC_CTL2, 3, + 1, 1), + SOC_SINGLE("ADC Soft Ramp Switch", CS4265_ADC_CTL2, 7, + 1, 0), + SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1, + 6, 1, 0), + SOC_ENUM("C Data Access", cam_mode_enum), + SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, + 3, 1, 0), + SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), + SOC_SINGLE("MMTLR Data Switch", 0, + 1, 1, 0), + SOC_ENUM("Mono Channel Select", spdif_mono_select_enum), + SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24), +}; + +static const struct snd_soc_dapm_widget cs4265_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("LINEINL"), + SND_SOC_DAPM_INPUT("LINEINR"), + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + + SND_SOC_DAPM_AIF_OUT("DOUT", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SPDIFOUT", NULL, 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("ADC Mux", SND_SOC_NOPM, 0, 0, &mic_linein_mux), + + SND_SOC_DAPM_ADC("ADC", NULL, CS4265_PWRCTL, 2, 1), + SND_SOC_DAPM_PGA("Pre-amp MIC", CS4265_PWRCTL, 3, + 1, NULL, 0), + + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, + 0, 0, &digital_input_mux), + + SND_SOC_DAPM_MIXER("SDIN1 Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SDIN2 Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SPDIF Transmitter", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Loopback", SND_SOC_NOPM, 0, 0, + &loopback_ctl), + SND_SOC_DAPM_SWITCH("SPDIF", SND_SOC_NOPM, 0, 0, + &spdif_switch), + SND_SOC_DAPM_SWITCH("DAC", CS4265_PWRCTL, 1, 1, + &dac_switch), + + SND_SOC_DAPM_AIF_IN("DIN1", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DIN2", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("TXIN", NULL, 0, + CS4265_SPDIF_CTL2, 5, 1), + + SND_SOC_DAPM_OUTPUT("LINEOUTL"), + SND_SOC_DAPM_OUTPUT("LINEOUTR"), + +}; + +static const struct snd_soc_dapm_route cs4265_audio_map[] = { + + {"DIN1", NULL, "DAI1 Playback"}, + {"DIN2", NULL, "DAI2 Playback"}, + {"SDIN1 Input Mixer", NULL, "DIN1"}, + {"SDIN2 Input Mixer", NULL, "DIN2"}, + {"Input Mux", "SDIN1", "SDIN1 Input Mixer"}, + {"Input Mux", "SDIN2", "SDIN2 Input Mixer"}, + {"DAC", "Switch", "Input Mux"}, + {"SPDIF", "Switch", "Input Mux"}, + {"LINEOUTL", NULL, "DAC"}, + {"LINEOUTR", NULL, "DAC"}, + {"SPDIFOUT", NULL, "SPDIF"}, + + {"ADC Mux", "LINEIN", "LINEINL"}, + {"ADC Mux", "LINEIN", "LINEINR"}, + {"ADC Mux", "MIC", "MICL"}, + {"ADC Mux", "MIC", "MICR"}, + {"ADC", NULL, "ADC Mux"}, + {"DOUT", NULL, "ADC"}, + {"DAI1 Capture", NULL, "DOUT"}, + {"DAI2 Capture", NULL, "DOUT"}, + + /* Loopback */ + {"Loopback", "Switch", "ADC"}, + {"DAC", NULL, "Loopback"}, +}; + +struct cs4265_clk_para { + u32 mclk; + u32 rate; + u8 fm_mode; /* values 1, 2, or 4 */ + u8 mclkdiv; +}; + +static const struct cs4265_clk_para clk_map_table[] = { + /*32k*/ + {8192000, 32000, 0, 0}, + {12288000, 32000, 0, 1}, + {16384000, 32000, 0, 2}, + {24576000, 32000, 0, 3}, + {32768000, 32000, 0, 4}, + + /*44.1k*/ + {11289600, 44100, 0, 0}, + {16934400, 44100, 0, 1}, + {22579200, 44100, 0, 2}, + {33868000, 44100, 0, 3}, + {45158400, 44100, 0, 4}, + + /*48k*/ + {12288000, 48000, 0, 0}, + {18432000, 48000, 0, 1}, + {24576000, 48000, 0, 2}, + {36864000, 48000, 0, 3}, + {49152000, 48000, 0, 4}, + + /*64k*/ + {8192000, 64000, 1, 0}, + {1228800, 64000, 1, 1}, + {1693440, 64000, 1, 2}, + {2457600, 64000, 1, 3}, + {3276800, 64000, 1, 4}, + + /* 88.2k */ + {11289600, 88200, 1, 0}, + {16934400, 88200, 1, 1}, + {22579200, 88200, 1, 2}, + {33868000, 88200, 1, 3}, + {45158400, 88200, 1, 4}, + + /* 96k */ + {12288000, 96000, 1, 0}, + {18432000, 96000, 1, 1}, + {24576000, 96000, 1, 2}, + {36864000, 96000, 1, 3}, + {49152000, 96000, 1, 4}, + + /* 128k */ + {8192000, 128000, 2, 0}, + {12288000, 128000, 2, 1}, + {16934400, 128000, 2, 2}, + {24576000, 128000, 2, 3}, + {32768000, 128000, 2, 4}, + + /* 176.4k */ + {11289600, 176400, 2, 0}, + {16934400, 176400, 2, 1}, + {22579200, 176400, 2, 2}, + {33868000, 176400, 2, 3}, + {49152000, 176400, 2, 4}, + + /* 192k */ + {12288000, 192000, 2, 0}, + {18432000, 192000, 2, 1}, + {24576000, 192000, 2, 2}, + {36864000, 192000, 2, 3}, + {49152000, 192000, 2, 4}, +}; + +static int cs4265_get_clk_index(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { + if (clk_map_table[i].rate == rate && + clk_map_table[i].mclk == mclk) + return i; + } + return -EINVAL; +} + +static int cs4265_set_sysclk(struct snd_soc_dai *codec_dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec); + int i; + + if (clk_id != 0) { + dev_err(codec->dev, "Invalid clk_id %d\n", clk_id); + return -EINVAL; + } + for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { + if (clk_map_table[i].mclk == freq) { + cs4265->sysclk = freq; + return 0; + } + } + cs4265->sysclk = 0; + dev_err(codec->dev, "Invalid freq parameter %d\n", freq); + return -EINVAL; +} + +static int cs4265_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec); + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_ADC_MASTER, + CS4265_ADC_MASTER); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_ADC_MASTER, + 0); + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= SND_SOC_DAIFMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface |= SND_SOC_DAIFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= SND_SOC_DAIFMT_LEFT_J; + break; + default: + return -EINVAL; + } + + cs4265->format = iface; + return 0; +} + +static int cs4265_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) { + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_MUTE, + CS4265_DAC_CTL_MUTE); + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, + CS4265_SPDIF_CTL2_MUTE, + CS4265_SPDIF_CTL2_MUTE); + } else { + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_MUTE, + 0); + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, + CS4265_SPDIF_CTL2_MUTE, + 0); + } + return 0; +} + +static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec); + int index; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + ((cs4265->format & SND_SOC_DAIFMT_FORMAT_MASK) + == SND_SOC_DAIFMT_RIGHT_J)) + return -EINVAL; + + index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params)); + if (index >= 0) { + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_ADC_FM, clk_map_table[index].fm_mode); + snd_soc_update_bits(codec, CS4265_MCLK_FREQ, + CS4265_MCLK_FREQ_MASK, + clk_map_table[index].mclkdiv); + + } else { + dev_err(codec->dev, "can't get correct mclk\n"); + return -EINVAL; + } + + switch (cs4265->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_DIF, (1 << 4)); + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_ADC_DIF, (1 << 4)); + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, + CS4265_SPDIF_CTL2_DIF, (1 << 6)); + break; + case SND_SOC_DAIFMT_RIGHT_J: + if (params_width(params) == 16) { + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_DIF, (1 << 5)); + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_SPDIF_CTL2_DIF, (1 << 7)); + } else { + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_DIF, (3 << 5)); + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_SPDIF_CTL2_DIF, (1 << 7)); + } + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_DIF, 0); + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_ADC_DIF, 0); + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_SPDIF_CTL2_DIF, (1 << 6)); + + break; + default: + return -EINVAL; + } + return 0; +} + +static int cs4265_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, CS4265_PWRCTL, + CS4265_PWRCTL_PDN, 0); + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, CS4265_PWRCTL, + CS4265_PWRCTL_PDN, + CS4265_PWRCTL_PDN); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, CS4265_PWRCTL, + CS4265_PWRCTL_PDN, + CS4265_PWRCTL_PDN); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define CS4265_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) + +#define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE) + +static const struct snd_soc_dai_ops cs4265_ops = { + .hw_params = cs4265_pcm_hw_params, + .digital_mute = cs4265_digital_mute, + .set_fmt = cs4265_set_fmt, + .set_sysclk = cs4265_set_sysclk, +}; + +static struct snd_soc_dai_driver cs4265_dai[] = { + { + .name = "cs4265-dai1", + .playback = { + .stream_name = "DAI1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS4265_RATES, + .formats = CS4265_FORMATS, + }, + .capture = { + .stream_name = "DAI1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS4265_RATES, + .formats = CS4265_FORMATS, + }, + .ops = &cs4265_ops, + }, + { + .name = "cs4265-dai2", + .playback = { + .stream_name = "DAI2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS4265_RATES, + .formats = CS4265_FORMATS, + }, + .capture = { + .stream_name = "DAI2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS4265_RATES, + .formats = CS4265_FORMATS, + }, + .ops = &cs4265_ops, + }, +}; + +static const struct snd_soc_codec_driver soc_codec_cs4265 = { + .set_bias_level = cs4265_set_bias_level, + + .dapm_widgets = cs4265_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4265_dapm_widgets), + .dapm_routes = cs4265_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs4265_audio_map), + + .controls = cs4265_snd_controls, + .num_controls = ARRAY_SIZE(cs4265_snd_controls), +}; + +static const struct regmap_config cs4265_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS4265_MAX_REGISTER, + .reg_defaults = cs4265_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4265_reg_defaults), + .readable_reg = cs4265_readable_register, + .volatile_reg = cs4265_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs4265_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs4265_private *cs4265; + int ret = 0; + unsigned int devid = 0; + unsigned int reg; + + cs4265 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4265_private), + GFP_KERNEL); + if (cs4265 == NULL) + return -ENOMEM; + cs4265->dev = &i2c_client->dev; + + cs4265->regmap = devm_regmap_init_i2c(i2c_client, &cs4265_regmap); + if (IS_ERR(cs4265->regmap)) { + ret = PTR_ERR(cs4265->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev, + "reset-gpios"); + if (IS_ERR(cs4265->reset_gpio)) { + ret = PTR_ERR(cs4265->reset_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + cs4265->reset_gpio = NULL; + } else { + ret = gpiod_direction_output(cs4265->reset_gpio, 0); + if (ret) + return ret; + mdelay(1); + gpiod_set_value_cansleep(cs4265->reset_gpio, 1); + + } + + i2c_set_clientdata(i2c_client, cs4265); + + ret = regmap_read(cs4265->regmap, CS4265_CHIP_ID, ®); + devid = reg & CS4265_CHIP_ID_MASK; + if (devid != CS4265_CHIP_ID_VAL) { + ret = -ENODEV; + dev_err(&i2c_client->dev, + "CS4265 Device ID (%X). Expected %X\n", + devid, CS4265_CHIP_ID); + return ret; + } + dev_info(&i2c_client->dev, + "CS4265 Version %x\n", + reg & CS4265_REV_ID_MASK); + + regmap_write(cs4265->regmap, CS4265_PWRCTL, 0x0F); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_cs4265, cs4265_dai, + ARRAY_SIZE(cs4265_dai)); + return ret; +} + +static int cs4265_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct of_device_id cs4265_of_match[] = { + { .compatible = "cirrus,cs4265", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs4265_of_match); + +static const struct i2c_device_id cs4265_id[] = { + { "cs4265", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, cs4265_id); + +static struct i2c_driver cs4265_i2c_driver = { + .driver = { + .name = "cs4265", + .owner = THIS_MODULE, + .of_match_table = cs4265_of_match, + }, + .id_table = cs4265_id, + .probe = cs4265_i2c_probe, + .remove = cs4265_i2c_remove, +}; + +module_i2c_driver(cs4265_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS4265 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <paul.handrigan@cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4265.h b/sound/soc/codecs/cs4265.h new file mode 100644 index 000000000000..0a80a8dcec67 --- /dev/null +++ b/sound/soc/codecs/cs4265.h @@ -0,0 +1,64 @@ +/* + * cs4265.h -- CS4265 ALSA SoC audio driver + * + * Copyright 2014 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan@cirrus.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 __CS4265_H__ +#define __CS4265_H__ + +#define CS4265_CHIP_ID 0x1 +#define CS4265_CHIP_ID_VAL 0xD0 +#define CS4265_CHIP_ID_MASK 0xF0 +#define CS4265_REV_ID_MASK 0x0F + +#define CS4265_PWRCTL 0x02 +#define CS4265_PWRCTL_PDN 1 + +#define CS4265_DAC_CTL 0x3 +#define CS4265_DAC_CTL_MUTE (1 << 2) +#define CS4265_DAC_CTL_DIF (3 << 4) + +#define CS4265_ADC_CTL 0x4 +#define CS4265_ADC_MASTER 1 +#define CS4265_ADC_DIF (1 << 4) +#define CS4265_ADC_FM (3 << 6) + +#define CS4265_MCLK_FREQ 0x5 +#define CS4265_MCLK_FREQ_MASK (7 << 4) + +#define CS4265_SIG_SEL 0x6 +#define CS4265_SIG_SEL_LOOP (1 << 1) + +#define CS4265_CHB_PGA_CTL 0x7 +#define CS4265_CHA_PGA_CTL 0x8 + +#define CS4265_ADC_CTL2 0x9 + +#define CS4265_DAC_CHA_VOL 0xA +#define CS4265_DAC_CHB_VOL 0xB + +#define CS4265_DAC_CTL2 0xC + +#define CS4265_INT_STATUS 0xD +#define CS4265_INT_MASK 0xE +#define CS4265_STATUS_MODE_MSB 0xF +#define CS4265_STATUS_MODE_LSB 0x10 + +#define CS4265_SPDIF_CTL1 0x11 + +#define CS4265_SPDIF_CTL2 0x12 +#define CS4265_SPDIF_CTL2_MUTE (1 << 4) +#define CS4265_SPDIF_CTL2_DIF (3 << 6) + +#define CS4265_C_DATA_BUFF 0x13 +#define CS4265_MAX_REGISTER 0x2A + +#endif diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 9947a9583679..e6d4ff9fd992 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -664,10 +664,8 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, cs4270 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4270_private), GFP_KERNEL); - if (!cs4270) { - dev_err(&i2c_client->dev, "could not allocate codec\n"); + if (!cs4270) return -ENOMEM; - } /* get the power supply regulators */ for (i = 0; i < ARRAY_SIZE(supply_names); i++) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 071fc77f2f06..969167d8b71e 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -399,15 +399,15 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL, - CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv), + CS42L52_HPB_VOL, 0, 0x34, 0xC0, hpd_tlv), SOC_ENUM("Headphone Analog Gain", hp_gain_enum), SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, - CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv), + CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), @@ -417,10 +417,10 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("MIC Bias Level", mic_bias_level_enum), SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL, - CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv), + CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 6, 0x7f, 0x19, ipd_tlv), + 0, 0x19, 0x7F, ipd_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), @@ -428,11 +428,11 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_MIXER_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL, - CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv), + CS42L52_PGAB_CTL, 0, 0x28, 0x24, pga_tlv), SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, - 0, 0x7f, 0x19, mix_tlv), + 0, 0x19, 0x7f, mix_tlv), SOC_DOUBLE_R("PCM Mixer Switch", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1), diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index fdc4bd27b0df..c766a5a9ce80 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -318,24 +318,32 @@ static const struct soc_enum adca_swap_enum = ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); +static const struct snd_kcontrol_new adca_swap_mux = + SOC_DAPM_ENUM("Route", adca_swap_enum); static const struct soc_enum pcma_swap_enum = SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 4, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); +static const struct snd_kcontrol_new pcma_swap_mux = + SOC_DAPM_ENUM("Route", pcma_swap_enum); static const struct soc_enum adcb_swap_enum = SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 2, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); +static const struct snd_kcontrol_new adcb_swap_mux = + SOC_DAPM_ENUM("Route", adcb_swap_enum); static const struct soc_enum pcmb_swap_enum = SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 6, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); +static const struct snd_kcontrol_new pcmb_swap_mux = + SOC_DAPM_ENUM("Route", pcmb_swap_enum); static const struct snd_kcontrol_new hpa_switch = SOC_DAPM_SINGLE("Switch", CS42L56_PWRCTL_2, 6, 1, 1); @@ -421,15 +429,15 @@ static const struct soc_enum ng_delay_enum = static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L56_MASTER_A_VOLUME, - CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xfd, adv_tlv), + CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xE4, adv_tlv), SOC_DOUBLE("Master Mute Switch", CS42L56_DSP_MUTE_CTL, 0, 1, 1, 1), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L56_ADCA_MIX_VOLUME, - CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv), + CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv), SOC_DOUBLE("ADC Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 6, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L56_PCMA_MIX_VOLUME, - CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv), + CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv), SOC_DOUBLE("PCM Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 4, 5, 1, 1), SOC_SINGLE_TLV("Analog Advisory Volume", @@ -438,16 +446,16 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { CS42L56_DIGINPUT_ADV_VOLUME, 0, 0x00, 1, adv_tlv), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L56_PGAA_MUX_VOLUME, - CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0xfd, pga_tlv), + CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0x24, pga_tlv), SOC_DOUBLE_R_TLV("ADC Volume", CS42L56_ADCA_ATTENUATOR, CS42L56_ADCB_ATTENUATOR, 0, 0x00, 1, adc_tlv), SOC_DOUBLE("ADC Mute Switch", CS42L56_MISC_ADC_CTL, 2, 3, 1, 1), SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPA_VOLUME, 0, 0x44, 0x55, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOA_VOLUME, 0, 0x44, 0x55, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), @@ -467,11 +475,6 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_SINGLE("ADCA Invert", CS42L56_MISC_ADC_CTL, 2, 1, 1), SOC_SINGLE("ADCB Invert", CS42L56_MISC_ADC_CTL, 3, 1, 1), - SOC_ENUM("PCMA Swap", pcma_swap_enum), - SOC_ENUM("PCMB Swap", pcmb_swap_enum), - SOC_ENUM("ADCA Swap", adca_swap_enum), - SOC_ENUM("ADCB Swap", adcb_swap_enum), - SOC_DOUBLE("HPF Switch", CS42L56_HPF_CTL, 5, 7, 1, 1), SOC_DOUBLE("HPF Freeze Switch", CS42L56_HPF_CTL, 4, 6, 1, 1), SOC_ENUM("HPFA Corner Freq", hpfa_freq_enum), @@ -570,6 +573,16 @@ static const struct snd_soc_dapm_widget cs42l56_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADCA", NULL, CS42L56_PWRCTL_1, 1, 1), SND_SOC_DAPM_ADC("ADCB", NULL, CS42L56_PWRCTL_1, 2, 1), + SND_SOC_DAPM_MUX("ADCA Swap Mux", SND_SOC_NOPM, 0, 0, + &adca_swap_mux), + SND_SOC_DAPM_MUX("ADCB Swap Mux", SND_SOC_NOPM, 0, 0, + &adcb_swap_mux), + + SND_SOC_DAPM_MUX("PCMA Swap Mux", SND_SOC_NOPM, 0, 0, + &pcma_swap_mux), + SND_SOC_DAPM_MUX("PCMB Swap Mux", SND_SOC_NOPM, 0, 0, + &pcmb_swap_mux), + SND_SOC_DAPM_DAC("DACA", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DACB", NULL, SND_SOC_NOPM, 0, 0), @@ -607,8 +620,19 @@ static const struct snd_soc_dapm_route cs42l56_audio_map[] = { {"Digital Output Mux", NULL, "ADCA"}, {"Digital Output Mux", NULL, "ADCB"}, - {"ADCB", NULL, "ADCB Mux"}, - {"ADCA", NULL, "ADCA Mux"}, + {"ADCB", NULL, "ADCB Swap Mux"}, + {"ADCA", NULL, "ADCA Swap Mux"}, + + {"ADCA Swap Mux", NULL, "ADCA"}, + {"ADCB Swap Mux", NULL, "ADCB"}, + + {"DACA", "Left", "ADCA Swap Mux"}, + {"DACA", "LR 2", "ADCA Swap Mux"}, + {"DACA", "Right", "ADCA Swap Mux"}, + + {"DACB", "Left", "ADCB Swap Mux"}, + {"DACB", "LR 2", "ADCB Swap Mux"}, + {"DACB", "Right", "ADCB Swap Mux"}, {"ADCA Mux", NULL, "AIN3A"}, {"ADCA Mux", NULL, "AIN2A"}, @@ -633,30 +657,32 @@ static const struct snd_soc_dapm_route cs42l56_audio_map[] = { {"PGAB Input Mux", NULL, "AIN2B"}, {"PGAB Input Mux", NULL, "AIN3B"}, - {"LOB", NULL, "Lineout Right"}, - {"LOA", NULL, "Lineout Left"}, - - {"Lineout Right", "Switch", "LINEOUTB Input Mux"}, - {"Lineout Left", "Switch", "LINEOUTA Input Mux"}, + {"LOB", "Switch", "LINEOUTB Input Mux"}, + {"LOA", "Switch", "LINEOUTA Input Mux"}, {"LINEOUTA Input Mux", "PGAA", "PGAA"}, {"LINEOUTB Input Mux", "PGAB", "PGAB"}, {"LINEOUTA Input Mux", "DACA", "DACA"}, {"LINEOUTB Input Mux", "DACB", "DACB"}, - {"HPA", NULL, "Headphone Left"}, - {"HPB", NULL, "Headphone Right"}, - - {"Headphone Right", "Switch", "HPB Input Mux"}, - {"Headphone Left", "Switch", "HPA Input Mux"}, + {"HPA", "Switch", "HPB Input Mux"}, + {"HPB", "Switch", "HPA Input Mux"}, {"HPA Input Mux", "PGAA", "PGAA"}, {"HPB Input Mux", "PGAB", "PGAB"}, {"HPA Input Mux", "DACA", "DACA"}, {"HPB Input Mux", "DACB", "DACB"}, - {"DACB", NULL, "HiFi Playback"}, - {"DACA", NULL, "HiFi Playback"}, + {"DACA", NULL, "PCMA Swap Mux"}, + {"DACB", NULL, "PCMB Swap Mux"}, + + {"PCMB Swap Mux", "Left", "HiFi Playback"}, + {"PCMB Swap Mux", "LR 2", "HiFi Playback"}, + {"PCMB Swap Mux", "Right", "HiFi Playback"}, + + {"PCMA Swap Mux", "Left", "HiFi Playback"}, + {"PCMA Swap Mux", "LR 2", "HiFi Playback"}, + {"PCMA Swap Mux", "Right", "HiFi Playback"}, }; diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index ae3717992d56..0e7b9eb2ba61 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv), SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, - CS42L73_MICBPREPGABVOL, 5, 0x34, + CS42L73_MICBPREPGABVOL, 0, 0x34, 0x24, micpga_tlv), SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, @@ -1408,10 +1408,8 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, cs42l73 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l73_private), GFP_KERNEL); - if (!cs42l73) { - dev_err(&i2c_client->dev, "could not allocate codec\n"); + if (!cs42l73) return -ENOMEM; - } cs42l73->regmap = devm_regmap_init_i2c(i2c_client, &cs42l73_regmap); if (IS_ERR(cs42l73->regmap)) { diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index a25bc6061a30..02b1520ae0bc 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -219,6 +219,9 @@ static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_RIGHT_J: val = CS42XX8_INTF_DAC_DIF_RIGHTJ | CS42XX8_INTF_ADC_DIF_RIGHTJ; break; + case SND_SOC_DAIFMT_DSP_A: + val = CS42XX8_INTF_DAC_DIF_TDM | CS42XX8_INTF_ADC_DIF_TDM; + break; default: dev_err(codec->dev, "unsupported dai format\n"); return -EINVAL; @@ -422,7 +425,7 @@ const struct cs42xx8_driver_data cs42888_data = { }; EXPORT_SYMBOL_GPL(cs42888_data); -const struct of_device_id cs42xx8_of_match[] = { +static const struct of_device_id cs42xx8_of_match[] = { { .compatible = "cirrus,cs42448", .data = &cs42448_data, }, { .compatible = "cirrus,cs42888", .data = &cs42888_data, }, { /* sentinel */ } diff --git a/sound/soc/codecs/cs42xx8.h b/sound/soc/codecs/cs42xx8.h index da0b94aee419..b2c10e537ef6 100644 --- a/sound/soc/codecs/cs42xx8.h +++ b/sound/soc/codecs/cs42xx8.h @@ -128,8 +128,8 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap); #define CS42XX8_INTF_DAC_DIF_RIGHTJ (2 << CS42XX8_INTF_DAC_DIF_SHIFT) #define CS42XX8_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_DAC_DIF_SHIFT) #define CS42XX8_INTF_DAC_DIF_ONELINE_20 (4 << CS42XX8_INTF_DAC_DIF_SHIFT) -#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (6 << CS42XX8_INTF_DAC_DIF_SHIFT) -#define CS42XX8_INTF_DAC_DIF_TDM (7 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (5 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_TDM (6 << CS42XX8_INTF_DAC_DIF_SHIFT) #define CS42XX8_INTF_ADC_DIF_SHIFT 0 #define CS42XX8_INTF_ADC_DIF_WIDTH 3 #define CS42XX8_INTF_ADC_DIF_MASK (((1 << CS42XX8_INTF_ADC_DIF_WIDTH) - 1) << CS42XX8_INTF_ADC_DIF_SHIFT) @@ -138,8 +138,8 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap); #define CS42XX8_INTF_ADC_DIF_RIGHTJ (2 << CS42XX8_INTF_ADC_DIF_SHIFT) #define CS42XX8_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_ADC_DIF_SHIFT) #define CS42XX8_INTF_ADC_DIF_ONELINE_20 (4 << CS42XX8_INTF_ADC_DIF_SHIFT) -#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (6 << CS42XX8_INTF_ADC_DIF_SHIFT) -#define CS42XX8_INTF_ADC_DIF_TDM (7 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (5 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_TDM (6 << CS42XX8_INTF_ADC_DIF_SHIFT) /* ADC Control & DAC De-Emphasis (Address 05h) */ #define CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index d5fd00a64748..8f95b0300f1a 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -253,7 +253,7 @@ static void v253_close(struct tty_struct *tty) /* Prevent the codec driver from further accessing the modem */ codec->hw_write = NULL; cx20442->control_data = NULL; - codec->card->pop_time = 0; + codec->component.card->pop_time = 0; } /* Line discipline .hangup() */ @@ -281,7 +281,7 @@ static void v253_receive(struct tty_struct *tty, /* Set up codec driver access to modem controls */ cx20442->control_data = tty; codec->hw_write = (hw_write_t)tty->ops->write; - codec->card->pop_time = 1; + codec->component.card->pop_time = 1; } } @@ -372,7 +372,7 @@ static int cx20442_codec_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, cx20442); codec->hw_write = NULL; - codec->card->pop_time = 0; + codec->component.card->pop_time = 0; return 0; } @@ -383,8 +383,8 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec) struct cx20442_priv *cx20442 = snd_soc_codec_get_drvdata(codec); if (cx20442->control_data) { - struct tty_struct *tty = cx20442->control_data; - tty_hangup(tty); + struct tty_struct *tty = cx20442->control_data; + tty_hangup(tty); } if (!IS_ERR(cx20442->por)) { diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 9134982807b5..2cd3e5427441 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1299,12 +1299,12 @@ static int max98088_dai2_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: snd_soc_update_bits(codec, M98088_REG_1C_DAI2_FORMAT, M98088_DAI_WS, 0); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: snd_soc_update_bits(codec, M98088_REG_1C_DAI2_FORMAT, M98088_DAI_WS, M98088_DAI_WS); break; diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f5fccc7a8e89..4a063fa88526 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -26,10 +26,6 @@ #include <sound/max98090.h> #include "max98090.h" -#define DEBUG -#define EXTMIC_METHOD -#define EXTMIC_METHOD_TEST - /* Allows for sparsely populated register maps */ static struct reg_default max98090_reg[] = { { 0x00, 0x00 }, /* 00 Software Reset */ @@ -820,7 +816,6 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w, else val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT; - if (val >= 1) { if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) { max98090->pa1en = val - 1; /* Update for volatile */ @@ -1140,7 +1135,6 @@ static const struct snd_kcontrol_new max98090_mixhprsel_mux = SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_INPUT("DMICL"), @@ -1304,7 +1298,6 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { }; static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("DMIC3"), SND_SOC_DAPM_INPUT("DMIC4"), @@ -1315,7 +1308,6 @@ static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { }; static const struct snd_soc_dapm_route max98090_dapm_routes[] = { - {"MIC1 Input", NULL, "MIC1"}, {"MIC2 Input", NULL, "MIC2"}, @@ -1493,17 +1485,14 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"SPKR", NULL, "SPK Right Out"}, {"RCVL", NULL, "RCV Left Out"}, {"RCVR", NULL, "RCV Right Out"}, - }; static const struct snd_soc_dapm_route max98091_dapm_routes[] = { - /* DMIC inputs */ {"DMIC3", NULL, "DMIC3_ENA"}, {"DMIC4", NULL, "DMIC4_ENA"}, {"DMIC3", NULL, "AHPF"}, {"DMIC4", NULL, "AHPF"}, - }; static int max98090_add_widgets(struct snd_soc_codec *codec) @@ -1531,7 +1520,6 @@ static int max98090_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, max98091_dapm_routes, ARRAY_SIZE(max98091_dapm_routes)); - } return 0; @@ -2212,22 +2200,11 @@ static struct snd_soc_dai_driver max98090_dai[] = { } }; -static void max98090_handle_pdata(struct snd_soc_codec *codec) -{ - struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); - struct max98090_pdata *pdata = max98090->pdata; - - if (!pdata) { - dev_err(codec->dev, "No platform data\n"); - return; - } - -} - static int max98090_probe(struct snd_soc_codec *codec) { struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); struct max98090_cdata *cdata; + enum max98090_type devtype; int ret = 0; dev_dbg(codec->dev, "max98090_probe\n"); @@ -2263,16 +2240,21 @@ static int max98090_probe(struct snd_soc_codec *codec) } if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) { - max98090->devtype = MAX98090; + devtype = MAX98090; dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret); } else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) { - max98090->devtype = MAX98091; + devtype = MAX98091; dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret); } else { - max98090->devtype = MAX98090; + devtype = MAX98090; dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret); } + if (max98090->devtype != devtype) { + dev_warn(codec->dev, "Mismatch in DT specified CODEC type.\n"); + max98090->devtype = devtype; + } + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work); @@ -2284,7 +2266,7 @@ static int max98090_probe(struct snd_soc_codec *codec) /* Register for interrupts */ dev_dbg(codec->dev, "irq = %d\n", max98090->irq); - ret = request_threaded_irq(max98090->irq, NULL, + ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL, max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "max98090_interrupt", codec); if (ret < 0) { @@ -2317,8 +2299,6 @@ static int max98090_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE, M98090_MBVSEL_MASK, M98090_MBVSEL_2V8); - max98090_handle_pdata(codec); - max98090_add_widgets(codec); err_access: @@ -2428,7 +2408,7 @@ static int max98090_runtime_suspend(struct device *dev) } #endif -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int max98090_resume(struct device *dev) { struct max98090_priv *max98090 = dev_get_drvdata(dev); @@ -2460,12 +2440,14 @@ static const struct dev_pm_ops max98090_pm = { static const struct i2c_device_id max98090_i2c_id[] = { { "max98090", MAX98090 }, + { "max98091", MAX98091 }, { } }; MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); static const struct of_device_id max98090_of_match[] = { { .compatible = "maxim,max98090", }, + { .compatible = "maxim,max98091", }, { } }; MODULE_DEVICE_TABLE(of, max98090_of_match); diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 89ec00424880..0ee6797d5083 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1280,12 +1280,12 @@ static int max98095_dai2_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: snd_soc_update_bits(codec, M98095_034_DAI2_FORMAT, M98095_DAI_WS, 0); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: snd_soc_update_bits(codec, M98095_034_DAI2_FORMAT, M98095_DAI_WS, M98095_DAI_WS); break; @@ -1341,12 +1341,12 @@ static int max98095_dai3_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: snd_soc_update_bits(codec, M98095_03E_DAI3_FORMAT, M98095_DAI_WS, 0); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: snd_soc_update_bits(codec, M98095_03E_DAI3_FORMAT, M98095_DAI_WS, M98095_DAI_WS); break; diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 9965277b595a..388f90a597fa 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -766,11 +766,11 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port); if (ret) - return ret; + goto out; ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port); if (ret) - return ret; + goto out; } dev_set_drvdata(&pdev->dev, priv); @@ -783,6 +783,8 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); +out: + of_node_put(np); return ret; } diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 3a80ba4452df..57b0c94a710b 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -36,6 +36,7 @@ #define PCM1792A_DAC_VOL_LEFT 0x10 #define PCM1792A_DAC_VOL_RIGHT 0x11 #define PCM1792A_FMT_CONTROL 0x12 +#define PCM1792A_MODE_CONTROL 0x13 #define PCM1792A_SOFT_MUTE PCM1792A_FMT_CONTROL #define PCM1792A_FMT_MASK 0x70 @@ -164,6 +165,8 @@ static const struct snd_kcontrol_new pcm1792a_controls[] = { SOC_DOUBLE_R_RANGE_TLV("DAC Playback Volume", PCM1792A_DAC_VOL_LEFT, PCM1792A_DAC_VOL_RIGHT, 0, 0xf, 0xff, 0, pcm1792a_dac_tlv), + SOC_SINGLE("DAC Invert Output Switch", PCM1792A_MODE_CONTROL, 7, 1, 0), + SOC_SINGLE("DAC Rolloff Filter Switch", PCM1792A_MODE_CONTROL, 1, 1, 0), }; static const struct snd_soc_dapm_widget pcm1792a_dapm_widgets[] = { diff --git a/sound/soc/codecs/pcm1792a.h b/sound/soc/codecs/pcm1792a.h index 7a83d1fc102a..51d5470fee16 100644 --- a/sound/soc/codecs/pcm1792a.h +++ b/sound/soc/codecs/pcm1792a.h @@ -18,7 +18,8 @@ #define __PCM1792A_H__ #define PCM1792A_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_8000_48000 | \ - SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000) #define PCM1792A_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S16_LE) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 7b82fbe0d14c..56650d6c2f53 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -11,25 +11,6 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/init.h> -#include <linux/delay.h> -#include <linux/pm.h> -#include <linux/gpio.h> -#include <linux/i2c.h> -#include <linux/regmap.h> -#include <linux/of.h> -#include <linux/of_gpio.h> -#include <linux/platform_device.h> -#include <linux/spi/spi.h> -#include <linux/acpi.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <sound/initval.h> -#include <sound/tlv.h> #include "rl6231.h" diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 000000000000..e4f6102efc1a --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1222 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 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 <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <linux/workqueue.h> +#include <sound/rt286.h> +#include <sound/hda_verbs.h> + +#include "rt286.h" + +#define RT286_VENDOR_ID 0x10ec0286 + +struct rt286_priv { + struct regmap *regmap; + struct rt286_platform_data pdata; + struct i2c_client *i2c; + struct snd_soc_jack *jack; + struct delayed_work jack_detect_work; + int sys_clk; + struct reg_default *index_cache; +}; + +static struct reg_default rt286_index_def[] = { + { 0x01, 0xaaaa }, + { 0x02, 0x8aaa }, + { 0x03, 0x0002 }, + { 0x04, 0xaf01 }, + { 0x08, 0x000d }, + { 0x09, 0xd810 }, + { 0x0a, 0x0060 }, + { 0x0b, 0x0000 }, + { 0x0d, 0x2800 }, + { 0x0f, 0x0000 }, + { 0x19, 0x0a17 }, + { 0x20, 0x0020 }, + { 0x33, 0x0208 }, + { 0x49, 0x0004 }, + { 0x4f, 0x50e9 }, + { 0x50, 0x2c00 }, + { 0x63, 0x2902 }, + { 0x67, 0x1111 }, + { 0x68, 0x1016 }, + { 0x69, 0x273f }, +}; +#define INDEX_CACHE_SIZE ARRAY_SIZE(rt286_index_def) + +static const struct reg_default rt286_reg[] = { + { 0x00170500, 0x00000400 }, + { 0x00220000, 0x00000031 }, + { 0x00239000, 0x0000007f }, + { 0x0023a000, 0x0000007f }, + { 0x00270500, 0x00000400 }, + { 0x00370500, 0x00000400 }, + { 0x00870500, 0x00000400 }, + { 0x00920000, 0x00000031 }, + { 0x00935000, 0x000000c3 }, + { 0x00936000, 0x000000c3 }, + { 0x00970500, 0x00000400 }, + { 0x00b37000, 0x00000097 }, + { 0x00b37200, 0x00000097 }, + { 0x00b37300, 0x00000097 }, + { 0x00c37000, 0x00000000 }, + { 0x00c37100, 0x00000080 }, + { 0x01270500, 0x00000400 }, + { 0x01370500, 0x00000400 }, + { 0x01371f00, 0x411111f0 }, + { 0x01439000, 0x00000080 }, + { 0x0143a000, 0x00000080 }, + { 0x01470700, 0x00000000 }, + { 0x01470500, 0x00000400 }, + { 0x01470c00, 0x00000000 }, + { 0x01470100, 0x00000000 }, + { 0x01837000, 0x00000000 }, + { 0x01870500, 0x00000400 }, + { 0x02050000, 0x00000000 }, + { 0x02139000, 0x00000080 }, + { 0x0213a000, 0x00000080 }, + { 0x02170100, 0x00000000 }, + { 0x02170500, 0x00000400 }, + { 0x02170700, 0x00000000 }, + { 0x02270100, 0x00000000 }, + { 0x02370100, 0x00000000 }, + { 0x02040000, 0x00004002 }, + { 0x01870700, 0x00000020 }, + { 0x00830000, 0x000000c3 }, + { 0x00930000, 0x000000c3 }, + { 0x01270700, 0x00000000 }, +}; + +static bool rt286_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT286_GET_HP_SENSE: + case RT286_GET_MIC1_SENSE: + case RT286_PROC_COEF: + return true; + default: + return false; + } + + +} + +static bool rt286_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT286_GET_HP_SENSE: + case RT286_GET_MIC1_SENSE: + case RT286_SET_AUDIO_POWER: + case RT286_SET_HPO_POWER: + case RT286_SET_SPK_POWER: + case RT286_SET_DMIC1_POWER: + case RT286_SPK_MUX: + case RT286_HPO_MUX: + case RT286_ADC0_MUX: + case RT286_ADC1_MUX: + case RT286_SET_MIC1: + case RT286_SET_PIN_HPO: + case RT286_SET_PIN_SPK: + case RT286_SET_PIN_DMIC1: + case RT286_SPK_EAPD: + case RT286_SET_AMP_GAIN_HPO: + case RT286_SET_DMIC2_DEFAULT: + case RT286_DACL_GAIN: + case RT286_DACR_GAIN: + case RT286_ADCL_GAIN: + case RT286_ADCR_GAIN: + case RT286_MIC_GAIN: + case RT286_SPOL_GAIN: + case RT286_SPOR_GAIN: + case RT286_HPOL_GAIN: + case RT286_HPOR_GAIN: + case RT286_F_DAC_SWITCH: + case RT286_F_RECMIX_SWITCH: + case RT286_REC_MIC_SWITCH: + case RT286_REC_I2S_SWITCH: + case RT286_REC_LINE_SWITCH: + case RT286_REC_BEEP_SWITCH: + case RT286_DAC_FORMAT: + case RT286_ADC_FORMAT: + case RT286_COEF_INDEX: + case RT286_PROC_COEF: + case RT286_SET_AMP_GAIN_ADC_IN1: + case RT286_SET_AMP_GAIN_ADC_IN2: + case RT286_SET_POWER(RT286_DAC_OUT1): + case RT286_SET_POWER(RT286_DAC_OUT2): + case RT286_SET_POWER(RT286_ADC_IN1): + case RT286_SET_POWER(RT286_ADC_IN2): + case RT286_SET_POWER(RT286_DMIC2): + case RT286_SET_POWER(RT286_MIC1): + return true; + default: + return false; + } +} + +static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) +{ + struct i2c_client *client = context; + struct rt286_priv *rt286 = i2c_get_clientdata(client); + u8 data[4]; + int ret, i; + + /*handle index registers*/ + if (reg <= 0xff) { + rt286_hw_write(client, RT286_COEF_INDEX, reg); + reg = RT286_PROC_COEF; + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + if (reg == rt286->index_cache[i].reg) { + rt286->index_cache[i].def = value; + break; + } + + } + } + + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + /* + * 4 bit VID: reg should be 0 + * 12 bit VID: value should be 0 + * So we use an OR operator to handle it rather than use if condition. + */ + data[2] = ((reg >> 8) & 0xff) | ((value >> 8) & 0xff); + data[3] = value & 0xff; + + ret = i2c_master_send(client, data, 4); + + if (ret == 4) + return 0; + else + pr_err("ret=%d\n", ret); + if (ret < 0) + return ret; + else + return -EIO; +} + +static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + int ret; + __be32 be_reg; + unsigned int index, vid, buf = 0x0; + + /*handle index registers*/ + if (reg <= 0xff) { + rt286_hw_write(client, RT286_COEF_INDEX, reg); + reg = RT286_PROC_COEF; + } + + reg = reg | 0x80000; + vid = (reg >> 8) & 0xfff; + + if (AC_VERB_GET_AMP_GAIN_MUTE == (vid & 0xf00)) { + index = (reg >> 8) & 0xf; + reg = (reg & ~0xf0f) | index; + } + be_reg = cpu_to_be32(reg); + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)&be_reg; + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + *value = be32_to_cpu(buf); + + return 0; +} + +static void rt286_index_sync(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + snd_soc_write(codec, rt286->index_cache[i].reg, + rt286->index_cache[i].def); + } +} + +static int rt286_support_power_controls[] = { + RT286_DAC_OUT1, + RT286_DAC_OUT2, + RT286_ADC_IN1, + RT286_ADC_IN2, + RT286_MIC1, + RT286_DMIC1, + RT286_DMIC2, + RT286_SPK_OUT, + RT286_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) +{ + unsigned int val, buf; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); + *hp = buf & 0x80000000; + if (*hp) { + /* power on HV,VERF */ + regmap_update_bits(rt286->regmap, + RT286_POWER_CTRL1, 0x1001, 0x0); + /* power LDO1 */ + regmap_update_bits(rt286->regmap, + RT286_POWER_CTRL2, 0x4, 0x4); + regmap_write(rt286->regmap, RT286_SET_MIC1, 0x24); + regmap_read(rt286->regmap, RT286_CBJ_CTRL2, &val); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + regmap_read(rt286->regmap, + RT286_CBJ_CTRL2, &val); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + regmap_write(rt286->regmap, + RT286_SET_MIC1, 0x20); + /* power off HV,VERF */ + regmap_update_bits(rt286->regmap, + RT286_POWER_CTRL1, 0x1001, 0x1001); + regmap_update_bits(rt286->regmap, + RT286_A_BIAS_CTRL3, 0xc000, 0x0000); + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0x0030, 0x0000); + regmap_update_bits(rt286->regmap, + RT286_A_BIAS_CTRL2, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + regmap_update_bits(rt286->regmap, + RT286_A_BIAS_CTRL3, 0xc000, 0x8000); + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0x0030, 0x0020); + regmap_update_bits(rt286->regmap, + RT286_A_BIAS_CTRL2, 0xc000, 0x8000); + } else { + *mic = false; + } + + regmap_update_bits(rt286->regmap, + RT286_MISC_CTRL1, + 0x0060, 0x0000); + } else { + regmap_update_bits(rt286->regmap, + RT286_MISC_CTRL1, + 0x0060, 0x0020); + regmap_update_bits(rt286->regmap, + RT286_A_BIAS_CTRL3, + 0xc000, 0x8000); + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, + 0x0030, 0x0020); + regmap_update_bits(rt286->regmap, + RT286_A_BIAS_CTRL2, + 0xc000, 0x8000); + + *mic = false; + } + } else { + regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); + *hp = buf & 0x80000000; + regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf); + *mic = buf & 0x80000000; + } + + return 0; +} + +static void rt286_jack_detect_work(struct work_struct *work) +{ + struct rt286_priv *rt286 = + container_of(work, struct rt286_priv, jack_detect_work.work); + int status = 0; + bool hp = false; + bool mic = false; + + rt286_jack_detect(rt286, &hp, &mic); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); +} + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + rt286->jack = jack; + + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, 0, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + return 0; +} +EXPORT_SYMBOL_GPL(rt286_mic_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN, + RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN, + RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN, + 0, 0x3, 0, mic_vol_tlv), + SOC_DOUBLE_R("Speaker Playback Switch", RT286_SPOL_GAIN, + RT286_SPOR_GAIN, RT286_MUTE_SFT, 1, 1), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT286_F_DAC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", RT286_F_RECMIX_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Mic1 Switch", RT286_REC_MIC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", RT286_REC_I2S_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", RT286_REC_LINE_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Beep Switch", RT286_REC_BEEP_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spo_enable_control = + SOC_DAPM_SINGLE("Switch", RT286_SET_PIN_SPK, + RT286_SET_PIN_SFT, 1, 0); + +static const struct snd_kcontrol_new hpol_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOL_GAIN, + RT286_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hpor_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOR_GAIN, + RT286_MUTE_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "RECMIX", "Dmic" +}; + +static const int rt286_adc_values[] = { + 0, 4, 5, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, RT286_ADC0_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_ENUM("ADC 0 source", rt286_adc0_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, RT286_ADC1_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_ENUM("ADC 1 source", rt286_adc1_enum); + +static const char * const rt286_dac_src[] = { + "Front", "Surround" +}; +/* HP-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, RT286_HPO_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_spo_enum, RT286_SPK_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +static int rt286_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_HIGH); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_LOW); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0x20); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int nid; + + nid = (w->reg >> 20) & 0xff; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7080); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", RT286_SET_POWER(RT286_DMIC1), 0, 1, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA("DMIC2", RT286_SET_POWER(RT286_DMIC2), 0, 1, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* Output Mux */ + SND_SOC_DAPM_MUX("SPK Mux", SND_SOC_NOPM, 0, 0, &rt286_spo_mux), + SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt286_hpo_mux), + + SND_SOC_DAPM_SUPPLY("HP Power", RT286_SET_PIN_HPO, + RT286_SET_PIN_SFT, 0, NULL, 0), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", RT286_SET_POWER(RT286_DAC_OUT1), 0, 1, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", RT286_SET_POWER(RT286_DAC_OUT2), 0, 1, + NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0, + &spo_enable_control, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0, + &hpol_enable_control), + SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0, + &hpor_enable_control), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "RECMIX", "RECMIX"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "RECMIX", "RECMIX"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", "Switch", "SPK Mux"}, + {"HPO L", "Switch", "HPO Mux"}, + {"HPO R", "Switch", "HPO Mux"}, + {"HPO L", NULL, "HP Power"}, + {"HPO R", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO L"}, + {"HPO Pin", NULL, "HPO R"}, +}; + +static int rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_width(params)) { + /* bit 6:4 Bits per Sample */ + case 16: + d_len_code = 0; + val |= (0x1 << 4); + break; + case 32: + d_len_code = 2; + val |= (0x4 << 4); + break; + case 20: + d_len_code = 1; + val |= (0x2 << 4); + break; + case 24: + d_len_code = 2; + val |= (0x3 << 4); + break; + case 8: + d_len_code = 3; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x0018, d_len_code << 3); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); + else + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x8000, 0); + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x8000, 0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x0100, 0x0); + snd_soc_update_bits(codec, + RT286_PLL_CTRL1, 0x20, 0x20); + } else { + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x0100, 0x0100); + snd_soc_update_bits(codec, + RT286_PLL_CTRL, 0x4, 0x4); + snd_soc_update_bits(codec, + RT286_PLL_CTRL1, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x40, 0x40); + break; + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x40, 0x0); + break; + case 12288000: + case 11289600: + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x8, 0x0); + snd_soc_update_bits(codec, + RT286_CLK_DIV, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x8, 0x8); + snd_soc_update_bits(codec, + RT286_CLK_DIV, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (50 == ratio) + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x1000, 0x1000); + else + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D0); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x200); + } + break; + + case SND_SOC_BIAS_ON: + mdelay(10); + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D3); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x0); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static irqreturn_t rt286_irq(int irq, void *data) +{ + struct rt286_priv *rt286 = data; + bool hp = false; + bool mic = false; + int status = 0; + + rt286_jack_detect(rt286, &hp, &mic); + + /* Clear IRQ */ + regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x1, 0x1); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt286->i2c->dev, 300); + + return IRQ_HANDLED; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + + if (rt286->i2c->irq) { + regmap_update_bits(rt286->regmap, + RT286_IRQ_CTRL, 0x2, 0x2); + + INIT_DELAYED_WORK(&rt286->jack_detect_work, + rt286_jack_detect_work); + schedule_delayed_work(&rt286->jack_detect_work, + msecs_to_jiffies(1250)); + } + + return 0; +} + +static int rt286_remove(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&rt286->jack_detect_work); + + return 0; +} + +#ifdef CONFIG_PM +static int rt286_suspend(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt286->regmap, true); + regcache_mark_dirty(rt286->regmap); + + return 0; +} + +static int rt286_resume(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt286->regmap, false); + rt286_index_sync(codec); + regcache_sync(rt286->regmap); + + return 0; +} +#else +#define rt286_suspend NULL +#define rt286_resume NULL +#endif + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_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 rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_bclk_ratio = rt286_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "rt286-aif2", + .id = RT286_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .remove = rt286_remove, + .suspend = rt286_suspend, + .resume = rt286_resume, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct regmap_config rt286_regmap = { + .reg_bits = 32, + .val_bits = 32, + .max_register = 0x02370100, + .volatile_reg = rt286_volatile_register, + .readable_reg = rt286_readable_register, + .reg_write = rt286_hw_write, + .reg_read = rt286_hw_read, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt286_reg, + .num_reg_defaults = ARRAY_SIZE(rt286_reg), +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt286", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static const struct acpi_device_id rt286_acpi_match[] = { + { "INT343A", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + int i, ret; + + rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt286_regmap); + if (IS_ERR(rt286->regmap)) { + ret = PTR_ERR(rt286->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt286->regmap, + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); + if (ret != RT286_VENDOR_ID) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt286\n", ret); + return -ENODEV; + } + + rt286->index_cache = rt286_index_def; + rt286->i2c = i2c; + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); + + for (i = 0; i < RT286_POWER_REG_LEN; i++) + regmap_write(rt286->regmap, + RT286_SET_POWER(rt286_support_power_controls[i]), + AC_PWRST_D1); + + if (!rt286->pdata.cbj_en) { + regmap_write(rt286->regmap, RT286_CBJ_CTRL2, 0x0000); + regmap_write(rt286->regmap, RT286_MIC1_DET_CTRL, 0x0816); + regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000); + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xf000, 0xb000); + } else { + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xf000, 0x5000); + } + + mdelay(10); + + if (!rt286->pdata.gpio2_en) + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0x4000); + else + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0); + + mdelay(10); + + /*Power down LDO2*/ + regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0); + + /*Set depop parameter*/ + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL2, 0x403a, 0x401a); + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); + + if (rt286->i2c->irq) { + ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); + if (ret != 0) { + dev_err(&i2c->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + + return ret; +} + +static int rt286_i2c_remove(struct i2c_client *i2c) +{ + struct rt286_priv *rt286 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt286); + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + + +static struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt286", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .remove = rt286_i2c_remove, + .id_table = rt286_i2c_id, +}; + +module_i2c_driver(rt286_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao <bardliao@realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 000000000000..b539b7320a79 --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,198 @@ +/* + * rt286.h -- RT286 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu@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 __RT286_H__ +#define __RT286_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT286_AUDIO_FUNCTION_GROUP 0x01 +#define RT286_DAC_OUT1 0x02 +#define RT286_DAC_OUT2 0x03 +#define RT286_ADC_IN1 0x09 +#define RT286_ADC_IN2 0x08 +#define RT286_MIXER_IN 0x0b +#define RT286_MIXER_OUT1 0x0c +#define RT286_MIXER_OUT2 0x0d +#define RT286_DMIC1 0x12 +#define RT286_DMIC2 0x13 +#define RT286_SPK_OUT 0x14 +#define RT286_MIC1 0x18 +#define RT286_LINE1 0x1a +#define RT286_BEEP 0x1d +#define RT286_SPDIF 0x1e +#define RT286_VENDOR_REGISTERS 0x20 +#define RT286_HP_OUT 0x21 +#define RT286_MIXER_IN1 0x22 +#define RT286_MIXER_IN2 0x23 + +#define RT286_SET_PIN_SFT 6 +#define RT286_SET_PIN_ENABLE 0x40 +#define RT286_SET_PIN_DISABLE 0 +#define RT286_SET_EAPD_HIGH 0x2 +#define RT286_SET_EAPD_LOW 0 + +#define RT286_MUTE_SFT 7 + +/* Verb commands */ +#define RT286_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM) +#define RT286_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0) +#define RT286_SET_AUDIO_POWER RT286_SET_POWER(RT286_AUDIO_FUNCTION_GROUP) +#define RT286_SET_HPO_POWER RT286_SET_POWER(RT286_HP_OUT) +#define RT286_SET_SPK_POWER RT286_SET_POWER(RT286_SPK_OUT) +#define RT286_SET_DMIC1_POWER RT286_SET_POWER(RT286_DMIC1) +#define RT286_SPK_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_SPK_OUT, 0) +#define RT286_HPO_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_HP_OUT, 0) +#define RT286_ADC0_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN1, 0) +#define RT286_ADC1_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN2, 0) +#define RT286_SET_MIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0) +#define RT286_SET_PIN_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0) +#define RT286_SET_PIN_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0) +#define RT286_SET_PIN_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0) +#define RT286_SPK_EAPD\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0) +#define RT286_SET_AMP_GAIN_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0) +#define RT286_SET_AMP_GAIN_ADC_IN1\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0) +#define RT286_SET_AMP_GAIN_ADC_IN2\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN2, 0) +#define RT286_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0) +#define RT286_GET_MIC1_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0) +#define RT286_SET_DMIC2_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0) +#define RT286_DACL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0xa000) +#define RT286_DACR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0x9000) +#define RT286_ADCL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x6000) +#define RT286_ADCR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x5000) +#define RT286_MIC_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIC1, 0x7000) +#define RT286_SPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0xa000) +#define RT286_SPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0x9000) +#define RT286_HPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000) +#define RT286_HPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000) +#define RT286_F_DAC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7000) +#define RT286_F_RECMIX_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7100) +#define RT286_REC_MIC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7000) +#define RT286_REC_I2S_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7100) +#define RT286_REC_LINE_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7200) +#define RT286_REC_BEEP_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7300) +#define RT286_DAC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_DAC_OUT1, 0) +#define RT286_ADC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_ADC_IN1, 0) +#define RT286_COEF_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, RT286_VENDOR_REGISTERS, 0) +#define RT286_PROC_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, RT286_VENDOR_REGISTERS, 0) + +/* Index registers */ +#define RT286_A_BIAS_CTRL1 0x01 +#define RT286_A_BIAS_CTRL2 0x02 +#define RT286_POWER_CTRL1 0x03 +#define RT286_A_BIAS_CTRL3 0x04 +#define RT286_POWER_CTRL2 0x08 +#define RT286_I2S_CTRL1 0x09 +#define RT286_I2S_CTRL2 0x0a +#define RT286_CLK_DIV 0x0b +#define RT286_DC_GAIN 0x0d +#define RT286_POWER_CTRL3 0x0f +#define RT286_MIC1_DET_CTRL 0x19 +#define RT286_MISC_CTRL1 0x20 +#define RT286_IRQ_CTRL 0x33 +#define RT286_PLL_CTRL1 0x49 +#define RT286_CBJ_CTRL1 0x4f +#define RT286_CBJ_CTRL2 0x50 +#define RT286_PLL_CTRL 0x63 +#define RT286_DEPOP_CTRL1 0x66 +#define RT286_DEPOP_CTRL2 0x67 +#define RT286_DEPOP_CTRL3 0x68 +#define RT286_DEPOP_CTRL4 0x69 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_M_SPK_MUX_SFT 14 +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_M_HP_MUX_SFT 14 +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIF2, + RT286_AIFS, +}; + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +#endif /* __RT286_H__ */ + diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 30e234708579..1ba27db660a6 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1370,16 +1370,16 @@ static int rt5631_hifi_pcm_params(struct snd_pcm_substream *substream, return coeff; } - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= RT5631_SDP_I2S_DL_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= RT5631_SDP_I2S_DL_24; break; - case SNDRV_PCM_FORMAT_S8: + case 8: iface |= RT5631_SDP_I2S_DL_8; break; default: diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index de80e89b5fd8..6bc6efdec550 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2215,14 +2215,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, rt5640->hp_mute = 1; - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, - rt5640_dai, ARRAY_SIZE(rt5640_dai)); - if (ret < 0) - goto err; - - return 0; -err: - return ret; + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, + rt5640_dai, ARRAY_SIZE(rt5640_dai)); } static int rt5640_i2c_remove(struct i2c_client *i2c) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 02147be2b302..a7762d0a623e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2345,14 +2345,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, - rt5645_dai, ARRAY_SIZE(rt5645_dai)); - if (ret < 0) - goto err; - - return 0; -err: - return ret; + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, + rt5645_dai, ARRAY_SIZE(rt5645_dai)); } static int rt5645_i2c_remove(struct i2c_client *i2c) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index ea4b1c652a26..bb0a3ab5416c 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1366,16 +1366,16 @@ static int rt5651_hw_params(struct snd_pcm_substream *substream, dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", bclk_ms, pre_div, dai->id); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: val_len |= RT5651_I2S_DL_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: val_len |= RT5651_I2S_DL_24; break; - case SNDRV_PCM_FORMAT_S8: + case 8: val_len |= RT5651_I2S_DL_8; break; default: diff --git a/sound/soc/codecs/rt5670-dsp.h b/sound/soc/codecs/rt5670-dsp.h new file mode 100644 index 000000000000..a34d0cdb8198 --- /dev/null +++ b/sound/soc/codecs/rt5670-dsp.h @@ -0,0 +1,54 @@ +/* + * rt5670-dsp.h -- RT5670 ALSA SoC DSP driver + * + * Copyright 2014 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 __RT5670_DSP_H__ +#define __RT5670_DSP_H__ + +#define RT5670_DSP_CTRL1 0xe0 +#define RT5670_DSP_CTRL2 0xe1 +#define RT5670_DSP_CTRL3 0xe2 +#define RT5670_DSP_CTRL4 0xe3 +#define RT5670_DSP_CTRL5 0xe4 + +/* DSP Control 1 (0xe0) */ +#define RT5670_DSP_CMD_MASK (0xff << 8) +#define RT5670_DSP_CMD_PE (0x0d << 8) /* Patch Entry */ +#define RT5670_DSP_CMD_MW (0x3b << 8) /* Memory Write */ +#define RT5670_DSP_CMD_MR (0x37 << 8) /* Memory Read */ +#define RT5670_DSP_CMD_RR (0x60 << 8) /* Register Read */ +#define RT5670_DSP_CMD_RW (0x68 << 8) /* Register Write */ +#define RT5670_DSP_REG_DATHI (0x26 << 8) /* High Data Addr */ +#define RT5670_DSP_REG_DATLO (0x25 << 8) /* Low Data Addr */ +#define RT5670_DSP_CLK_MASK (0x3 << 6) +#define RT5670_DSP_CLK_SFT 6 +#define RT5670_DSP_CLK_768K (0x0 << 6) +#define RT5670_DSP_CLK_384K (0x1 << 6) +#define RT5670_DSP_CLK_192K (0x2 << 6) +#define RT5670_DSP_CLK_96K (0x3 << 6) +#define RT5670_DSP_BUSY_MASK (0x1 << 5) +#define RT5670_DSP_RW_MASK (0x1 << 4) +#define RT5670_DSP_DL_MASK (0x3 << 2) +#define RT5670_DSP_DL_0 (0x0 << 2) +#define RT5670_DSP_DL_1 (0x1 << 2) +#define RT5670_DSP_DL_2 (0x2 << 2) +#define RT5670_DSP_DL_3 (0x3 << 2) +#define RT5670_DSP_I2C_AL_16 (0x1 << 1) +#define RT5670_DSP_CMD_EN (0x1) + +struct rt5670_dsp_param { + u16 cmd_fmt; + u16 addr; + u16 data; + u8 cmd; +}; + +#endif /* __RT5670_DSP_H__ */ + diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c new file mode 100644 index 000000000000..ba9d9b4d4857 --- /dev/null +++ b/sound/soc/codecs/rt5670.c @@ -0,0 +1,2657 @@ +/* + * rt5670.c -- RT5670 ALSA SoC audio codec driver + * + * Copyright 2014 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 <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/rt5670.h> + +#include "rl6231.h" +#include "rt5670.h" +#include "rt5670-dsp.h" + +#define RT5670_DEVICE_ID 0x6271 + +#define RT5670_PR_RANGE_BASE (0xff + 1) +#define RT5670_PR_SPACING 0x100 + +#define RT5670_PR_BASE (RT5670_PR_RANGE_BASE + (0 * RT5670_PR_SPACING)) + +static const struct regmap_range_cfg rt5670_ranges[] = { + { .name = "PR", .range_min = RT5670_PR_BASE, + .range_max = RT5670_PR_BASE + 0xf8, + .selector_reg = RT5670_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5670_PRIV_DATA, + .window_len = 0x1, }, +}; + +static struct reg_default init_list[] = { + { RT5670_PR_BASE + 0x14, 0x9a8a }, + { RT5670_PR_BASE + 0x38, 0x3ba1 }, + { RT5670_PR_BASE + 0x3d, 0x3640 }, +}; +#define RT5670_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt5670_reg[] = { + { 0x00, 0x0000 }, + { 0x02, 0x8888 }, + { 0x03, 0x8888 }, + { 0x0a, 0x0001 }, + { 0x0b, 0x0827 }, + { 0x0c, 0x0000 }, + { 0x0d, 0x0008 }, + { 0x0e, 0x0000 }, + { 0x0f, 0x0808 }, + { 0x19, 0xafaf }, + { 0x1a, 0xafaf }, + { 0x1b, 0x0011 }, + { 0x1c, 0x2f2f }, + { 0x1d, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x1f, 0x2f2f }, + { 0x20, 0x0000 }, + { 0x26, 0x7860 }, + { 0x27, 0x7860 }, + { 0x28, 0x7871 }, + { 0x29, 0x8080 }, + { 0x2a, 0x5656 }, + { 0x2b, 0x5454 }, + { 0x2c, 0xaaa0 }, + { 0x2d, 0x0000 }, + { 0x2e, 0x2f2f }, + { 0x2f, 0x1002 }, + { 0x30, 0x0000 }, + { 0x31, 0x5f00 }, + { 0x32, 0x0000 }, + { 0x33, 0x0000 }, + { 0x34, 0x0000 }, + { 0x35, 0x0000 }, + { 0x36, 0x0000 }, + { 0x37, 0x0000 }, + { 0x38, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x45, 0xe00f }, + { 0x4c, 0x5380 }, + { 0x4f, 0x0073 }, + { 0x52, 0x00d3 }, + { 0x53, 0xf0f0 }, + { 0x61, 0x0000 }, + { 0x62, 0x0001 }, + { 0x63, 0x00c3 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x6f, 0x8000 }, + { 0x70, 0x8000 }, + { 0x71, 0x8000 }, + { 0x72, 0x8000 }, + { 0x73, 0x1110 }, + { 0x74, 0x0e00 }, + { 0x75, 0x1505 }, + { 0x76, 0x0015 }, + { 0x77, 0x0c00 }, + { 0x78, 0x4000 }, + { 0x79, 0x0123 }, + { 0x7f, 0x1100 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x83, 0x0000 }, + { 0x84, 0x0000 }, + { 0x85, 0x0000 }, + { 0x86, 0x0008 }, + { 0x87, 0x0000 }, + { 0x88, 0x0000 }, + { 0x89, 0x0000 }, + { 0x8a, 0x0000 }, + { 0x8b, 0x0000 }, + { 0x8c, 0x0007 }, + { 0x8d, 0x0000 }, + { 0x8e, 0x0004 }, + { 0x8f, 0x1100 }, + { 0x90, 0x0646 }, + { 0x91, 0x0c06 }, + { 0x93, 0x0000 }, + { 0x94, 0x0000 }, + { 0x95, 0x0000 }, + { 0x97, 0x0000 }, + { 0x98, 0x0000 }, + { 0x99, 0x0000 }, + { 0x9a, 0x2184 }, + { 0x9b, 0x010a }, + { 0x9c, 0x0aea }, + { 0x9d, 0x000c }, + { 0x9e, 0x0400 }, + { 0xae, 0x7000 }, + { 0xaf, 0x0000 }, + { 0xb0, 0x6000 }, + { 0xb1, 0x0000 }, + { 0xb2, 0x0000 }, + { 0xb3, 0x001f }, + { 0xb4, 0x2206 }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xb7, 0x0000 }, + { 0xbb, 0x0000 }, + { 0xbc, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0000 }, + { 0xc0, 0x0000 }, + { 0xc1, 0x0000 }, + { 0xc2, 0x0000 }, + { 0xcd, 0x0000 }, + { 0xce, 0x0000 }, + { 0xcf, 0x1813 }, + { 0xd0, 0x0690 }, + { 0xd1, 0x1c17 }, + { 0xd3, 0xb320 }, + { 0xd4, 0x0000 }, + { 0xd6, 0x0400 }, + { 0xd9, 0x0809 }, + { 0xda, 0x0000 }, + { 0xdb, 0x0001 }, + { 0xdc, 0x0049 }, + { 0xdd, 0x0009 }, + { 0xe6, 0x8000 }, + { 0xe7, 0x0000 }, + { 0xec, 0xb300 }, + { 0xed, 0x0000 }, + { 0xee, 0xb300 }, + { 0xef, 0x0000 }, + { 0xf8, 0x0000 }, + { 0xf9, 0x0000 }, + { 0xfa, 0x8010 }, + { 0xfb, 0x0033 }, + { 0xfc, 0x0080 }, +}; + +static bool rt5670_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5670_ranges); i++) { + if ((reg >= rt5670_ranges[i].window_start && + reg <= rt5670_ranges[i].window_start + + rt5670_ranges[i].window_len) || + (reg >= rt5670_ranges[i].range_min && + reg <= rt5670_ranges[i].range_max)) { + return true; + } + } + + switch (reg) { + case RT5670_RESET: + case RT5670_PDM_DATA_CTRL1: + case RT5670_PDM1_DATA_CTRL4: + case RT5670_PDM2_DATA_CTRL4: + case RT5670_PRIV_DATA: + case RT5670_ASRC_5: + case RT5670_CJ_CTRL1: + case RT5670_CJ_CTRL2: + case RT5670_CJ_CTRL3: + case RT5670_A_JD_CTRL1: + case RT5670_A_JD_CTRL2: + case RT5670_VAD_CTRL5: + case RT5670_ADC_EQ_CTRL1: + case RT5670_EQ_CTRL1: + case RT5670_ALC_CTRL_1: + case RT5670_IRQ_CTRL1: + case RT5670_IRQ_CTRL2: + case RT5670_INT_IRQ_ST: + case RT5670_IL_CMD: + case RT5670_DSP_CTRL1: + case RT5670_DSP_CTRL2: + case RT5670_DSP_CTRL3: + case RT5670_DSP_CTRL4: + case RT5670_DSP_CTRL5: + case RT5670_VENDOR_ID: + case RT5670_VENDOR_ID1: + case RT5670_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool rt5670_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5670_ranges); i++) { + if ((reg >= rt5670_ranges[i].window_start && + reg <= rt5670_ranges[i].window_start + + rt5670_ranges[i].window_len) || + (reg >= rt5670_ranges[i].range_min && + reg <= rt5670_ranges[i].range_max)) { + return true; + } + } + + switch (reg) { + case RT5670_RESET: + case RT5670_HP_VOL: + case RT5670_LOUT1: + case RT5670_CJ_CTRL1: + case RT5670_CJ_CTRL2: + case RT5670_CJ_CTRL3: + case RT5670_IN2: + case RT5670_INL1_INR1_VOL: + case RT5670_DAC1_DIG_VOL: + case RT5670_DAC2_DIG_VOL: + case RT5670_DAC_CTRL: + case RT5670_STO1_ADC_DIG_VOL: + case RT5670_MONO_ADC_DIG_VOL: + case RT5670_STO2_ADC_DIG_VOL: + case RT5670_ADC_BST_VOL1: + case RT5670_ADC_BST_VOL2: + case RT5670_STO2_ADC_MIXER: + case RT5670_STO1_ADC_MIXER: + case RT5670_MONO_ADC_MIXER: + case RT5670_AD_DA_MIXER: + case RT5670_STO_DAC_MIXER: + case RT5670_DD_MIXER: + case RT5670_DIG_MIXER: + case RT5670_DSP_PATH1: + case RT5670_DSP_PATH2: + case RT5670_DIG_INF1_DATA: + case RT5670_DIG_INF2_DATA: + case RT5670_PDM_OUT_CTRL: + case RT5670_PDM_DATA_CTRL1: + case RT5670_PDM1_DATA_CTRL2: + case RT5670_PDM1_DATA_CTRL3: + case RT5670_PDM1_DATA_CTRL4: + case RT5670_PDM2_DATA_CTRL2: + case RT5670_PDM2_DATA_CTRL3: + case RT5670_PDM2_DATA_CTRL4: + case RT5670_REC_L1_MIXER: + case RT5670_REC_L2_MIXER: + case RT5670_REC_R1_MIXER: + case RT5670_REC_R2_MIXER: + case RT5670_HPO_MIXER: + case RT5670_MONO_MIXER: + case RT5670_OUT_L1_MIXER: + case RT5670_OUT_R1_MIXER: + case RT5670_LOUT_MIXER: + case RT5670_PWR_DIG1: + case RT5670_PWR_DIG2: + case RT5670_PWR_ANLG1: + case RT5670_PWR_ANLG2: + case RT5670_PWR_MIXER: + case RT5670_PWR_VOL: + case RT5670_PRIV_INDEX: + case RT5670_PRIV_DATA: + case RT5670_I2S4_SDP: + case RT5670_I2S1_SDP: + case RT5670_I2S2_SDP: + case RT5670_I2S3_SDP: + case RT5670_ADDA_CLK1: + case RT5670_ADDA_CLK2: + case RT5670_DMIC_CTRL1: + case RT5670_DMIC_CTRL2: + case RT5670_TDM_CTRL_1: + case RT5670_TDM_CTRL_2: + case RT5670_TDM_CTRL_3: + case RT5670_DSP_CLK: + case RT5670_GLB_CLK: + case RT5670_PLL_CTRL1: + case RT5670_PLL_CTRL2: + case RT5670_ASRC_1: + case RT5670_ASRC_2: + case RT5670_ASRC_3: + case RT5670_ASRC_4: + case RT5670_ASRC_5: + case RT5670_ASRC_7: + case RT5670_ASRC_8: + case RT5670_ASRC_9: + case RT5670_ASRC_10: + case RT5670_ASRC_11: + case RT5670_ASRC_12: + case RT5670_ASRC_13: + case RT5670_ASRC_14: + case RT5670_DEPOP_M1: + case RT5670_DEPOP_M2: + case RT5670_DEPOP_M3: + case RT5670_CHARGE_PUMP: + case RT5670_MICBIAS: + case RT5670_A_JD_CTRL1: + case RT5670_A_JD_CTRL2: + case RT5670_VAD_CTRL1: + case RT5670_VAD_CTRL2: + case RT5670_VAD_CTRL3: + case RT5670_VAD_CTRL4: + case RT5670_VAD_CTRL5: + case RT5670_ADC_EQ_CTRL1: + case RT5670_ADC_EQ_CTRL2: + case RT5670_EQ_CTRL1: + case RT5670_EQ_CTRL2: + case RT5670_ALC_DRC_CTRL1: + case RT5670_ALC_DRC_CTRL2: + case RT5670_ALC_CTRL_1: + case RT5670_ALC_CTRL_2: + case RT5670_ALC_CTRL_3: + case RT5670_JD_CTRL: + case RT5670_IRQ_CTRL1: + case RT5670_IRQ_CTRL2: + case RT5670_INT_IRQ_ST: + case RT5670_GPIO_CTRL1: + case RT5670_GPIO_CTRL2: + case RT5670_GPIO_CTRL3: + case RT5670_SCRABBLE_FUN: + case RT5670_SCRABBLE_CTRL: + case RT5670_BASE_BACK: + case RT5670_MP3_PLUS1: + case RT5670_MP3_PLUS2: + case RT5670_ADJ_HPF1: + case RT5670_ADJ_HPF2: + case RT5670_HP_CALIB_AMP_DET: + case RT5670_SV_ZCD1: + case RT5670_SV_ZCD2: + case RT5670_IL_CMD: + case RT5670_IL_CMD2: + case RT5670_IL_CMD3: + case RT5670_DRC_HL_CTRL1: + case RT5670_DRC_HL_CTRL2: + case RT5670_ADC_MONO_HP_CTRL1: + case RT5670_ADC_MONO_HP_CTRL2: + case RT5670_ADC_STO2_HP_CTRL1: + case RT5670_ADC_STO2_HP_CTRL2: + case RT5670_JD_CTRL3: + case RT5670_JD_CTRL4: + case RT5670_DIG_MISC: + case RT5670_DSP_CTRL1: + case RT5670_DSP_CTRL2: + case RT5670_DSP_CTRL3: + case RT5670_DSP_CTRL4: + case RT5670_DSP_CTRL5: + case RT5670_GEN_CTRL2: + case RT5670_GEN_CTRL3: + case RT5670_VENDOR_ID: + case RT5670_VENDOR_ID1: + case RT5670_VENDOR_ID2: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 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 unsigned int bst_tlv[] = { + TLV_DB_RANGE_HEAD(7), + 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 rt5670_data_select[] = { + "Normal", "Swap", "left copy to right", "right copy to left" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_if2_dac_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_DAC_SEL_SFT, rt5670_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_ADC_SEL_SFT, rt5670_data_select); + +static const struct snd_kcontrol_new rt5670_snd_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE("HP Playback Switch", RT5670_HP_VOL, + RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 39, 0, out_vol_tlv), + /* OUTPUT Control */ + SOC_DOUBLE("OUT Channel Switch", RT5670_LOUT1, + RT5670_VOL_L_SFT, RT5670_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5670_LOUT1, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, 39, 1, out_vol_tlv), + /* DAC Digital Volume */ + SOC_DOUBLE("DAC2 Playback Switch", RT5670_DAC_CTRL, + RT5670_M_DAC_L2_VOL_SFT, RT5670_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5670_DAC1_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 175, 0, dac_vol_tlv), + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5670_DAC2_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 175, 0, dac_vol_tlv), + /* IN1/IN2 Control */ + SOC_SINGLE_TLV("IN1 Boost Volume", RT5670_CJ_CTRL1, + RT5670_BST_SFT1, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN2 Boost Volume", RT5670_IN2, + RT5670_BST_SFT1, 8, 0, bst_tlv), + /* INL/INR Volume Control */ + SOC_DOUBLE_TLV("IN Capture Volume", RT5670_INL1_INR1_VOL, + RT5670_INL_VOL_SFT, RT5670_INR_VOL_SFT, + 31, 1, in_vol_tlv), + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 127, 0, adc_vol_tlv), + + SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5670_MONO_ADC_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5670_ADC_BST_VOL1, + RT5670_STO1_ADC_L_BST_SFT, RT5670_STO1_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), + + SOC_DOUBLE_TLV("STO2 ADC Boost Gain Volume", RT5670_ADC_BST_VOL1, + RT5670_STO2_ADC_L_BST_SFT, RT5670_STO2_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), + + SOC_ENUM("ADC IF2 Data Switch", rt5670_if2_adc_enum), + SOC_ENUM("DAC IF2 Data Switch", rt5670_if2_dac_enum), +}; + +/** + * 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_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + int idx = -EINVAL; + + idx = rl6231_calc_dmic_clk(rt5670->sysclk); + + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5670_DMIC_CTRL1, + RT5670_DMIC_CLK_MASK, idx << RT5670_DMIC_CLK_SFT); + return idx; +} + +static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + + val = snd_soc_read(source->codec, RT5670_GLB_CLK); + val &= RT5670_SCLK_SRC_MASK; + if (val == RT5670_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int is_using_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + + switch (source->shift) { + case 0: + reg = RT5670_ASRC_3; + shift = 0; + break; + case 1: + reg = RT5670_ASRC_3; + shift = 4; + break; + case 2: + reg = RT5670_ASRC_5; + shift = 12; + break; + case 3: + reg = RT5670_ASRC_2; + shift = 0; + break; + case 8: + reg = RT5670_ASRC_2; + shift = 4; + break; + case 9: + reg = RT5670_ASRC_2; + shift = 8; + break; + case 10: + reg = RT5670_ASRC_2; + shift = 12; + break; + default: + return 0; + } + + val = (snd_soc_read(source->codec, reg) >> shift) & 0xf; + switch (val) { + case 1: + case 2: + case 3: + case 4: + return 1; + default: + return 0; + } + +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto2_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto2_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER, + RT5670_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER, + RT5670_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER, + RT5670_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER, + RT5670_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L1_STO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L1_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L2_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R2_MONO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R1_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R2_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L2_MONO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dig_l_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5670_DIG_MIXER, + RT5670_M_STO_L_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_L2_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_R2_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dig_r_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5670_DIG_MIXER, + RT5670_M_STO_R_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_R2_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_L2_DAC_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5670_rec_l_mix[] = { + SOC_DAPM_SINGLE("INL Switch", RT5670_REC_L2_MIXER, + RT5670_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5670_REC_L2_MIXER, + RT5670_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5670_REC_L2_MIXER, + RT5670_M_BST1_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_rec_r_mix[] = { + SOC_DAPM_SINGLE("INR Switch", RT5670_REC_R2_MIXER, + RT5670_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5670_REC_R2_MIXER, + RT5670_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5670_REC_R2_MIXER, + RT5670_M_BST1_RM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5670_OUT_L1_MIXER, + RT5670_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_DAC_L2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_BST2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5670_OUT_R1_MIXER, + RT5670_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_DAC_R2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_DAC_R1_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpo_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPVOL Switch", RT5670_HPO_MIXER, + RT5670_M_HPVOL_HM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpvoll_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACL1_HML_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5670_HPO_MIXER, + RT5670_M_INL1_HML_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpvolr_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACR1_HMR_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5670_HPO_MIXER, + RT5670_M_INR1_HMR_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_lout_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_LOUT_MIXER, + RT5670_M_DAC_L1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_LOUT_MIXER, + RT5670_M_DAC_R1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX L Switch", RT5670_LOUT_MIXER, + RT5670_M_OV_L_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX R Switch", RT5670_LOUT_MIXER, + RT5670_M_OV_R_LM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpl_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACL1_HML_SFT, 1, 1), + SOC_DAPM_SINGLE("INL1 Switch", RT5670_HPO_MIXER, + RT5670_M_INL1_HML_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpr_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACR1_HMR_SFT, 1, 1), + SOC_DAPM_SINGLE("INR1 Switch", RT5670_HPO_MIXER, + RT5670_M_INR1_HMR_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new lout_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5670_LOUT1, + RT5670_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new lout_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5670_LOUT1, + RT5670_R_MUTE_SFT, 1, 1); + +/* DAC1 L/R source */ /* MX-29 [9:8] [11:10] */ +static const char * const rt5670_dac1_src[] = { + "IF1 DAC", "IF2 DAC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_dac1l_enum, RT5670_AD_DA_MIXER, + RT5670_DAC1_L_SEL_SFT, rt5670_dac1_src); + +static const struct snd_kcontrol_new rt5670_dac1l_mux = + SOC_DAPM_ENUM("DAC1 L source", rt5670_dac1l_enum); + +static SOC_ENUM_SINGLE_DECL(rt5670_dac1r_enum, RT5670_AD_DA_MIXER, + RT5670_DAC1_R_SEL_SFT, rt5670_dac1_src); + +static const struct snd_kcontrol_new rt5670_dac1r_mux = + SOC_DAPM_ENUM("DAC1 R source", rt5670_dac1r_enum); + +/*DAC2 L/R source*/ /* MX-1B [6:4] [2:0] */ +/* TODO Use SOC_VALUE_ENUM_SINGLE_DECL */ +static const char * const rt5670_dac12_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "TxDC DAC", + "Bass", "VAD_ADC", "IF4 DAC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_dac2l_enum, RT5670_DAC_CTRL, + RT5670_DAC2_L_SEL_SFT, rt5670_dac12_src); + +static const struct snd_kcontrol_new rt5670_dac_l2_mux = + SOC_DAPM_ENUM("DAC2 L source", rt5670_dac2l_enum); + +static const char * const rt5670_dacr2_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "TxDC DAC", "TxDP ADC", "IF4 DAC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_dac2r_enum, RT5670_DAC_CTRL, + RT5670_DAC2_R_SEL_SFT, rt5670_dacr2_src); + +static const struct snd_kcontrol_new rt5670_dac_r2_mux = + SOC_DAPM_ENUM("DAC2 R source", rt5670_dac2r_enum); + +/*RxDP source*/ /* MX-2D [15:13] */ +static const char * const rt5670_rxdp_src[] = { + "IF2 DAC", "IF1 DAC", "STO1 ADC Mixer", "STO2 ADC Mixer", + "Mono ADC Mixer L", "Mono ADC Mixer R", "DAC1" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_rxdp_enum, RT5670_DSP_PATH1, + RT5670_RXDP_SEL_SFT, rt5670_rxdp_src); + +static const struct snd_kcontrol_new rt5670_rxdp_mux = + SOC_DAPM_ENUM("DAC2 L source", rt5670_rxdp_enum); + +/* MX-2D [1] [0] */ +static const char * const rt5670_dsp_bypass_src[] = { + "DSP", "Bypass" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_dsp_ul_enum, RT5670_DSP_PATH1, + RT5670_DSP_UL_SFT, rt5670_dsp_bypass_src); + +static const struct snd_kcontrol_new rt5670_dsp_ul_mux = + SOC_DAPM_ENUM("DSP UL source", rt5670_dsp_ul_enum); + +static SOC_ENUM_SINGLE_DECL(rt5670_dsp_dl_enum, RT5670_DSP_PATH1, + RT5670_DSP_DL_SFT, rt5670_dsp_bypass_src); + +static const struct snd_kcontrol_new rt5670_dsp_dl_mux = + SOC_DAPM_ENUM("DSP DL source", rt5670_dsp_dl_enum); + +/* Stereo2 ADC source */ +/* MX-26 [15] */ +static const char * const rt5670_stereo2_adc_lr_src[] = { + "L", "LR" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc_lr_enum, RT5670_STO2_ADC_MIXER, + RT5670_STO2_ADC_SRC_SFT, rt5670_stereo2_adc_lr_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_lr_mux = + SOC_DAPM_ENUM("Stereo2 ADC LR source", rt5670_stereo2_adc_lr_enum); + +/* Stereo1 ADC source */ +/* MX-27 MX-26 [12] */ +static const char * const rt5670_stereo_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc1_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_l1_mux = + SOC_DAPM_ENUM("Stereo1 ADC L1 source", rt5670_stereo1_adc1_enum); + +static const struct snd_kcontrol_new rt5670_sto_adc_r1_mux = + SOC_DAPM_ENUM("Stereo1 ADC R1 source", rt5670_stereo1_adc1_enum); + +static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc1_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_l1_mux = + SOC_DAPM_ENUM("Stereo2 ADC L1 source", rt5670_stereo2_adc1_enum); + +static const struct snd_kcontrol_new rt5670_sto2_adc_r1_mux = + SOC_DAPM_ENUM("Stereo2 ADC R1 source", rt5670_stereo2_adc1_enum); + +/* MX-27 MX-26 [11] */ +static const char * const rt5670_stereo_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc2_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_l2_mux = + SOC_DAPM_ENUM("Stereo1 ADC L2 source", rt5670_stereo1_adc2_enum); + +static const struct snd_kcontrol_new rt5670_sto_adc_r2_mux = + SOC_DAPM_ENUM("Stereo1 ADC R2 source", rt5670_stereo1_adc2_enum); + +static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc2_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_l2_mux = + SOC_DAPM_ENUM("Stereo2 ADC L2 source", rt5670_stereo2_adc2_enum); + +static const struct snd_kcontrol_new rt5670_sto2_adc_r2_mux = + SOC_DAPM_ENUM("Stereo2 ADC R2 source", rt5670_stereo2_adc2_enum); + +/* MX-27 MX26 [10] */ +static const char * const rt5670_stereo_adc_src[] = { + "ADC1L ADC2R", "ADC3" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_mux = + SOC_DAPM_ENUM("Stereo1 ADC source", rt5670_stereo1_adc_enum); + +static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_mux = + SOC_DAPM_ENUM("Stereo2 ADC source", rt5670_stereo2_adc_enum); + +/* MX-27 MX-26 [9:8] */ +static const char * const rt5670_stereo_dmic_src[] = { + "DMIC1", "DMIC2", "DMIC3" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_dmic_enum, RT5670_STO1_ADC_MIXER, + RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src); + +static const struct snd_kcontrol_new rt5670_sto1_dmic_mux = + SOC_DAPM_ENUM("Stereo1 DMIC source", rt5670_stereo1_dmic_enum); + +static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_dmic_enum, RT5670_STO2_ADC_MIXER, + RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src); + +static const struct snd_kcontrol_new rt5670_sto2_dmic_mux = + SOC_DAPM_ENUM("Stereo2 DMIC source", rt5670_stereo2_dmic_enum); + +/* MX-27 [0] */ +static const char * const rt5670_stereo_dmic3_src[] = { + "DMIC3", "PDM ADC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_stereo_dmic3_enum, RT5670_STO1_ADC_MIXER, + RT5670_DMIC3_SRC_SFT, rt5670_stereo_dmic3_src); + +static const struct snd_kcontrol_new rt5670_sto_dmic3_mux = + SOC_DAPM_ENUM("Stereo DMIC3 source", rt5670_stereo_dmic3_enum); + +/* Mono ADC source */ +/* MX-28 [12] */ +static const char * const rt5670_mono_adc_l1_src[] = { + "Mono DAC MIXL", "ADC1" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_mono_adc_l1_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_L1_SRC_SFT, rt5670_mono_adc_l1_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_l1_mux = + SOC_DAPM_ENUM("Mono ADC1 left source", rt5670_mono_adc_l1_enum); +/* MX-28 [11] */ +static const char * const rt5670_mono_adc_l2_src[] = { + "Mono DAC MIXL", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_mono_adc_l2_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_L2_SRC_SFT, rt5670_mono_adc_l2_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_l2_mux = + SOC_DAPM_ENUM("Mono ADC2 left source", rt5670_mono_adc_l2_enum); + +/* MX-28 [9:8] */ +static const char * const rt5670_mono_dmic_src[] = { + "DMIC1", "DMIC2", "DMIC3" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_mono_dmic_l_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_DMIC_L_SRC_SFT, rt5670_mono_dmic_src); + +static const struct snd_kcontrol_new rt5670_mono_dmic_l_mux = + SOC_DAPM_ENUM("Mono DMIC left source", rt5670_mono_dmic_l_enum); +/* MX-28 [1:0] */ +static SOC_ENUM_SINGLE_DECL(rt5670_mono_dmic_r_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_DMIC_R_SRC_SFT, rt5670_mono_dmic_src); + +static const struct snd_kcontrol_new rt5670_mono_dmic_r_mux = + SOC_DAPM_ENUM("Mono DMIC Right source", rt5670_mono_dmic_r_enum); +/* MX-28 [4] */ +static const char * const rt5670_mono_adc_r1_src[] = { + "Mono DAC MIXR", "ADC2" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_mono_adc_r1_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_R1_SRC_SFT, rt5670_mono_adc_r1_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_r1_mux = + SOC_DAPM_ENUM("Mono ADC1 right source", rt5670_mono_adc_r1_enum); +/* MX-28 [3] */ +static const char * const rt5670_mono_adc_r2_src[] = { + "Mono DAC MIXR", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_mono_adc_r2_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_R2_SRC_SFT, rt5670_mono_adc_r2_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_r2_mux = + SOC_DAPM_ENUM("Mono ADC2 right source", rt5670_mono_adc_r2_enum); + +/* MX-2D [3:2] */ +static const char * const rt5670_txdp_slot_src[] = { + "Slot 0-1", "Slot 2-3", "Slot 4-5", "Slot 6-7" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_txdp_slot_enum, RT5670_DSP_PATH1, + RT5670_TXDP_SLOT_SEL_SFT, rt5670_txdp_slot_src); + +static const struct snd_kcontrol_new rt5670_txdp_slot_mux = + SOC_DAPM_ENUM("TxDP Slot source", rt5670_txdp_slot_enum); + +/* MX-2F [15] */ +static const char * const rt5670_if1_adc2_in_src[] = { + "IF_ADC2", "VAD_ADC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_if1_adc2_in_enum, RT5670_DIG_INF1_DATA, + RT5670_IF1_ADC2_IN_SFT, rt5670_if1_adc2_in_src); + +static const struct snd_kcontrol_new rt5670_if1_adc2_in_mux = + SOC_DAPM_ENUM("IF1 ADC2 IN source", rt5670_if1_adc2_in_enum); + +/* MX-2F [14:12] */ +static const char * const rt5670_if2_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "IF_ADC3", "TxDC_DAC", "TxDP_ADC", "VAD_ADC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_in_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_ADC_IN_SFT, rt5670_if2_adc_in_src); + +static const struct snd_kcontrol_new rt5670_if2_adc_in_mux = + SOC_DAPM_ENUM("IF2 ADC IN source", rt5670_if2_adc_in_enum); + +/* MX-30 [5:4] */ +static const char * const rt5670_if4_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "IF_ADC3" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_if4_adc_in_enum, RT5670_DIG_INF2_DATA, + RT5670_IF4_ADC_IN_SFT, rt5670_if4_adc_in_src); + +static const struct snd_kcontrol_new rt5670_if4_adc_in_mux = + SOC_DAPM_ENUM("IF4 ADC IN source", rt5670_if4_adc_in_enum); + +/* MX-31 [15] [13] [11] [9] */ +static const char * const rt5670_pdm_src[] = { + "Mono DAC", "Stereo DAC" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_pdm1_l_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM1_L_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm1_l_mux = + SOC_DAPM_ENUM("PDM1 L source", rt5670_pdm1_l_enum); + +static SOC_ENUM_SINGLE_DECL(rt5670_pdm1_r_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM1_R_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm1_r_mux = + SOC_DAPM_ENUM("PDM1 R source", rt5670_pdm1_r_enum); + +static SOC_ENUM_SINGLE_DECL(rt5670_pdm2_l_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM2_L_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm2_l_mux = + SOC_DAPM_ENUM("PDM2 L source", rt5670_pdm2_l_enum); + +static SOC_ENUM_SINGLE_DECL(rt5670_pdm2_r_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM2_R_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm2_r_mux = + SOC_DAPM_ENUM("PDM2 R source", rt5670_pdm2_r_enum); + +/* MX-FA [12] */ +static const char * const rt5670_if1_adc1_in1_src[] = { + "IF_ADC1", "IF1_ADC3" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_if1_adc1_in1_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC1_IN1_SFT, rt5670_if1_adc1_in1_src); + +static const struct snd_kcontrol_new rt5670_if1_adc1_in1_mux = + SOC_DAPM_ENUM("IF1 ADC1 IN1 source", rt5670_if1_adc1_in1_enum); + +/* MX-FA [11] */ +static const char * const rt5670_if1_adc1_in2_src[] = { + "IF1_ADC1_IN1", "IF1_ADC4" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_if1_adc1_in2_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC1_IN2_SFT, rt5670_if1_adc1_in2_src); + +static const struct snd_kcontrol_new rt5670_if1_adc1_in2_mux = + SOC_DAPM_ENUM("IF1 ADC1 IN2 source", rt5670_if1_adc1_in2_enum); + +/* MX-FA [10] */ +static const char * const rt5670_if1_adc2_in1_src[] = { + "IF1_ADC2_IN", "IF1_ADC4" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_if1_adc2_in1_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC2_IN1_SFT, rt5670_if1_adc2_in1_src); + +static const struct snd_kcontrol_new rt5670_if1_adc2_in1_mux = + SOC_DAPM_ENUM("IF1 ADC2 IN1 source", rt5670_if1_adc2_in1_enum); + +/* MX-9D [9:8] */ +static const char * const rt5670_vad_adc_src[] = { + "Sto1 ADC L", "Mono ADC L", "Mono ADC R", "Sto2 ADC L" +}; + +static SOC_ENUM_SINGLE_DECL(rt5670_vad_adc_enum, RT5670_VAD_CTRL4, + RT5670_VAD_SEL_SFT, rt5670_vad_adc_src); + +static const struct snd_kcontrol_new rt5670_vad_adc_mux = + SOC_DAPM_ENUM("VAD ADC source", rt5670_vad_adc_enum); + +static int rt5670_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(rt5670->regmap, RT5670_CHARGE_PUMP, + RT5670_PM_HP_MASK, RT5670_PM_HP_HV); + regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2, + 0x0400, 0x0400); + /* headphone amp power on */ + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_HA | RT5670_PWR_FV1 | + RT5670_PWR_FV2, RT5670_PWR_HA | + RT5670_PWR_FV1 | RT5670_PWR_FV2); + /* depop parameters */ + regmap_write(rt5670->regmap, RT5670_DEPOP_M2, 0x3100); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8009); + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_HP_DCC_INT1, 0x9f00); + mdelay(20); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x0004); + msleep(30); + break; + default: + return 0; + } + + return 0; +} + +static int rt5670_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* headphone unmute sequence */ + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xb400); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0772); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x805d); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x831d); + regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2, + 0x0300, 0x0300); + regmap_update_bits(rt5670->regmap, RT5670_HP_VOL, + RT5670_L_MUTE | RT5670_R_MUTE, 0); + msleep(80); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + break; + + case SND_SOC_DAPM_PRE_PMD: + /* headphone mute sequence */ + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xb400); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0772); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x803d); + mdelay(10); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x831d); + mdelay(10); + regmap_update_bits(rt5670->regmap, RT5670_HP_VOL, + RT5670_L_MUTE | RT5670_R_MUTE, + RT5670_L_MUTE | RT5670_R_MUTE); + msleep(20); + regmap_update_bits(rt5670->regmap, + RT5670_GEN_CTRL2, 0x0300, 0x0); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0707); + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xfc00); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_bst1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST1_P, RT5670_PWR_BST1_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST1_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_bst2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST2_P, RT5670_PWR_BST2_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST2_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL1", RT5670_PWR_ANLG2, + RT5670_PWR_PLL_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S DSP", RT5670_PWR_DIG2, + RT5670_PWR_I2S_DSP_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5670_PWR_VOL, + RT5670_PWR_MIC_DET_BIT, 0, NULL, 0), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5670_ASRC_1, + 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("I2S2 ASRC", 1, RT5670_ASRC_1, + 12, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC STO ASRC", 1, RT5670_ASRC_1, + 10, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO L ASRC", 1, RT5670_ASRC_1, + 9, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO R ASRC", 1, RT5670_ASRC_1, + 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5670_ASRC_1, + 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO2 ASRC", 1, RT5670_ASRC_1, + 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO L ASRC", 1, RT5670_ASRC_1, + 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO R ASRC", 1, RT5670_ASRC_1, + 0, 0, NULL, 0), + + /* Input Side */ + /* micbias */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5670_PWR_ANLG2, + RT5670_PWR_MB1_BIT, 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + SND_SOC_DAPM_INPUT("DMIC L2"), + SND_SOC_DAPM_INPUT("DMIC R2"), + SND_SOC_DAPM_INPUT("DMIC L3"), + SND_SOC_DAPM_INPUT("DMIC R3"), + + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC3", SND_SOC_NOPM, 0, 0, NULL, 0), + + 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", RT5670_DMIC_CTRL1, + RT5670_DMIC_1_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_2_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC3 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_3_EN_SFT, 0, NULL, 0), + /* Boost */ + SND_SOC_DAPM_PGA_E("BST1", RT5670_PWR_ANLG2, RT5670_PWR_BST1_BIT, + 0, NULL, 0, rt5670_bst1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("BST2", RT5670_PWR_ANLG2, RT5670_PWR_BST2_BIT, + 0, NULL, 0, rt5670_bst2_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + /* Input Volume */ + SND_SOC_DAPM_PGA("INL VOL", RT5670_PWR_VOL, + RT5670_PWR_IN_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR VOL", RT5670_PWR_VOL, + RT5670_PWR_IN_R_BIT, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5670_PWR_MIXER, RT5670_PWR_RM_L_BIT, 0, + rt5670_rec_l_mix, ARRAY_SIZE(rt5670_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5670_PWR_MIXER, RT5670_PWR_RM_R_BIT, 0, + rt5670_rec_r_mix, ARRAY_SIZE(rt5670_rec_r_mix)), + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 2", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_PGA("ADC 1_2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("ADC 1 power", RT5670_PWR_DIG1, + RT5670_PWR_ADC_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC 2 power", RT5670_PWR_DIG1, + RT5670_PWR_ADC_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC clock", RT5670_PR_BASE + + RT5670_CHOP_DAC_ADC, 12, 0, NULL, 0), + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto1_dmic_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_l2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_r2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_r1_mux), + SND_SOC_DAPM_MUX("Stereo2 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_dmic_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_l2_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_r2_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_r1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC LR Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_lr_mux), + SND_SOC_DAPM_MUX("Mono DMIC L Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_dmic_l_mux), + SND_SOC_DAPM_MUX("Mono DMIC R Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_dmic_r_mux), + SND_SOC_DAPM_MUX("Mono ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_l2_mux), + SND_SOC_DAPM_MUX("Mono ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_l1_mux), + SND_SOC_DAPM_MUX("Mono ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_r1_mux), + SND_SOC_DAPM_MUX("Mono ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_r2_mux), + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Stereo2 Filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_S2F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_MUTE_SFT, 1, rt5670_sto1_adc_l_mix, + ARRAY_SIZE(rt5670_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", RT5670_STO1_ADC_DIG_VOL, + RT5670_R_MUTE_SFT, 1, rt5670_sto1_adc_r_mix, + ARRAY_SIZE(rt5670_sto1_adc_r_mix)), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_sto2_adc_l_mix, + ARRAY_SIZE(rt5670_sto2_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_sto2_adc_r_mix, + ARRAY_SIZE(rt5670_sto2_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("ADC Mono Left Filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXL", RT5670_MONO_ADC_DIG_VOL, + RT5670_L_MUTE_SFT, 1, rt5670_mono_adc_l_mix, + ARRAY_SIZE(rt5670_mono_adc_l_mix)), + SND_SOC_DAPM_SUPPLY("ADC Mono Right Filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXR", RT5670_MONO_ADC_DIG_VOL, + RT5670_R_MUTE_SFT, 1, rt5670_mono_adc_r_mix, + ARRAY_SIZE(rt5670_mono_adc_r_mix)), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo1 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Sto2 ADC LR MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("VAD_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DSP */ + SND_SOC_DAPM_PGA("TxDP_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDP_ADC_L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDP_ADC_R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDC_DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("TDM Data Mux", SND_SOC_NOPM, 0, 0, + &rt5670_txdp_slot_mux), + + SND_SOC_DAPM_MUX("DSP UL Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dsp_ul_mux), + SND_SOC_DAPM_MUX("DSP DL Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dsp_dl_mux), + + SND_SOC_DAPM_MUX("RxDP Mux", SND_SOC_NOPM, 0, 0, + &rt5670_rxdp_mux), + + /* IF2 Mux */ + SND_SOC_DAPM_MUX("IF2 ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if2_adc_in_mux), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5670_PWR_DIG1, + RT5670_PWR_I2S1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2", 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), + SND_SOC_DAPM_PGA("IF1 DAC2 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5670_PWR_DIG1, + RT5670_PWR_I2S2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 ADC1 IN1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc1_in1_mux), + SND_SOC_DAPM_MUX("IF1 ADC1 IN2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc1_in2_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 IN Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc2_in_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 IN1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc2_in1_mux), + SND_SOC_DAPM_MUX("VAD ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_vad_adc_mux), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, + RT5670_GPIO_CTRL1, RT5670_I2S2_PIN_SFT, 1), + + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5670_dac_l_mix, ARRAY_SIZE(rt5670_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5670_dac_r_mix, ARRAY_SIZE(rt5670_dac_r_mix)), + SND_SOC_DAPM_PGA("DAC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac_r2_mux), + SND_SOC_DAPM_PGA("DAC L2 Volume", RT5670_PWR_DIG1, + RT5670_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC R2 Volume", RT5670_PWR_DIG1, + RT5670_PWR_DAC_R2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DAC1 L Mux", SND_SOC_NOPM, 0, 0, &rt5670_dac1l_mux), + SND_SOC_DAPM_MUX("DAC1 R Mux", SND_SOC_NOPM, 0, 0, &rt5670_dac1r_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Mono Left Filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Mono Right Filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_sto_dac_l_mix, + ARRAY_SIZE(rt5670_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_sto_dac_r_mix, + ARRAY_SIZE(rt5670_sto_dac_r_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_mono_dac_l_mix, + ARRAY_SIZE(rt5670_mono_dac_l_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_mono_dac_r_mix, + ARRAY_SIZE(rt5670_mono_dac_r_mix)), + SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_dig_l_mix, + ARRAY_SIZE(rt5670_dig_l_mix)), + SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_dig_r_mix, + ARRAY_SIZE(rt5670_dig_r_mix)), + + /* DACs */ + SND_SOC_DAPM_SUPPLY("DAC L1 Power", RT5670_PWR_DIG1, + RT5670_PWR_DAC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R1 Power", RT5670_PWR_DIG1, + RT5670_PWR_DAC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DAC L1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5670_PWR_DIG1, + RT5670_PWR_DAC_L2_BIT, 0), + + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5670_PWR_DIG1, + RT5670_PWR_DAC_R2_BIT, 0), + /* OUT Mixer */ + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5670_PWR_MIXER, RT5670_PWR_OM_L_BIT, + 0, rt5670_out_l_mix, ARRAY_SIZE(rt5670_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5670_PWR_MIXER, RT5670_PWR_OM_R_BIT, + 0, rt5670_out_r_mix, ARRAY_SIZE(rt5670_out_r_mix)), + /* Ouput Volume */ + SND_SOC_DAPM_MIXER("HPOVOL MIXL", RT5670_PWR_VOL, + RT5670_PWR_HV_L_BIT, 0, + rt5670_hpvoll_mix, ARRAY_SIZE(rt5670_hpvoll_mix)), + SND_SOC_DAPM_MIXER("HPOVOL MIXR", RT5670_PWR_VOL, + RT5670_PWR_HV_R_BIT, 0, + rt5670_hpvolr_mix, ARRAY_SIZE(rt5670_hpvolr_mix)), + SND_SOC_DAPM_PGA("DAC 1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC 2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, + rt5670_hpo_mix, ARRAY_SIZE(rt5670_hpo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", RT5670_PWR_ANLG1, RT5670_PWR_LM_BIT, + 0, rt5670_lout_mix, ARRAY_SIZE(rt5670_lout_mix)), + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, 0, 0, + rt5670_hp_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5670_PWR_ANLG1, + RT5670_PWR_HP_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5670_PWR_ANLG1, + RT5670_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5670_hp_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("LOUT L Playback", SND_SOC_NOPM, 0, 0, + &lout_l_enable_control), + SND_SOC_DAPM_SWITCH("LOUT R Playback", SND_SOC_NOPM, 0, 0, + &lout_r_enable_control), + SND_SOC_DAPM_PGA("LOUT Amp", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* PDM */ + SND_SOC_DAPM_SUPPLY("PDM1 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MUX("PDM1 L Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_L_SFT, 1, &rt5670_pdm1_l_mux), + SND_SOC_DAPM_MUX("PDM1 R Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_R_SFT, 1, &rt5670_pdm1_r_mux), + SND_SOC_DAPM_MUX("PDM2 L Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_L_SFT, 1, &rt5670_pdm2_l_mux), + SND_SOC_DAPM_MUX("PDM2 R Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_R_SFT, 1, &rt5670_pdm2_r_mux), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("PDM1L"), + SND_SOC_DAPM_OUTPUT("PDM1R"), + SND_SOC_DAPM_OUTPUT("PDM2L"), + SND_SOC_DAPM_OUTPUT("PDM2R"), +}; + +static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { + { "ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc }, + { "ADC Stereo2 Filter", NULL, "ADC STO2 ASRC", is_using_asrc }, + { "ADC Mono Left Filter", NULL, "ADC MONO L ASRC", is_using_asrc }, + { "ADC Mono Right Filter", NULL, "ADC MONO R ASRC", is_using_asrc }, + { "DAC Mono Left Filter", NULL, "DAC MONO L ASRC", is_using_asrc }, + { "DAC Mono Right Filter", NULL, "DAC MONO R ASRC", is_using_asrc }, + { "DAC Stereo1 Filter", NULL, "DAC STO ASRC", is_using_asrc }, + + { "I2S1", NULL, "I2S1 ASRC" }, + { "I2S2", NULL, "I2S2 ASRC" }, + + { "DMIC1", NULL, "DMIC L1" }, + { "DMIC1", NULL, "DMIC R1" }, + { "DMIC2", NULL, "DMIC L2" }, + { "DMIC2", NULL, "DMIC R2" }, + { "DMIC3", NULL, "DMIC L3" }, + { "DMIC3", NULL, "DMIC R3" }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST1", NULL, "Mic Det Power" }, + { "BST2", NULL, "IN2P" }, + { "BST2", NULL, "IN2N" }, + + { "INL VOL", NULL, "IN2P" }, + { "INR VOL", NULL, "IN2N" }, + + { "RECMIXL", "INL Switch", "INL VOL" }, + { "RECMIXL", "BST2 Switch", "BST2" }, + { "RECMIXL", "BST1 Switch", "BST1" }, + + { "RECMIXR", "INR Switch", "INR VOL" }, + { "RECMIXR", "BST2 Switch", "BST2" }, + { "RECMIXR", "BST1 Switch", "BST1" }, + + { "ADC 1", NULL, "RECMIXL" }, + { "ADC 1", NULL, "ADC 1 power" }, + { "ADC 1", NULL, "ADC clock" }, + { "ADC 2", NULL, "RECMIXR" }, + { "ADC 2", NULL, "ADC 2 power" }, + { "ADC 2", NULL, "ADC clock" }, + + { "DMIC L1", NULL, "DMIC CLK" }, + { "DMIC L1", NULL, "DMIC1 Power" }, + { "DMIC R1", NULL, "DMIC CLK" }, + { "DMIC R1", NULL, "DMIC1 Power" }, + { "DMIC L2", NULL, "DMIC CLK" }, + { "DMIC L2", NULL, "DMIC2 Power" }, + { "DMIC R2", NULL, "DMIC CLK" }, + { "DMIC R2", NULL, "DMIC2 Power" }, + { "DMIC L3", NULL, "DMIC CLK" }, + { "DMIC L3", NULL, "DMIC3 Power" }, + { "DMIC R3", NULL, "DMIC CLK" }, + { "DMIC R3", NULL, "DMIC3 Power" }, + + { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo1 DMIC Mux", "DMIC3", "DMIC3" }, + + { "Stereo2 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo2 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo2 DMIC Mux", "DMIC3", "DMIC3" }, + + { "Mono DMIC L Mux", "DMIC1", "DMIC L1" }, + { "Mono DMIC L Mux", "DMIC2", "DMIC L2" }, + { "Mono DMIC L Mux", "DMIC3", "DMIC L3" }, + + { "Mono DMIC R Mux", "DMIC1", "DMIC R1" }, + { "Mono DMIC R Mux", "DMIC2", "DMIC R2" }, + { "Mono DMIC R Mux", "DMIC3", "DMIC R3" }, + + { "ADC 1_2", NULL, "ADC 1" }, + { "ADC 1_2", NULL, "ADC 2" }, + + { "Stereo1 ADC L2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, + { "Stereo1 ADC L1 Mux", "ADC", "ADC 1_2" }, + { "Stereo1 ADC L1 Mux", "DAC MIX", "DAC MIXL" }, + + { "Stereo1 ADC R1 Mux", "ADC", "ADC 1_2" }, + { "Stereo1 ADC R1 Mux", "DAC MIX", "DAC MIXR" }, + { "Stereo1 ADC R2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC R2 Mux", "DAC MIX", "DAC MIXR" }, + + { "Mono ADC L2 Mux", "DMIC", "Mono DMIC L Mux" }, + { "Mono ADC L2 Mux", "Mono DAC MIXL", "Mono DAC MIXL" }, + { "Mono ADC L1 Mux", "Mono DAC MIXL", "Mono DAC MIXL" }, + { "Mono ADC L1 Mux", "ADC1", "ADC 1" }, + + { "Mono ADC R1 Mux", "Mono DAC MIXR", "Mono DAC MIXR" }, + { "Mono ADC R1 Mux", "ADC2", "ADC 2" }, + { "Mono ADC R2 Mux", "DMIC", "Mono DMIC R Mux" }, + { "Mono ADC R2 Mux", "Mono DAC MIXR", "Mono DAC MIXR" }, + + { "Sto1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux" }, + { "Sto1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux" }, + { "Sto1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux" }, + { "Sto1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux" }, + + { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, + { "Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter" }, + { "ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, + { "Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter" }, + { "ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Mono ADC MIXL", "ADC1 Switch", "Mono ADC L1 Mux" }, + { "Mono ADC MIXL", "ADC2 Switch", "Mono ADC L2 Mux" }, + { "Mono ADC MIXL", NULL, "ADC Mono Left Filter" }, + { "ADC Mono Left Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Mono ADC MIXR", "ADC1 Switch", "Mono ADC R1 Mux" }, + { "Mono ADC MIXR", "ADC2 Switch", "Mono ADC R2 Mux" }, + { "Mono ADC MIXR", NULL, "ADC Mono Right Filter" }, + { "ADC Mono Right Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo2 ADC L2 Mux", "DMIC", "Stereo2 DMIC Mux" }, + { "Stereo2 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, + { "Stereo2 ADC L1 Mux", "ADC", "ADC 1_2" }, + { "Stereo2 ADC L1 Mux", "DAC MIX", "DAC MIXL" }, + + { "Stereo2 ADC R1 Mux", "ADC", "ADC 1_2" }, + { "Stereo2 ADC R1 Mux", "DAC MIX", "DAC MIXR" }, + { "Stereo2 ADC R2 Mux", "DMIC", "Stereo2 DMIC Mux" }, + { "Stereo2 ADC R2 Mux", "DAC MIX", "DAC MIXR" }, + + { "Sto2 ADC MIXL", "ADC1 Switch", "Stereo2 ADC L1 Mux" }, + { "Sto2 ADC MIXL", "ADC2 Switch", "Stereo2 ADC L2 Mux" }, + { "Sto2 ADC MIXR", "ADC1 Switch", "Stereo2 ADC R1 Mux" }, + { "Sto2 ADC MIXR", "ADC2 Switch", "Stereo2 ADC R2 Mux" }, + + { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXL" }, + { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXR" }, + + { "Stereo2 ADC LR Mux", "L", "Sto2 ADC MIXL" }, + { "Stereo2 ADC LR Mux", "LR", "Sto2 ADC LR MIX" }, + + { "Stereo2 ADC MIXL", NULL, "Stereo2 ADC LR Mux" }, + { "Stereo2 ADC MIXL", NULL, "ADC Stereo2 Filter" }, + { "ADC Stereo2 Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo2 ADC MIXR", NULL, "Sto2 ADC MIXR" }, + { "Stereo2 ADC MIXR", NULL, "ADC Stereo2 Filter" }, + { "ADC Stereo2 Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "VAD ADC Mux", "Sto1 ADC L", "Stereo1 ADC MIXL" }, + { "VAD ADC Mux", "Mono ADC L", "Mono ADC MIXL" }, + { "VAD ADC Mux", "Mono ADC R", "Mono ADC MIXR" }, + { "VAD ADC Mux", "Sto2 ADC L", "Sto2 ADC MIXL" }, + + { "VAD_ADC", NULL, "VAD ADC Mux" }, + + { "IF_ADC1", NULL, "Stereo1 ADC MIXL" }, + { "IF_ADC1", NULL, "Stereo1 ADC MIXR" }, + { "IF_ADC2", NULL, "Mono ADC MIXL" }, + { "IF_ADC2", NULL, "Mono ADC MIXR" }, + { "IF_ADC3", NULL, "Stereo2 ADC MIXL" }, + { "IF_ADC3", NULL, "Stereo2 ADC MIXR" }, + + { "IF1 ADC1 IN1 Mux", "IF_ADC1", "IF_ADC1" }, + { "IF1 ADC1 IN1 Mux", "IF1_ADC3", "IF1_ADC3" }, + + { "IF1 ADC1 IN2 Mux", "IF1_ADC1_IN1", "IF1 ADC1 IN1 Mux" }, + { "IF1 ADC1 IN2 Mux", "IF1_ADC4", "IF1_ADC4" }, + + { "IF1 ADC2 IN Mux", "IF_ADC2", "IF_ADC2" }, + { "IF1 ADC2 IN Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF1 ADC2 IN1 Mux", "IF1_ADC2_IN", "IF1 ADC2 IN Mux" }, + { "IF1 ADC2 IN1 Mux", "IF1_ADC4", "IF1_ADC4" }, + + { "IF1_ADC1" , NULL, "IF1 ADC1 IN2 Mux" }, + { "IF1_ADC2" , NULL, "IF1 ADC2 IN1 Mux" }, + + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL" }, + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR" }, + { "Stereo2 ADC MIX", NULL, "Sto2 ADC MIXL" }, + { "Stereo2 ADC MIX", NULL, "Sto2 ADC MIXR" }, + { "Mono ADC MIX", NULL, "Mono ADC MIXL" }, + { "Mono ADC MIX", NULL, "Mono ADC MIXR" }, + + { "RxDP Mux", "IF2 DAC", "IF2 DAC" }, + { "RxDP Mux", "IF1 DAC", "IF1 DAC2" }, + { "RxDP Mux", "STO1 ADC Mixer", "Stereo1 ADC MIX" }, + { "RxDP Mux", "STO2 ADC Mixer", "Stereo2 ADC MIX" }, + { "RxDP Mux", "Mono ADC Mixer L", "Mono ADC MIXL" }, + { "RxDP Mux", "Mono ADC Mixer R", "Mono ADC MIXR" }, + { "RxDP Mux", "DAC1", "DAC MIX" }, + + { "TDM Data Mux", "Slot 0-1", "Stereo1 ADC MIX" }, + { "TDM Data Mux", "Slot 2-3", "Mono ADC MIX" }, + { "TDM Data Mux", "Slot 4-5", "Stereo2 ADC MIX" }, + { "TDM Data Mux", "Slot 6-7", "IF2 DAC" }, + + { "DSP UL Mux", "Bypass", "TDM Data Mux" }, + { "DSP UL Mux", NULL, "I2S DSP" }, + { "DSP DL Mux", "Bypass", "RxDP Mux" }, + { "DSP DL Mux", NULL, "I2S DSP" }, + + { "TxDP_ADC_L", NULL, "DSP UL Mux" }, + { "TxDP_ADC_R", NULL, "DSP UL Mux" }, + { "TxDC_DAC", NULL, "DSP DL Mux" }, + + { "TxDP_ADC", NULL, "TxDP_ADC_L" }, + { "TxDP_ADC", NULL, "TxDP_ADC_R" }, + + { "IF1 ADC", NULL, "I2S1" }, + { "IF1 ADC", NULL, "IF1_ADC1" }, + { "IF1 ADC", NULL, "IF1_ADC2" }, + { "IF1 ADC", NULL, "IF_ADC3" }, + { "IF1 ADC", NULL, "TxDP_ADC" }, + + { "IF2 ADC Mux", "IF_ADC1", "IF_ADC1" }, + { "IF2 ADC Mux", "IF_ADC2", "IF_ADC2" }, + { "IF2 ADC Mux", "IF_ADC3", "IF_ADC3" }, + { "IF2 ADC Mux", "TxDC_DAC", "TxDC_DAC" }, + { "IF2 ADC Mux", "TxDP_ADC", "TxDP_ADC" }, + { "IF2 ADC Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF2 ADC L", NULL, "IF2 ADC Mux" }, + { "IF2 ADC R", NULL, "IF2 ADC Mux" }, + + { "IF2 ADC", NULL, "I2S2" }, + { "IF2 ADC", NULL, "IF2 ADC L" }, + { "IF2 ADC", NULL, "IF2 ADC R" }, + + { "AIF1TX", NULL, "IF1 ADC" }, + { "AIF2TX", NULL, "IF2 ADC" }, + + { "IF1 DAC1", NULL, "AIF1RX" }, + { "IF1 DAC2", NULL, "AIF1RX" }, + { "IF2 DAC", NULL, "AIF2RX" }, + + { "IF1 DAC1", NULL, "I2S1" }, + { "IF1 DAC2", NULL, "I2S1" }, + { "IF2 DAC", NULL, "I2S2" }, + + { "IF1 DAC2 L", NULL, "IF1 DAC2" }, + { "IF1 DAC2 R", NULL, "IF1 DAC2" }, + { "IF1 DAC1 L", NULL, "IF1 DAC1" }, + { "IF1 DAC1 R", NULL, "IF1 DAC1" }, + { "IF2 DAC L", NULL, "IF2 DAC" }, + { "IF2 DAC R", NULL, "IF2 DAC" }, + + { "DAC1 L Mux", "IF1 DAC", "IF1 DAC1 L" }, + { "DAC1 L Mux", "IF2 DAC", "IF2 DAC L" }, + + { "DAC1 R Mux", "IF1 DAC", "IF1 DAC1 R" }, + { "DAC1 R Mux", "IF2 DAC", "IF2 DAC R" }, + + { "DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL" }, + { "DAC1 MIXL", "DAC1 Switch", "DAC1 L Mux" }, + { "DAC1 MIXL", NULL, "DAC Stereo1 Filter" }, + { "DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR" }, + { "DAC1 MIXR", "DAC1 Switch", "DAC1 R Mux" }, + { "DAC1 MIXR", NULL, "DAC Stereo1 Filter" }, + + { "DAC MIX", NULL, "DAC1 MIXL" }, + { "DAC MIX", NULL, "DAC1 MIXR" }, + + { "Audio DSP", NULL, "DAC1 MIXL" }, + { "Audio DSP", NULL, "DAC1 MIXR" }, + + { "DAC L2 Mux", "IF1 DAC", "IF1 DAC2 L" }, + { "DAC L2 Mux", "IF2 DAC", "IF2 DAC L" }, + { "DAC L2 Mux", "TxDC DAC", "TxDC_DAC" }, + { "DAC L2 Mux", "VAD_ADC", "VAD_ADC" }, + { "DAC L2 Volume", NULL, "DAC L2 Mux" }, + { "DAC L2 Volume", NULL, "DAC Mono Left Filter" }, + + { "DAC R2 Mux", "IF1 DAC", "IF1 DAC2 R" }, + { "DAC R2 Mux", "IF2 DAC", "IF2 DAC R" }, + { "DAC R2 Mux", "TxDC DAC", "TxDC_DAC" }, + { "DAC R2 Mux", "TxDP ADC", "TxDP_ADC" }, + { "DAC R2 Volume", NULL, "DAC R2 Mux" }, + { "DAC R2 Volume", NULL, "DAC Mono Right Filter" }, + + { "Stereo DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXL", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "Stereo DAC MIXL", NULL, "DAC Stereo1 Filter" }, + { "Stereo DAC MIXL", NULL, "DAC L1 Power" }, + { "Stereo DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXR", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "Stereo DAC MIXR", NULL, "DAC Stereo1 Filter" }, + { "Stereo DAC MIXR", NULL, "DAC R1 Power" }, + + { "Mono DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" }, + { "Mono DAC MIXL", NULL, "DAC Mono Left Filter" }, + { "Mono DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, + { "Mono DAC MIXR", NULL, "DAC Mono Right Filter" }, + + { "DAC MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, + { "DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" }, + { "DAC MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, + { "DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, + + { "DAC L1", NULL, "DAC L1 Power" }, + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC L1", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC R1", NULL, "DAC R1 Power" }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + { "DAC R1", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC L2", NULL, "Mono DAC MIXL" }, + { "DAC L2", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC R2", NULL, "Mono DAC MIXR" }, + { "DAC R2", NULL, "PLL1", is_sys_clk_from_pll }, + + { "OUT MIXL", "BST1 Switch", "BST1" }, + { "OUT MIXL", "INL Switch", "INL VOL" }, + { "OUT MIXL", "DAC L2 Switch", "DAC L2" }, + { "OUT MIXL", "DAC L1 Switch", "DAC L1" }, + + { "OUT MIXR", "BST2 Switch", "BST2" }, + { "OUT MIXR", "INR Switch", "INR VOL" }, + { "OUT MIXR", "DAC R2 Switch", "DAC R2" }, + { "OUT MIXR", "DAC R1 Switch", "DAC R1" }, + + { "HPOVOL MIXL", "DAC1 Switch", "DAC L1" }, + { "HPOVOL MIXL", "INL Switch", "INL VOL" }, + { "HPOVOL MIXR", "DAC1 Switch", "DAC R1" }, + { "HPOVOL MIXR", "INR Switch", "INR VOL" }, + + { "DAC 2", NULL, "DAC L2" }, + { "DAC 2", NULL, "DAC R2" }, + { "DAC 1", NULL, "DAC L1" }, + { "DAC 1", NULL, "DAC R1" }, + { "HPOVOL", NULL, "HPOVOL MIXL" }, + { "HPOVOL", NULL, "HPOVOL MIXR" }, + { "HPO MIX", "DAC1 Switch", "DAC 1" }, + { "HPO MIX", "HPVOL Switch", "HPOVOL" }, + + { "LOUT MIX", "DAC L1 Switch", "DAC L1" }, + { "LOUT MIX", "DAC R1 Switch", "DAC R1" }, + { "LOUT MIX", "OUTMIX L Switch", "OUT MIXL" }, + { "LOUT MIX", "OUTMIX R Switch", "OUT MIXR" }, + + { "PDM1 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM1 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM1 L Mux", NULL, "PDM1 Power" }, + { "PDM1 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM1 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM1 R Mux", NULL, "PDM1 Power" }, + { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM2 L Mux", NULL, "PDM2 Power" }, + { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM2 R Mux", NULL, "PDM2 Power" }, + + { "HP Amp", NULL, "HPO MIX" }, + { "HP Amp", NULL, "Mic Det Power" }, + { "HPOL", NULL, "HP Amp" }, + { "HPOL", NULL, "HP L Amp" }, + { "HPOL", NULL, "Improve HP Amp Drv" }, + { "HPOR", NULL, "HP Amp" }, + { "HPOR", NULL, "HP R Amp" }, + { "HPOR", NULL, "Improve HP Amp Drv" }, + + { "LOUT Amp", NULL, "LOUT MIX" }, + { "LOUT L Playback", "Switch", "LOUT Amp" }, + { "LOUT R Playback", "Switch", "LOUT Amp" }, + { "LOUTL", NULL, "LOUT L Playback" }, + { "LOUTR", NULL, "LOUT R Playback" }, + { "LOUTL", NULL, "Improve HP Amp Drv" }, + { "LOUTR", NULL, "Improve HP Amp Drv" }, + + { "PDM1L", NULL, "PDM1 L Mux" }, + { "PDM1R", NULL, "PDM1 R Mux" }, + { "PDM2L", NULL, "PDM2 L Mux" }, + { "PDM2R", NULL, "PDM2 R Mux" }, +}; + +static int rt5670_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + rt5670->lrck[dai->id] = params_rate(params); + pre_div = rl6231_get_clk_info(rt5670->sysclk, rt5670->lrck[dai->id]); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5670->lrck[dai->id], dai->id); + return -EINVAL; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + bclk_ms = frame_size > 32; + rt5670->bclk[dai->id] = rt5670->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5670->bclk[dai->id], rt5670->lrck[dai->id]); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= RT5670_I2S_DL_20; + break; + case 24: + val_len |= RT5670_I2S_DL_24; + break; + case 8: + val_len |= RT5670_I2S_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5670_AIF1: + mask_clk = RT5670_I2S_BCLK_MS1_MASK | RT5670_I2S_PD1_MASK; + val_clk = bclk_ms << RT5670_I2S_BCLK_MS1_SFT | + pre_div << RT5670_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5670_I2S1_SDP, + RT5670_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5670_ADDA_CLK1, mask_clk, val_clk); + break; + case RT5670_AIF2: + mask_clk = RT5670_I2S_BCLK_MS2_MASK | RT5670_I2S_PD2_MASK; + val_clk = bclk_ms << RT5670_I2S_BCLK_MS2_SFT | + pre_div << RT5670_I2S_PD2_SFT; + snd_soc_update_bits(codec, RT5670_I2S2_SDP, + RT5670_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5670_ADDA_CLK1, mask_clk, val_clk); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5670_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5670->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5670_I2S_MS_S; + rt5670->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 |= RT5670_I2S_BP_INV; + 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 |= RT5670_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5670_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5670_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5670_AIF1: + snd_soc_update_bits(codec, RT5670_I2S1_SDP, + RT5670_I2S_MS_MASK | RT5670_I2S_BP_MASK | + RT5670_I2S_DF_MASK, reg_val); + break; + case RT5670_AIF2: + snd_soc_update_bits(codec, RT5670_I2S2_SDP, + RT5670_I2S_MS_MASK | RT5670_I2S_BP_MASK | + RT5670_I2S_DF_MASK, reg_val); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5670->sysclk && clk_id == rt5670->sysclk_src) + return 0; + + switch (clk_id) { + case RT5670_SCLK_S_MCLK: + reg_val |= RT5670_SCLK_SRC_MCLK; + break; + case RT5670_SCLK_S_PLL1: + reg_val |= RT5670_SCLK_SRC_PLL1; + break; + case RT5670_SCLK_S_RCCLK: + reg_val |= RT5670_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, reg_val); + rt5670->sysclk = freq; + rt5670->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +static int rt5670_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt5670->pll_src && freq_in == rt5670->pll_in && + freq_out == rt5670->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5670->pll_in = 0; + rt5670->pll_out = 0; + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5670_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_MCLK); + break; + case RT5670_PLL1_S_BCLK1: + case RT5670_PLL1_S_BCLK2: + case RT5670_PLL1_S_BCLK3: + case RT5670_PLL1_S_BCLK4: + switch (dai->id) { + case RT5670_AIF1: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_BCLK1); + break; + case RT5670_AIF2: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_BCLK2); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->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_write(codec, RT5670_PLL_CTRL1, + pll_code.n_code << RT5670_PLL_N_SFT | pll_code.k_code); + snd_soc_write(codec, RT5670_PLL_CTRL2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5670_PLL_M_SFT | + pll_code.m_bp << RT5670_PLL_M_BP_SFT); + + rt5670->pll_in = freq_in; + rt5670->pll_out = freq_out; + rt5670->pll_src = source; + + return 0; +} + +static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + + if (rx_mask || tx_mask) + val |= (1 << 14); + + switch (slots) { + case 4: + val |= (1 << 12); + break; + case 6: + val |= (2 << 12); + break; + case 8: + val |= (3 << 12); + break; + case 2: + break; + default: + return -EINVAL; + } + + switch (slot_width) { + case 20: + val |= (1 << 10); + break; + case 24: + val |= (2 << 10); + break; + case 32: + val |= (3 << 10); + break; + case 16: + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5670_TDM_CTRL_1, 0x7c00, val); + + return 0; +} + +static int rt5670_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2); + mdelay(10); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_FV1 | RT5670_PWR_FV2, + RT5670_PWR_FV1 | RT5670_PWR_FV2); + snd_soc_update_bits(codec, RT5670_CHARGE_PUMP, + RT5670_OSW_L_MASK | RT5670_OSW_R_MASK, + RT5670_OSW_L_DIS | RT5670_OSW_R_DIS); + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x1); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_LDO_SEL_MASK, 0x3); + } + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, RT5670_PWR_DIG1, 0x0000); + snd_soc_write(codec, RT5670_PWR_DIG2, 0x0001); + snd_soc_write(codec, RT5670_PWR_VOL, 0x0000); + snd_soc_write(codec, RT5670_PWR_MIXER, 0x0001); + snd_soc_write(codec, RT5670_PWR_ANLG1, 0x2800); + snd_soc_write(codec, RT5670_PWR_ANLG2, 0x0004); + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_LDO_SEL_MASK, 0x1); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5670_probe(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + rt5670->codec = codec; + + return 0; +} + +static int rt5670_remove(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + return 0; +} + +#ifdef CONFIG_PM +static int rt5670_suspend(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5670->regmap, true); + regcache_mark_dirty(rt5670->regmap); + return 0; +} + +static int rt5670_resume(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5670->regmap, false); + regcache_sync(rt5670->regmap); + + return 0; +} +#else +#define rt5670_suspend NULL +#define rt5670_resume NULL +#endif + +#define RT5670_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5670_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static struct snd_soc_dai_ops rt5670_aif_dai_ops = { + .hw_params = rt5670_hw_params, + .set_fmt = rt5670_set_dai_fmt, + .set_sysclk = rt5670_set_dai_sysclk, + .set_tdm_slot = rt5670_set_tdm_slot, + .set_pll = rt5670_set_dai_pll, +}; + +static struct snd_soc_dai_driver rt5670_dai[] = { + { + .name = "rt5670-aif1", + .id = RT5670_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .ops = &rt5670_aif_dai_ops, + }, + { + .name = "rt5670-aif2", + .id = RT5670_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .ops = &rt5670_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5670 = { + .probe = rt5670_probe, + .remove = rt5670_remove, + .suspend = rt5670_suspend, + .resume = rt5670_resume, + .set_bias_level = rt5670_set_bias_level, + .idle_bias_off = true, + .controls = rt5670_snd_controls, + .num_controls = ARRAY_SIZE(rt5670_snd_controls), + .dapm_widgets = rt5670_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5670_dapm_widgets), + .dapm_routes = rt5670_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5670_dapm_routes), +}; + +static const struct regmap_config rt5670_regmap = { + .reg_bits = 8, + .val_bits = 16, + .max_register = RT5670_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5670_ranges) * + RT5670_PR_SPACING), + .volatile_reg = rt5670_volatile_register, + .readable_reg = rt5670_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5670_reg, + .num_reg_defaults = ARRAY_SIZE(rt5670_reg), + .ranges = rt5670_ranges, + .num_ranges = ARRAY_SIZE(rt5670_ranges), +}; + +static const struct i2c_device_id rt5670_i2c_id[] = { + { "rt5670", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); + +static int rt5670_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5670_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5670_priv *rt5670; + int ret; + unsigned int val; + + rt5670 = devm_kzalloc(&i2c->dev, + sizeof(struct rt5670_priv), + GFP_KERNEL); + if (NULL == rt5670) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5670); + + if (pdata) + rt5670->pdata = *pdata; + + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); + if (IS_ERR(rt5670->regmap)) { + ret = PTR_ERR(rt5670->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5670->regmap, RT5670_VENDOR_ID2, &val); + if (val != RT5670_DEVICE_ID) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt5670/72\n", val); + return -ENODEV; + } + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_HP_L | RT5670_PWR_HP_R | + RT5670_PWR_VREF2, RT5670_PWR_VREF2); + msleep(100); + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + + ret = regmap_register_patch(rt5670->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5670->pdata.in2_diff) + regmap_update_bits(rt5670->regmap, RT5670_IN2, + RT5670_IN_DF2, RT5670_IN_DF2); + + if (i2c->irq) { + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, + RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT); + + } + + if (rt5670->pdata.jd_mode) { + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_MB, RT5670_PWR_MB); + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG2, + RT5670_PWR_JD1, RT5670_PWR_JD1); + regmap_update_bits(rt5670->regmap, RT5670_IRQ_CTRL1, + RT5670_JD1_1_EN_MASK, RT5670_JD1_1_EN); + regmap_update_bits(rt5670->regmap, RT5670_JD_CTRL3, + RT5670_JD_TRI_CBJ_SEL_MASK | + RT5670_JD_TRI_HPO_SEL_MASK, + RT5670_JD_CBJ_JD1_1 | RT5670_JD_HPO_JD1_1); + switch (rt5670->pdata.jd_mode) { + case 1: + regmap_update_bits(rt5670->regmap, RT5670_A_JD_CTRL1, + RT5670_JD1_MODE_MASK, + RT5670_JD1_MODE_0); + break; + case 2: + regmap_update_bits(rt5670->regmap, RT5670_A_JD_CTRL1, + RT5670_JD1_MODE_MASK, + RT5670_JD1_MODE_1); + break; + case 3: + regmap_update_bits(rt5670->regmap, RT5670_A_JD_CTRL1, + RT5670_JD1_MODE_MASK, + RT5670_JD1_MODE_2); + break; + default: + break; + } + } + + if (rt5670->pdata.dmic_en) { + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP2_PIN_MASK, + RT5670_GP2_PIN_DMIC1_SCL); + + switch (rt5670->pdata.dmic1_data_pin) { + case RT5670_DMIC_DATA_IN2P: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_1_DP_MASK, + RT5670_DMIC_1_DP_IN2P); + break; + + case RT5670_DMIC_DATA_GPIO6: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_1_DP_MASK, + RT5670_DMIC_1_DP_GPIO6); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP6_PIN_MASK, + RT5670_GP6_PIN_DMIC1_SDA); + break; + + case RT5670_DMIC_DATA_GPIO7: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_1_DP_MASK, + RT5670_DMIC_1_DP_GPIO7); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP7_PIN_MASK, + RT5670_GP7_PIN_DMIC1_SDA); + break; + + default: + break; + } + + switch (rt5670->pdata.dmic2_data_pin) { + case RT5670_DMIC_DATA_IN3N: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_2_DP_MASK, + RT5670_DMIC_2_DP_IN3N); + break; + + case RT5670_DMIC_DATA_GPIO8: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_2_DP_MASK, + RT5670_DMIC_2_DP_GPIO8); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP8_PIN_MASK, + RT5670_GP8_PIN_DMIC2_SDA); + break; + + default: + break; + } + + switch (rt5670->pdata.dmic3_data_pin) { + case RT5670_DMIC_DATA_GPIO5: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL2, + RT5670_DMIC_3_DP_MASK, + RT5670_DMIC_3_DP_GPIO5); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP5_PIN_MASK, + RT5670_GP5_PIN_DMIC3_SDA); + break; + + case RT5670_DMIC_DATA_GPIO9: + case RT5670_DMIC_DATA_GPIO10: + dev_err(&i2c->dev, + "Always use GPIO5 as DMIC3 data pin\n"); + break; + + default: + break; + } + + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670, + rt5670_dai, ARRAY_SIZE(rt5670_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt5670_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +static struct i2c_driver rt5670_i2c_driver = { + .driver = { + .name = "rt5670", + .owner = THIS_MODULE, + }, + .probe = rt5670_i2c_probe, + .remove = rt5670_i2c_remove, + .id_table = rt5670_i2c_id, +}; + +module_i2c_driver(rt5670_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5670 driver"); +MODULE_AUTHOR("Bard Liao <bardliao@realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h new file mode 100644 index 000000000000..a0b5c855b492 --- /dev/null +++ b/sound/soc/codecs/rt5670.h @@ -0,0 +1,2000 @@ +/* + * rt5670.h -- RT5670 ALSA SoC audio driver + * + * Copyright 2014 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 __RT5670_H__ +#define __RT5670_H__ + +#include <sound/rt5670.h> + +/* Info */ +#define RT5670_RESET 0x00 +#define RT5670_VENDOR_ID 0xfd +#define RT5670_VENDOR_ID1 0xfe +#define RT5670_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5670_HP_VOL 0x02 +#define RT5670_LOUT1 0x03 +/* I/O - Input */ +#define RT5670_CJ_CTRL1 0x0a +#define RT5670_CJ_CTRL2 0x0b +#define RT5670_CJ_CTRL3 0x0c +#define RT5670_IN2 0x0e +#define RT5670_INL1_INR1_VOL 0x0f +/* I/O - ADC/DAC/DMIC */ +#define RT5670_DAC1_DIG_VOL 0x19 +#define RT5670_DAC2_DIG_VOL 0x1a +#define RT5670_DAC_CTRL 0x1b +#define RT5670_STO1_ADC_DIG_VOL 0x1c +#define RT5670_MONO_ADC_DIG_VOL 0x1d +#define RT5670_ADC_BST_VOL1 0x1e +#define RT5670_STO2_ADC_DIG_VOL 0x1f +/* Mixer - D-D */ +#define RT5670_ADC_BST_VOL2 0x20 +#define RT5670_STO2_ADC_MIXER 0x26 +#define RT5670_STO1_ADC_MIXER 0x27 +#define RT5670_MONO_ADC_MIXER 0x28 +#define RT5670_AD_DA_MIXER 0x29 +#define RT5670_STO_DAC_MIXER 0x2a +#define RT5670_DD_MIXER 0x2b +#define RT5670_DIG_MIXER 0x2c +#define RT5670_DSP_PATH1 0x2d +#define RT5670_DSP_PATH2 0x2e +#define RT5670_DIG_INF1_DATA 0x2f +#define RT5670_DIG_INF2_DATA 0x30 +/* Mixer - PDM */ +#define RT5670_PDM_OUT_CTRL 0x31 +#define RT5670_PDM_DATA_CTRL1 0x32 +#define RT5670_PDM1_DATA_CTRL2 0x33 +#define RT5670_PDM1_DATA_CTRL3 0x34 +#define RT5670_PDM1_DATA_CTRL4 0x35 +#define RT5670_PDM2_DATA_CTRL2 0x36 +#define RT5670_PDM2_DATA_CTRL3 0x37 +#define RT5670_PDM2_DATA_CTRL4 0x38 +/* Mixer - ADC */ +#define RT5670_REC_L1_MIXER 0x3b +#define RT5670_REC_L2_MIXER 0x3c +#define RT5670_REC_R1_MIXER 0x3d +#define RT5670_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5670_HPO_MIXER 0x45 +#define RT5670_MONO_MIXER 0x4c +#define RT5670_OUT_L1_MIXER 0x4f +#define RT5670_OUT_R1_MIXER 0x52 +#define RT5670_LOUT_MIXER 0x53 +/* Power */ +#define RT5670_PWR_DIG1 0x61 +#define RT5670_PWR_DIG2 0x62 +#define RT5670_PWR_ANLG1 0x63 +#define RT5670_PWR_ANLG2 0x64 +#define RT5670_PWR_MIXER 0x65 +#define RT5670_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5670_PRIV_INDEX 0x6a +#define RT5670_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5670_I2S4_SDP 0x6f +#define RT5670_I2S1_SDP 0x70 +#define RT5670_I2S2_SDP 0x71 +#define RT5670_I2S3_SDP 0x72 +#define RT5670_ADDA_CLK1 0x73 +#define RT5670_ADDA_CLK2 0x74 +#define RT5670_DMIC_CTRL1 0x75 +#define RT5670_DMIC_CTRL2 0x76 +/* Format - TDM Control */ +#define RT5670_TDM_CTRL_1 0x77 +#define RT5670_TDM_CTRL_2 0x78 +#define RT5670_TDM_CTRL_3 0x79 + +/* Function - Analog */ +#define RT5670_DSP_CLK 0x7f +#define RT5670_GLB_CLK 0x80 +#define RT5670_PLL_CTRL1 0x81 +#define RT5670_PLL_CTRL2 0x82 +#define RT5670_ASRC_1 0x83 +#define RT5670_ASRC_2 0x84 +#define RT5670_ASRC_3 0x85 +#define RT5670_ASRC_4 0x86 +#define RT5670_ASRC_5 0x87 +#define RT5670_ASRC_7 0x89 +#define RT5670_ASRC_8 0x8a +#define RT5670_ASRC_9 0x8b +#define RT5670_ASRC_10 0x8c +#define RT5670_ASRC_11 0x8d +#define RT5670_DEPOP_M1 0x8e +#define RT5670_DEPOP_M2 0x8f +#define RT5670_DEPOP_M3 0x90 +#define RT5670_CHARGE_PUMP 0x91 +#define RT5670_MICBIAS 0x93 +#define RT5670_A_JD_CTRL1 0x94 +#define RT5670_A_JD_CTRL2 0x95 +#define RT5670_ASRC_12 0x97 +#define RT5670_ASRC_13 0x98 +#define RT5670_ASRC_14 0x99 +#define RT5670_VAD_CTRL1 0x9a +#define RT5670_VAD_CTRL2 0x9b +#define RT5670_VAD_CTRL3 0x9c +#define RT5670_VAD_CTRL4 0x9d +#define RT5670_VAD_CTRL5 0x9e +/* Function - Digital */ +#define RT5670_ADC_EQ_CTRL1 0xae +#define RT5670_ADC_EQ_CTRL2 0xaf +#define RT5670_EQ_CTRL1 0xb0 +#define RT5670_EQ_CTRL2 0xb1 +#define RT5670_ALC_DRC_CTRL1 0xb2 +#define RT5670_ALC_DRC_CTRL2 0xb3 +#define RT5670_ALC_CTRL_1 0xb4 +#define RT5670_ALC_CTRL_2 0xb5 +#define RT5670_ALC_CTRL_3 0xb6 +#define RT5670_ALC_CTRL_4 0xb7 +#define RT5670_JD_CTRL 0xbb +#define RT5670_IRQ_CTRL1 0xbd +#define RT5670_IRQ_CTRL2 0xbe +#define RT5670_INT_IRQ_ST 0xbf +#define RT5670_GPIO_CTRL1 0xc0 +#define RT5670_GPIO_CTRL2 0xc1 +#define RT5670_GPIO_CTRL3 0xc2 +#define RT5670_SCRABBLE_FUN 0xcd +#define RT5670_SCRABBLE_CTRL 0xce +#define RT5670_BASE_BACK 0xcf +#define RT5670_MP3_PLUS1 0xd0 +#define RT5670_MP3_PLUS2 0xd1 +#define RT5670_ADJ_HPF1 0xd3 +#define RT5670_ADJ_HPF2 0xd4 +#define RT5670_HP_CALIB_AMP_DET 0xd6 +#define RT5670_SV_ZCD1 0xd9 +#define RT5670_SV_ZCD2 0xda +#define RT5670_IL_CMD 0xdb +#define RT5670_IL_CMD2 0xdc +#define RT5670_IL_CMD3 0xdd +#define RT5670_DRC_HL_CTRL1 0xe6 +#define RT5670_DRC_HL_CTRL2 0xe7 +#define RT5670_ADC_MONO_HP_CTRL1 0xec +#define RT5670_ADC_MONO_HP_CTRL2 0xed +#define RT5670_ADC_STO2_HP_CTRL1 0xee +#define RT5670_ADC_STO2_HP_CTRL2 0xef +#define RT5670_JD_CTRL3 0xf8 +#define RT5670_JD_CTRL4 0xf9 +/* General Control */ +#define RT5670_DIG_MISC 0xfa +#define RT5670_GEN_CTRL2 0xfb +#define RT5670_GEN_CTRL3 0xfc + + +/* Index of Codec Private Register definition */ +#define RT5670_DIG_VOL 0x00 +#define RT5670_PR_ALC_CTRL_1 0x01 +#define RT5670_PR_ALC_CTRL_2 0x02 +#define RT5670_PR_ALC_CTRL_3 0x03 +#define RT5670_PR_ALC_CTRL_4 0x04 +#define RT5670_PR_ALC_CTRL_5 0x05 +#define RT5670_PR_ALC_CTRL_6 0x06 +#define RT5670_BIAS_CUR1 0x12 +#define RT5670_BIAS_CUR3 0x14 +#define RT5670_CLSD_INT_REG1 0x1c +#define RT5670_MAMP_INT_REG2 0x37 +#define RT5670_CHOP_DAC_ADC 0x3d +#define RT5670_MIXER_INT_REG 0x3f +#define RT5670_3D_SPK 0x63 +#define RT5670_WND_1 0x6c +#define RT5670_WND_2 0x6d +#define RT5670_WND_3 0x6e +#define RT5670_WND_4 0x6f +#define RT5670_WND_5 0x70 +#define RT5670_WND_8 0x73 +#define RT5670_DIP_SPK_INF 0x75 +#define RT5670_HP_DCC_INT1 0x77 +#define RT5670_EQ_BW_LOP 0xa0 +#define RT5670_EQ_GN_LOP 0xa1 +#define RT5670_EQ_FC_BP1 0xa2 +#define RT5670_EQ_BW_BP1 0xa3 +#define RT5670_EQ_GN_BP1 0xa4 +#define RT5670_EQ_FC_BP2 0xa5 +#define RT5670_EQ_BW_BP2 0xa6 +#define RT5670_EQ_GN_BP2 0xa7 +#define RT5670_EQ_FC_BP3 0xa8 +#define RT5670_EQ_BW_BP3 0xa9 +#define RT5670_EQ_GN_BP3 0xaa +#define RT5670_EQ_FC_BP4 0xab +#define RT5670_EQ_BW_BP4 0xac +#define RT5670_EQ_GN_BP4 0xad +#define RT5670_EQ_FC_HIP1 0xae +#define RT5670_EQ_GN_HIP1 0xaf +#define RT5670_EQ_FC_HIP2 0xb0 +#define RT5670_EQ_BW_HIP2 0xb1 +#define RT5670_EQ_GN_HIP2 0xb2 +#define RT5670_EQ_PRE_VOL 0xb3 +#define RT5670_EQ_PST_VOL 0xb4 + + +/* global definition */ +#define RT5670_L_MUTE (0x1 << 15) +#define RT5670_L_MUTE_SFT 15 +#define RT5670_VOL_L_MUTE (0x1 << 14) +#define RT5670_VOL_L_SFT 14 +#define RT5670_R_MUTE (0x1 << 7) +#define RT5670_R_MUTE_SFT 7 +#define RT5670_VOL_R_MUTE (0x1 << 6) +#define RT5670_VOL_R_SFT 6 +#define RT5670_L_VOL_MASK (0x3f << 8) +#define RT5670_L_VOL_SFT 8 +#define RT5670_R_VOL_MASK (0x3f) +#define RT5670_R_VOL_SFT 0 + +/* Combo Jack Control 1 (0x0a) */ +#define RT5670_CBJ_BST1_MASK (0xf << 12) +#define RT5670_CBJ_BST1_SFT (12) +#define RT5670_CBJ_JD_HP_EN (0x1 << 9) +#define RT5670_CBJ_JD_MIC_EN (0x1 << 8) +#define RT5670_CBJ_BST1_EN (0x1 << 2) + +/* Combo Jack Control 1 (0x0b) */ +#define RT5670_CBJ_MN_JD (0x1 << 12) +#define RT5670_CAPLESS_EN (0x1 << 11) +#define RT5670_CBJ_DET_MODE (0x1 << 7) + +/* IN2 Control (0x0e) */ +#define RT5670_BST_MASK1 (0xf<<12) +#define RT5670_BST_SFT1 12 +#define RT5670_BST_MASK2 (0xf<<8) +#define RT5670_BST_SFT2 8 +#define RT5670_IN_DF1 (0x1 << 7) +#define RT5670_IN_SFT1 7 +#define RT5670_IN_DF2 (0x1 << 6) +#define RT5670_IN_SFT2 6 + +/* INL and INR Volume Control (0x0f) */ +#define RT5670_INL_SEL_MASK (0x1 << 15) +#define RT5670_INL_SEL_SFT 15 +#define RT5670_INL_SEL_IN4P (0x0 << 15) +#define RT5670_INL_SEL_MONOP (0x1 << 15) +#define RT5670_INL_VOL_MASK (0x1f << 8) +#define RT5670_INL_VOL_SFT 8 +#define RT5670_INR_SEL_MASK (0x1 << 7) +#define RT5670_INR_SEL_SFT 7 +#define RT5670_INR_SEL_IN4N (0x0 << 7) +#define RT5670_INR_SEL_MONON (0x1 << 7) +#define RT5670_INR_VOL_MASK (0x1f) +#define RT5670_INR_VOL_SFT 0 + +/* Sidetone Control (0x18) */ +#define RT5670_ST_SEL_MASK (0x7 << 9) +#define RT5670_ST_SEL_SFT 9 +#define RT5670_M_ST_DACR2 (0x1 << 8) +#define RT5670_M_ST_DACR2_SFT 8 +#define RT5670_M_ST_DACL2 (0x1 << 7) +#define RT5670_M_ST_DACL2_SFT 7 +#define RT5670_ST_EN (0x1 << 6) +#define RT5670_ST_EN_SFT 6 + +/* DAC1 Digital Volume (0x19) */ +#define RT5670_DAC_L1_VOL_MASK (0xff << 8) +#define RT5670_DAC_L1_VOL_SFT 8 +#define RT5670_DAC_R1_VOL_MASK (0xff) +#define RT5670_DAC_R1_VOL_SFT 0 + +/* DAC2 Digital Volume (0x1a) */ +#define RT5670_DAC_L2_VOL_MASK (0xff << 8) +#define RT5670_DAC_L2_VOL_SFT 8 +#define RT5670_DAC_R2_VOL_MASK (0xff) +#define RT5670_DAC_R2_VOL_SFT 0 + +/* DAC2 Control (0x1b) */ +#define RT5670_M_DAC_L2_VOL (0x1 << 13) +#define RT5670_M_DAC_L2_VOL_SFT 13 +#define RT5670_M_DAC_R2_VOL (0x1 << 12) +#define RT5670_M_DAC_R2_VOL_SFT 12 +#define RT5670_DAC2_L_SEL_MASK (0x7 << 4) +#define RT5670_DAC2_L_SEL_SFT 4 +#define RT5670_DAC2_R_SEL_MASK (0x7 << 0) +#define RT5670_DAC2_R_SEL_SFT 0 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5670_ADC_L_VOL_MASK (0x7f << 8) +#define RT5670_ADC_L_VOL_SFT 8 +#define RT5670_ADC_R_VOL_MASK (0x7f) +#define RT5670_ADC_R_VOL_SFT 0 + +/* Mono ADC Digital Volume Control (0x1d) */ +#define RT5670_MONO_ADC_L_VOL_MASK (0x7f << 8) +#define RT5670_MONO_ADC_L_VOL_SFT 8 +#define RT5670_MONO_ADC_R_VOL_MASK (0x7f) +#define RT5670_MONO_ADC_R_VOL_SFT 0 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5670_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5670_STO1_ADC_L_BST_SFT 14 +#define RT5670_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5670_STO1_ADC_R_BST_SFT 12 +#define RT5670_STO1_ADC_COMP_MASK (0x3 << 10) +#define RT5670_STO1_ADC_COMP_SFT 10 +#define RT5670_STO2_ADC_L_BST_MASK (0x3 << 8) +#define RT5670_STO2_ADC_L_BST_SFT 8 +#define RT5670_STO2_ADC_R_BST_MASK (0x3 << 6) +#define RT5670_STO2_ADC_R_BST_SFT 6 +#define RT5670_STO2_ADC_COMP_MASK (0x3 << 4) +#define RT5670_STO2_ADC_COMP_SFT 4 + +/* Stereo2 ADC Mixer Control (0x26) */ +#define RT5670_STO2_ADC_SRC_MASK (0x1 << 15) +#define RT5670_STO2_ADC_SRC_SFT 15 + +/* Stereo ADC Mixer Control (0x26 0x27) */ +#define RT5670_M_ADC_L1 (0x1 << 14) +#define RT5670_M_ADC_L1_SFT 14 +#define RT5670_M_ADC_L2 (0x1 << 13) +#define RT5670_M_ADC_L2_SFT 13 +#define RT5670_ADC_1_SRC_MASK (0x1 << 12) +#define RT5670_ADC_1_SRC_SFT 12 +#define RT5670_ADC_1_SRC_ADC (0x1 << 12) +#define RT5670_ADC_1_SRC_DACMIX (0x0 << 12) +#define RT5670_ADC_2_SRC_MASK (0x1 << 11) +#define RT5670_ADC_2_SRC_SFT 11 +#define RT5670_ADC_SRC_MASK (0x1 << 10) +#define RT5670_ADC_SRC_SFT 10 +#define RT5670_DMIC_SRC_MASK (0x3 << 8) +#define RT5670_DMIC_SRC_SFT 8 +#define RT5670_M_ADC_R1 (0x1 << 6) +#define RT5670_M_ADC_R1_SFT 6 +#define RT5670_M_ADC_R2 (0x1 << 5) +#define RT5670_M_ADC_R2_SFT 5 +#define RT5670_DMIC3_SRC_MASK (0x1 << 1) +#define RT5670_DMIC3_SRC_SFT 0 + +/* Mono ADC Mixer Control (0x28) */ +#define RT5670_M_MONO_ADC_L1 (0x1 << 14) +#define RT5670_M_MONO_ADC_L1_SFT 14 +#define RT5670_M_MONO_ADC_L2 (0x1 << 13) +#define RT5670_M_MONO_ADC_L2_SFT 13 +#define RT5670_MONO_ADC_L1_SRC_MASK (0x1 << 12) +#define RT5670_MONO_ADC_L1_SRC_SFT 12 +#define RT5670_MONO_ADC_L1_SRC_DACMIXL (0x0 << 12) +#define RT5670_MONO_ADC_L1_SRC_ADCL (0x1 << 12) +#define RT5670_MONO_ADC_L2_SRC_MASK (0x1 << 11) +#define RT5670_MONO_ADC_L2_SRC_SFT 11 +#define RT5670_MONO_ADC_L_SRC_MASK (0x1 << 10) +#define RT5670_MONO_ADC_L_SRC_SFT 10 +#define RT5670_MONO_DMIC_L_SRC_MASK (0x3 << 8) +#define RT5670_MONO_DMIC_L_SRC_SFT 8 +#define RT5670_M_MONO_ADC_R1 (0x1 << 6) +#define RT5670_M_MONO_ADC_R1_SFT 6 +#define RT5670_M_MONO_ADC_R2 (0x1 << 5) +#define RT5670_M_MONO_ADC_R2_SFT 5 +#define RT5670_MONO_ADC_R1_SRC_MASK (0x1 << 4) +#define RT5670_MONO_ADC_R1_SRC_SFT 4 +#define RT5670_MONO_ADC_R1_SRC_ADCR (0x1 << 4) +#define RT5670_MONO_ADC_R1_SRC_DACMIXR (0x0 << 4) +#define RT5670_MONO_ADC_R2_SRC_MASK (0x1 << 3) +#define RT5670_MONO_ADC_R2_SRC_SFT 3 +#define RT5670_MONO_DMIC_R_SRC_MASK (0x3) +#define RT5670_MONO_DMIC_R_SRC_SFT 0 + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5670_M_ADCMIX_L (0x1 << 15) +#define RT5670_M_ADCMIX_L_SFT 15 +#define RT5670_M_DAC1_L (0x1 << 14) +#define RT5670_M_DAC1_L_SFT 14 +#define RT5670_DAC1_R_SEL_MASK (0x3 << 10) +#define RT5670_DAC1_R_SEL_SFT 10 +#define RT5670_DAC1_R_SEL_IF1 (0x0 << 10) +#define RT5670_DAC1_R_SEL_IF2 (0x1 << 10) +#define RT5670_DAC1_R_SEL_IF3 (0x2 << 10) +#define RT5670_DAC1_R_SEL_IF4 (0x3 << 10) +#define RT5670_DAC1_L_SEL_MASK (0x3 << 8) +#define RT5670_DAC1_L_SEL_SFT 8 +#define RT5670_DAC1_L_SEL_IF1 (0x0 << 8) +#define RT5670_DAC1_L_SEL_IF2 (0x1 << 8) +#define RT5670_DAC1_L_SEL_IF3 (0x2 << 8) +#define RT5670_DAC1_L_SEL_IF4 (0x3 << 8) +#define RT5670_M_ADCMIX_R (0x1 << 7) +#define RT5670_M_ADCMIX_R_SFT 7 +#define RT5670_M_DAC1_R (0x1 << 6) +#define RT5670_M_DAC1_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5670_M_DAC_L1 (0x1 << 14) +#define RT5670_M_DAC_L1_SFT 14 +#define RT5670_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5670_DAC_L1_STO_L_VOL_SFT 13 +#define RT5670_M_DAC_L2 (0x1 << 12) +#define RT5670_M_DAC_L2_SFT 12 +#define RT5670_DAC_L2_STO_L_VOL_MASK (0x1 << 11) +#define RT5670_DAC_L2_STO_L_VOL_SFT 11 +#define RT5670_M_DAC_R1_STO_L (0x1 << 9) +#define RT5670_M_DAC_R1_STO_L_SFT 9 +#define RT5670_DAC_R1_STO_L_VOL_MASK (0x1 << 8) +#define RT5670_DAC_R1_STO_L_VOL_SFT 8 +#define RT5670_M_DAC_R1 (0x1 << 6) +#define RT5670_M_DAC_R1_SFT 6 +#define RT5670_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5670_DAC_R1_STO_R_VOL_SFT 5 +#define RT5670_M_DAC_R2 (0x1 << 4) +#define RT5670_M_DAC_R2_SFT 4 +#define RT5670_DAC_R2_STO_R_VOL_MASK (0x1 << 3) +#define RT5670_DAC_R2_STO_R_VOL_SFT 3 +#define RT5670_M_DAC_L1_STO_R (0x1 << 1) +#define RT5670_M_DAC_L1_STO_R_SFT 1 +#define RT5670_DAC_L1_STO_R_VOL_MASK (0x1) +#define RT5670_DAC_L1_STO_R_VOL_SFT 0 + +/* Mono DAC Mixer Control (0x2b) */ +#define RT5670_M_DAC_L1_MONO_L (0x1 << 14) +#define RT5670_M_DAC_L1_MONO_L_SFT 14 +#define RT5670_DAC_L1_MONO_L_VOL_MASK (0x1 << 13) +#define RT5670_DAC_L1_MONO_L_VOL_SFT 13 +#define RT5670_M_DAC_L2_MONO_L (0x1 << 12) +#define RT5670_M_DAC_L2_MONO_L_SFT 12 +#define RT5670_DAC_L2_MONO_L_VOL_MASK (0x1 << 11) +#define RT5670_DAC_L2_MONO_L_VOL_SFT 11 +#define RT5670_M_DAC_R2_MONO_L (0x1 << 10) +#define RT5670_M_DAC_R2_MONO_L_SFT 10 +#define RT5670_DAC_R2_MONO_L_VOL_MASK (0x1 << 9) +#define RT5670_DAC_R2_MONO_L_VOL_SFT 9 +#define RT5670_M_DAC_R1_MONO_R (0x1 << 6) +#define RT5670_M_DAC_R1_MONO_R_SFT 6 +#define RT5670_DAC_R1_MONO_R_VOL_MASK (0x1 << 5) +#define RT5670_DAC_R1_MONO_R_VOL_SFT 5 +#define RT5670_M_DAC_R2_MONO_R (0x1 << 4) +#define RT5670_M_DAC_R2_MONO_R_SFT 4 +#define RT5670_DAC_R2_MONO_R_VOL_MASK (0x1 << 3) +#define RT5670_DAC_R2_MONO_R_VOL_SFT 3 +#define RT5670_M_DAC_L2_MONO_R (0x1 << 2) +#define RT5670_M_DAC_L2_MONO_R_SFT 2 +#define RT5670_DAC_L2_MONO_R_VOL_MASK (0x1 << 1) +#define RT5670_DAC_L2_MONO_R_VOL_SFT 1 + +/* Digital Mixer Control (0x2c) */ +#define RT5670_M_STO_L_DAC_L (0x1 << 15) +#define RT5670_M_STO_L_DAC_L_SFT 15 +#define RT5670_STO_L_DAC_L_VOL_MASK (0x1 << 14) +#define RT5670_STO_L_DAC_L_VOL_SFT 14 +#define RT5670_M_DAC_L2_DAC_L (0x1 << 13) +#define RT5670_M_DAC_L2_DAC_L_SFT 13 +#define RT5670_DAC_L2_DAC_L_VOL_MASK (0x1 << 12) +#define RT5670_DAC_L2_DAC_L_VOL_SFT 12 +#define RT5670_M_STO_R_DAC_R (0x1 << 11) +#define RT5670_M_STO_R_DAC_R_SFT 11 +#define RT5670_STO_R_DAC_R_VOL_MASK (0x1 << 10) +#define RT5670_STO_R_DAC_R_VOL_SFT 10 +#define RT5670_M_DAC_R2_DAC_R (0x1 << 9) +#define RT5670_M_DAC_R2_DAC_R_SFT 9 +#define RT5670_DAC_R2_DAC_R_VOL_MASK (0x1 << 8) +#define RT5670_DAC_R2_DAC_R_VOL_SFT 8 +#define RT5670_M_DAC_R2_DAC_L (0x1 << 7) +#define RT5670_M_DAC_R2_DAC_L_SFT 7 +#define RT5670_DAC_R2_DAC_L_VOL_MASK (0x1 << 6) +#define RT5670_DAC_R2_DAC_L_VOL_SFT 6 +#define RT5670_M_DAC_L2_DAC_R (0x1 << 5) +#define RT5670_M_DAC_L2_DAC_R_SFT 5 +#define RT5670_DAC_L2_DAC_R_VOL_MASK (0x1 << 4) +#define RT5670_DAC_L2_DAC_R_VOL_SFT 4 + +/* DSP Path Control 1 (0x2d) */ +#define RT5670_RXDP_SEL_MASK (0x7 << 13) +#define RT5670_RXDP_SEL_SFT 13 +#define RT5670_RXDP_SRC_MASK (0x3 << 11) +#define RT5670_RXDP_SRC_SFT 11 +#define RT5670_RXDP_SRC_NOR (0x0 << 11) +#define RT5670_RXDP_SRC_DIV2 (0x1 << 11) +#define RT5670_RXDP_SRC_DIV3 (0x2 << 11) +#define RT5670_TXDP_SRC_MASK (0x3 << 4) +#define RT5670_TXDP_SRC_SFT 4 +#define RT5670_TXDP_SRC_NOR (0x0 << 4) +#define RT5670_TXDP_SRC_DIV2 (0x1 << 4) +#define RT5670_TXDP_SRC_DIV3 (0x2 << 4) +#define RT5670_TXDP_SLOT_SEL_MASK (0x3 << 2) +#define RT5670_TXDP_SLOT_SEL_SFT 2 +#define RT5670_DSP_UL_SEL (0x1 << 1) +#define RT5670_DSP_UL_SFT 1 +#define RT5670_DSP_DL_SEL 0x1 +#define RT5670_DSP_DL_SFT 0 + +/* DSP Path Control 2 (0x2e) */ +#define RT5670_TXDP_L_VOL_MASK (0x7f << 8) +#define RT5670_TXDP_L_VOL_SFT 8 +#define RT5670_TXDP_R_VOL_MASK (0x7f) +#define RT5670_TXDP_R_VOL_SFT 0 + +/* Digital Interface Data Control (0x2f) */ +#define RT5670_IF1_ADC2_IN_SEL (0x1 << 15) +#define RT5670_IF1_ADC2_IN_SFT 15 +#define RT5670_IF2_ADC_IN_MASK (0x7 << 12) +#define RT5670_IF2_ADC_IN_SFT 12 +#define RT5670_IF2_DAC_SEL_MASK (0x3 << 10) +#define RT5670_IF2_DAC_SEL_SFT 10 +#define RT5670_IF2_ADC_SEL_MASK (0x3 << 8) +#define RT5670_IF2_ADC_SEL_SFT 8 + +/* Digital Interface Data Control (0x30) */ +#define RT5670_IF4_ADC_IN_MASK (0x3 << 4) +#define RT5670_IF4_ADC_IN_SFT 4 + +/* PDM Output Control (0x31) */ +#define RT5670_PDM1_L_MASK (0x1 << 15) +#define RT5670_PDM1_L_SFT 15 +#define RT5670_M_PDM1_L (0x1 << 14) +#define RT5670_M_PDM1_L_SFT 14 +#define RT5670_PDM1_R_MASK (0x1 << 13) +#define RT5670_PDM1_R_SFT 13 +#define RT5670_M_PDM1_R (0x1 << 12) +#define RT5670_M_PDM1_R_SFT 12 +#define RT5670_PDM2_L_MASK (0x1 << 11) +#define RT5670_PDM2_L_SFT 11 +#define RT5670_M_PDM2_L (0x1 << 10) +#define RT5670_M_PDM2_L_SFT 10 +#define RT5670_PDM2_R_MASK (0x1 << 9) +#define RT5670_PDM2_R_SFT 9 +#define RT5670_M_PDM2_R (0x1 << 8) +#define RT5670_M_PDM2_R_SFT 8 +#define RT5670_PDM2_BUSY (0x1 << 7) +#define RT5670_PDM1_BUSY (0x1 << 6) +#define RT5670_PDM_PATTERN (0x1 << 5) +#define RT5670_PDM_GAIN (0x1 << 4) +#define RT5670_PDM_DIV_MASK (0x3) + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5670_G_HP_L_RM_L_MASK (0x7 << 13) +#define RT5670_G_HP_L_RM_L_SFT 13 +#define RT5670_G_IN_L_RM_L_MASK (0x7 << 10) +#define RT5670_G_IN_L_RM_L_SFT 10 +#define RT5670_G_BST4_RM_L_MASK (0x7 << 7) +#define RT5670_G_BST4_RM_L_SFT 7 +#define RT5670_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5670_G_BST3_RM_L_SFT 4 +#define RT5670_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5670_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5670_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5670_G_BST1_RM_L_SFT 13 +#define RT5670_M_IN_L_RM_L (0x1 << 5) +#define RT5670_M_IN_L_RM_L_SFT 5 +#define RT5670_M_BST2_RM_L (0x1 << 3) +#define RT5670_M_BST2_RM_L_SFT 3 +#define RT5670_M_BST1_RM_L (0x1 << 1) +#define RT5670_M_BST1_RM_L_SFT 1 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5670_G_HP_R_RM_R_MASK (0x7 << 13) +#define RT5670_G_HP_R_RM_R_SFT 13 +#define RT5670_G_IN_R_RM_R_MASK (0x7 << 10) +#define RT5670_G_IN_R_RM_R_SFT 10 +#define RT5670_G_BST4_RM_R_MASK (0x7 << 7) +#define RT5670_G_BST4_RM_R_SFT 7 +#define RT5670_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5670_G_BST3_RM_R_SFT 4 +#define RT5670_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5670_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5670_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5670_G_BST1_RM_R_SFT 13 +#define RT5670_M_IN_R_RM_R (0x1 << 5) +#define RT5670_M_IN_R_RM_R_SFT 5 +#define RT5670_M_BST2_RM_R (0x1 << 3) +#define RT5670_M_BST2_RM_R_SFT 3 +#define RT5670_M_BST1_RM_R (0x1 << 1) +#define RT5670_M_BST1_RM_R_SFT 1 + +/* HPMIX Control (0x45) */ +#define RT5670_M_DAC2_HM (0x1 << 15) +#define RT5670_M_DAC2_HM_SFT 15 +#define RT5670_M_HPVOL_HM (0x1 << 14) +#define RT5670_M_HPVOL_HM_SFT 14 +#define RT5670_M_DAC1_HM (0x1 << 13) +#define RT5670_M_DAC1_HM_SFT 13 +#define RT5670_G_HPOMIX_MASK (0x1 << 12) +#define RT5670_G_HPOMIX_SFT 12 +#define RT5670_M_INR1_HMR (0x1 << 3) +#define RT5670_M_INR1_HMR_SFT 3 +#define RT5670_M_DACR1_HMR (0x1 << 2) +#define RT5670_M_DACR1_HMR_SFT 2 +#define RT5670_M_INL1_HML (0x1 << 1) +#define RT5670_M_INL1_HML_SFT 1 +#define RT5670_M_DACL1_HML (0x1) +#define RT5670_M_DACL1_HML_SFT 0 + +/* Mono Output Mixer Control (0x4c) */ +#define RT5670_M_DAC_R2_MA (0x1 << 15) +#define RT5670_M_DAC_R2_MA_SFT 15 +#define RT5670_M_DAC_L2_MA (0x1 << 14) +#define RT5670_M_DAC_L2_MA_SFT 14 +#define RT5670_M_OV_R_MM (0x1 << 13) +#define RT5670_M_OV_R_MM_SFT 13 +#define RT5670_M_OV_L_MM (0x1 << 12) +#define RT5670_M_OV_L_MM_SFT 12 +#define RT5670_G_MONOMIX_MASK (0x1 << 10) +#define RT5670_G_MONOMIX_SFT 10 +#define RT5670_M_DAC_R2_MM (0x1 << 9) +#define RT5670_M_DAC_R2_MM_SFT 9 +#define RT5670_M_DAC_L2_MM (0x1 << 8) +#define RT5670_M_DAC_L2_MM_SFT 8 +#define RT5670_M_BST4_MM (0x1 << 7) +#define RT5670_M_BST4_MM_SFT 7 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5670_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5670_G_BST3_OM_L_SFT 13 +#define RT5670_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5670_G_BST2_OM_L_SFT 10 +#define RT5670_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5670_G_BST1_OM_L_SFT 7 +#define RT5670_G_IN_L_OM_L_MASK (0x7 << 4) +#define RT5670_G_IN_L_OM_L_SFT 4 +#define RT5670_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5670_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5670_G_DAC_R2_OM_L_MASK (0x7 << 13) +#define RT5670_G_DAC_R2_OM_L_SFT 13 +#define RT5670_G_DAC_L2_OM_L_MASK (0x7 << 10) +#define RT5670_G_DAC_L2_OM_L_SFT 10 +#define RT5670_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5670_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5670_M_BST1_OM_L (0x1 << 5) +#define RT5670_M_BST1_OM_L_SFT 5 +#define RT5670_M_IN_L_OM_L (0x1 << 4) +#define RT5670_M_IN_L_OM_L_SFT 4 +#define RT5670_M_DAC_L2_OM_L (0x1 << 1) +#define RT5670_M_DAC_L2_OM_L_SFT 1 +#define RT5670_M_DAC_L1_OM_L (0x1) +#define RT5670_M_DAC_L1_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5670_G_BST4_OM_R_MASK (0x7 << 13) +#define RT5670_G_BST4_OM_R_SFT 13 +#define RT5670_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5670_G_BST2_OM_R_SFT 10 +#define RT5670_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5670_G_BST1_OM_R_SFT 7 +#define RT5670_G_IN_R_OM_R_MASK (0x7 << 4) +#define RT5670_G_IN_R_OM_R_SFT 4 +#define RT5670_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5670_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5670_G_DAC_L2_OM_R_MASK (0x7 << 13) +#define RT5670_G_DAC_L2_OM_R_SFT 13 +#define RT5670_G_DAC_R2_OM_R_MASK (0x7 << 10) +#define RT5670_G_DAC_R2_OM_R_SFT 10 +#define RT5670_G_DAC_R1_OM_R_MASK (0x7 << 7) +#define RT5670_G_DAC_R1_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5670_M_BST2_OM_R (0x1 << 6) +#define RT5670_M_BST2_OM_R_SFT 6 +#define RT5670_M_IN_R_OM_R (0x1 << 4) +#define RT5670_M_IN_R_OM_R_SFT 4 +#define RT5670_M_DAC_R2_OM_R (0x1 << 1) +#define RT5670_M_DAC_R2_OM_R_SFT 1 +#define RT5670_M_DAC_R1_OM_R (0x1) +#define RT5670_M_DAC_R1_OM_R_SFT 0 + +/* LOUT Mixer Control (0x53) */ +#define RT5670_M_DAC_L1_LM (0x1 << 15) +#define RT5670_M_DAC_L1_LM_SFT 15 +#define RT5670_M_DAC_R1_LM (0x1 << 14) +#define RT5670_M_DAC_R1_LM_SFT 14 +#define RT5670_M_OV_L_LM (0x1 << 13) +#define RT5670_M_OV_L_LM_SFT 13 +#define RT5670_M_OV_R_LM (0x1 << 12) +#define RT5670_M_OV_R_LM_SFT 12 +#define RT5670_G_LOUTMIX_MASK (0x1 << 11) +#define RT5670_G_LOUTMIX_SFT 11 + +/* Power Management for Digital 1 (0x61) */ +#define RT5670_PWR_I2S1 (0x1 << 15) +#define RT5670_PWR_I2S1_BIT 15 +#define RT5670_PWR_I2S2 (0x1 << 14) +#define RT5670_PWR_I2S2_BIT 14 +#define RT5670_PWR_DAC_L1 (0x1 << 12) +#define RT5670_PWR_DAC_L1_BIT 12 +#define RT5670_PWR_DAC_R1 (0x1 << 11) +#define RT5670_PWR_DAC_R1_BIT 11 +#define RT5670_PWR_DAC_L2 (0x1 << 7) +#define RT5670_PWR_DAC_L2_BIT 7 +#define RT5670_PWR_DAC_R2 (0x1 << 6) +#define RT5670_PWR_DAC_R2_BIT 6 +#define RT5670_PWR_ADC_L (0x1 << 2) +#define RT5670_PWR_ADC_L_BIT 2 +#define RT5670_PWR_ADC_R (0x1 << 1) +#define RT5670_PWR_ADC_R_BIT 1 +#define RT5670_PWR_CLS_D (0x1) +#define RT5670_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5670_PWR_ADC_S1F (0x1 << 15) +#define RT5670_PWR_ADC_S1F_BIT 15 +#define RT5670_PWR_ADC_MF_L (0x1 << 14) +#define RT5670_PWR_ADC_MF_L_BIT 14 +#define RT5670_PWR_ADC_MF_R (0x1 << 13) +#define RT5670_PWR_ADC_MF_R_BIT 13 +#define RT5670_PWR_I2S_DSP (0x1 << 12) +#define RT5670_PWR_I2S_DSP_BIT 12 +#define RT5670_PWR_DAC_S1F (0x1 << 11) +#define RT5670_PWR_DAC_S1F_BIT 11 +#define RT5670_PWR_DAC_MF_L (0x1 << 10) +#define RT5670_PWR_DAC_MF_L_BIT 10 +#define RT5670_PWR_DAC_MF_R (0x1 << 9) +#define RT5670_PWR_DAC_MF_R_BIT 9 +#define RT5670_PWR_ADC_S2F (0x1 << 8) +#define RT5670_PWR_ADC_S2F_BIT 8 +#define RT5670_PWR_PDM1 (0x1 << 7) +#define RT5670_PWR_PDM1_BIT 7 +#define RT5670_PWR_PDM2 (0x1 << 6) +#define RT5670_PWR_PDM2_BIT 6 + +/* Power Management for Analog 1 (0x63) */ +#define RT5670_PWR_VREF1 (0x1 << 15) +#define RT5670_PWR_VREF1_BIT 15 +#define RT5670_PWR_FV1 (0x1 << 14) +#define RT5670_PWR_FV1_BIT 14 +#define RT5670_PWR_MB (0x1 << 13) +#define RT5670_PWR_MB_BIT 13 +#define RT5670_PWR_LM (0x1 << 12) +#define RT5670_PWR_LM_BIT 12 +#define RT5670_PWR_BG (0x1 << 11) +#define RT5670_PWR_BG_BIT 11 +#define RT5670_PWR_HP_L (0x1 << 7) +#define RT5670_PWR_HP_L_BIT 7 +#define RT5670_PWR_HP_R (0x1 << 6) +#define RT5670_PWR_HP_R_BIT 6 +#define RT5670_PWR_HA (0x1 << 5) +#define RT5670_PWR_HA_BIT 5 +#define RT5670_PWR_VREF2 (0x1 << 4) +#define RT5670_PWR_VREF2_BIT 4 +#define RT5670_PWR_FV2 (0x1 << 3) +#define RT5670_PWR_FV2_BIT 3 +#define RT5670_LDO_SEL_MASK (0x3) +#define RT5670_LDO_SEL_SFT 0 + +/* Power Management for Analog 2 (0x64) */ +#define RT5670_PWR_BST1 (0x1 << 15) +#define RT5670_PWR_BST1_BIT 15 +#define RT5670_PWR_BST2 (0x1 << 13) +#define RT5670_PWR_BST2_BIT 13 +#define RT5670_PWR_MB1 (0x1 << 11) +#define RT5670_PWR_MB1_BIT 11 +#define RT5670_PWR_MB2 (0x1 << 10) +#define RT5670_PWR_MB2_BIT 10 +#define RT5670_PWR_PLL (0x1 << 9) +#define RT5670_PWR_PLL_BIT 9 +#define RT5670_PWR_BST1_P (0x1 << 6) +#define RT5670_PWR_BST1_P_BIT 6 +#define RT5670_PWR_BST2_P (0x1 << 4) +#define RT5670_PWR_BST2_P_BIT 4 +#define RT5670_PWR_JD1 (0x1 << 2) +#define RT5670_PWR_JD1_BIT 2 +#define RT5670_PWR_JD (0x1 << 1) +#define RT5670_PWR_JD_BIT 1 + +/* Power Management for Mixer (0x65) */ +#define RT5670_PWR_OM_L (0x1 << 15) +#define RT5670_PWR_OM_L_BIT 15 +#define RT5670_PWR_OM_R (0x1 << 14) +#define RT5670_PWR_OM_R_BIT 14 +#define RT5670_PWR_RM_L (0x1 << 11) +#define RT5670_PWR_RM_L_BIT 11 +#define RT5670_PWR_RM_R (0x1 << 10) +#define RT5670_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5670_PWR_HV_L (0x1 << 11) +#define RT5670_PWR_HV_L_BIT 11 +#define RT5670_PWR_HV_R (0x1 << 10) +#define RT5670_PWR_HV_R_BIT 10 +#define RT5670_PWR_IN_L (0x1 << 9) +#define RT5670_PWR_IN_L_BIT 9 +#define RT5670_PWR_IN_R (0x1 << 8) +#define RT5670_PWR_IN_R_BIT 8 +#define RT5670_PWR_MIC_DET (0x1 << 5) +#define RT5670_PWR_MIC_DET_BIT 5 + +/* I2S1/2/3 Audio Serial Data Port Control (0x70 0x71 0x72) */ +#define RT5670_I2S_MS_MASK (0x1 << 15) +#define RT5670_I2S_MS_SFT 15 +#define RT5670_I2S_MS_M (0x0 << 15) +#define RT5670_I2S_MS_S (0x1 << 15) +#define RT5670_I2S_IF_MASK (0x7 << 12) +#define RT5670_I2S_IF_SFT 12 +#define RT5670_I2S_O_CP_MASK (0x3 << 10) +#define RT5670_I2S_O_CP_SFT 10 +#define RT5670_I2S_O_CP_OFF (0x0 << 10) +#define RT5670_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5670_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5670_I2S_I_CP_MASK (0x3 << 8) +#define RT5670_I2S_I_CP_SFT 8 +#define RT5670_I2S_I_CP_OFF (0x0 << 8) +#define RT5670_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5670_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5670_I2S_BP_MASK (0x1 << 7) +#define RT5670_I2S_BP_SFT 7 +#define RT5670_I2S_BP_NOR (0x0 << 7) +#define RT5670_I2S_BP_INV (0x1 << 7) +#define RT5670_I2S_DL_MASK (0x3 << 2) +#define RT5670_I2S_DL_SFT 2 +#define RT5670_I2S_DL_16 (0x0 << 2) +#define RT5670_I2S_DL_20 (0x1 << 2) +#define RT5670_I2S_DL_24 (0x2 << 2) +#define RT5670_I2S_DL_8 (0x3 << 2) +#define RT5670_I2S_DF_MASK (0x3) +#define RT5670_I2S_DF_SFT 0 +#define RT5670_I2S_DF_I2S (0x0) +#define RT5670_I2S_DF_LEFT (0x1) +#define RT5670_I2S_DF_PCM_A (0x2) +#define RT5670_I2S_DF_PCM_B (0x3) + +/* I2S2 Audio Serial Data Port Control (0x71) */ +#define RT5670_I2S2_SDI_MASK (0x1 << 6) +#define RT5670_I2S2_SDI_SFT 6 +#define RT5670_I2S2_SDI_I2S1 (0x0 << 6) +#define RT5670_I2S2_SDI_I2S2 (0x1 << 6) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5670_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5670_I2S_BCLK_MS1_SFT 15 +#define RT5670_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5670_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5670_I2S_PD1_MASK (0x7 << 12) +#define RT5670_I2S_PD1_SFT 12 +#define RT5670_I2S_PD1_1 (0x0 << 12) +#define RT5670_I2S_PD1_2 (0x1 << 12) +#define RT5670_I2S_PD1_3 (0x2 << 12) +#define RT5670_I2S_PD1_4 (0x3 << 12) +#define RT5670_I2S_PD1_6 (0x4 << 12) +#define RT5670_I2S_PD1_8 (0x5 << 12) +#define RT5670_I2S_PD1_12 (0x6 << 12) +#define RT5670_I2S_PD1_16 (0x7 << 12) +#define RT5670_I2S_BCLK_MS2_MASK (0x1 << 11) +#define RT5670_I2S_BCLK_MS2_SFT 11 +#define RT5670_I2S_BCLK_MS2_32 (0x0 << 11) +#define RT5670_I2S_BCLK_MS2_64 (0x1 << 11) +#define RT5670_I2S_PD2_MASK (0x7 << 8) +#define RT5670_I2S_PD2_SFT 8 +#define RT5670_I2S_PD2_1 (0x0 << 8) +#define RT5670_I2S_PD2_2 (0x1 << 8) +#define RT5670_I2S_PD2_3 (0x2 << 8) +#define RT5670_I2S_PD2_4 (0x3 << 8) +#define RT5670_I2S_PD2_6 (0x4 << 8) +#define RT5670_I2S_PD2_8 (0x5 << 8) +#define RT5670_I2S_PD2_12 (0x6 << 8) +#define RT5670_I2S_PD2_16 (0x7 << 8) +#define RT5670_I2S_BCLK_MS3_MASK (0x1 << 7) +#define RT5670_I2S_BCLK_MS3_SFT 7 +#define RT5670_I2S_BCLK_MS3_32 (0x0 << 7) +#define RT5670_I2S_BCLK_MS3_64 (0x1 << 7) +#define RT5670_I2S_PD3_MASK (0x7 << 4) +#define RT5670_I2S_PD3_SFT 4 +#define RT5670_I2S_PD3_1 (0x0 << 4) +#define RT5670_I2S_PD3_2 (0x1 << 4) +#define RT5670_I2S_PD3_3 (0x2 << 4) +#define RT5670_I2S_PD3_4 (0x3 << 4) +#define RT5670_I2S_PD3_6 (0x4 << 4) +#define RT5670_I2S_PD3_8 (0x5 << 4) +#define RT5670_I2S_PD3_12 (0x6 << 4) +#define RT5670_I2S_PD3_16 (0x7 << 4) +#define RT5670_DAC_OSR_MASK (0x3 << 2) +#define RT5670_DAC_OSR_SFT 2 +#define RT5670_DAC_OSR_128 (0x0 << 2) +#define RT5670_DAC_OSR_64 (0x1 << 2) +#define RT5670_DAC_OSR_32 (0x2 << 2) +#define RT5670_DAC_OSR_16 (0x3 << 2) +#define RT5670_ADC_OSR_MASK (0x3) +#define RT5670_ADC_OSR_SFT 0 +#define RT5670_ADC_OSR_128 (0x0) +#define RT5670_ADC_OSR_64 (0x1) +#define RT5670_ADC_OSR_32 (0x2) +#define RT5670_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5670_DAC_L_OSR_MASK (0x3 << 14) +#define RT5670_DAC_L_OSR_SFT 14 +#define RT5670_DAC_L_OSR_128 (0x0 << 14) +#define RT5670_DAC_L_OSR_64 (0x1 << 14) +#define RT5670_DAC_L_OSR_32 (0x2 << 14) +#define RT5670_DAC_L_OSR_16 (0x3 << 14) +#define RT5670_ADC_R_OSR_MASK (0x3 << 12) +#define RT5670_ADC_R_OSR_SFT 12 +#define RT5670_ADC_R_OSR_128 (0x0 << 12) +#define RT5670_ADC_R_OSR_64 (0x1 << 12) +#define RT5670_ADC_R_OSR_32 (0x2 << 12) +#define RT5670_ADC_R_OSR_16 (0x3 << 12) +#define RT5670_DAHPF_EN (0x1 << 11) +#define RT5670_DAHPF_EN_SFT 11 +#define RT5670_ADHPF_EN (0x1 << 10) +#define RT5670_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5670_DMIC_1_EN_MASK (0x1 << 15) +#define RT5670_DMIC_1_EN_SFT 15 +#define RT5670_DMIC_1_DIS (0x0 << 15) +#define RT5670_DMIC_1_EN (0x1 << 15) +#define RT5670_DMIC_2_EN_MASK (0x1 << 14) +#define RT5670_DMIC_2_EN_SFT 14 +#define RT5670_DMIC_2_DIS (0x0 << 14) +#define RT5670_DMIC_2_EN (0x1 << 14) +#define RT5670_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5670_DMIC_1L_LH_SFT 13 +#define RT5670_DMIC_1L_LH_FALLING (0x0 << 13) +#define RT5670_DMIC_1L_LH_RISING (0x1 << 13) +#define RT5670_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5670_DMIC_1R_LH_SFT 12 +#define RT5670_DMIC_1R_LH_FALLING (0x0 << 12) +#define RT5670_DMIC_1R_LH_RISING (0x1 << 12) +#define RT5670_DMIC_2_DP_MASK (0x1 << 10) +#define RT5670_DMIC_2_DP_SFT 10 +#define RT5670_DMIC_2_DP_GPIO8 (0x0 << 10) +#define RT5670_DMIC_2_DP_IN3N (0x1 << 10) +#define RT5670_DMIC_2L_LH_MASK (0x1 << 9) +#define RT5670_DMIC_2L_LH_SFT 9 +#define RT5670_DMIC_2L_LH_FALLING (0x0 << 9) +#define RT5670_DMIC_2L_LH_RISING (0x1 << 9) +#define RT5670_DMIC_2R_LH_MASK (0x1 << 8) +#define RT5670_DMIC_2R_LH_SFT 8 +#define RT5670_DMIC_2R_LH_FALLING (0x0 << 8) +#define RT5670_DMIC_2R_LH_RISING (0x1 << 8) +#define RT5670_DMIC_CLK_MASK (0x7 << 5) +#define RT5670_DMIC_CLK_SFT 5 +#define RT5670_DMIC_3_EN_MASK (0x1 << 4) +#define RT5670_DMIC_3_EN_SFT 4 +#define RT5670_DMIC_3_DIS (0x0 << 4) +#define RT5670_DMIC_3_EN (0x1 << 4) +#define RT5670_DMIC_1_DP_MASK (0x3 << 0) +#define RT5670_DMIC_1_DP_SFT 0 +#define RT5670_DMIC_1_DP_GPIO6 (0x0 << 0) +#define RT5670_DMIC_1_DP_IN2P (0x1 << 0) +#define RT5670_DMIC_1_DP_GPIO7 (0x2 << 0) + +/* Digital Microphone Control2 (0x76) */ +#define RT5670_DMIC_3_DP_MASK (0x3 << 6) +#define RT5670_DMIC_3_DP_SFT 6 +#define RT5670_DMIC_3_DP_GPIO9 (0x0 << 6) +#define RT5670_DMIC_3_DP_GPIO10 (0x1 << 6) +#define RT5670_DMIC_3_DP_GPIO5 (0x2 << 6) + +/* Global Clock Control (0x80) */ +#define RT5670_SCLK_SRC_MASK (0x3 << 14) +#define RT5670_SCLK_SRC_SFT 14 +#define RT5670_SCLK_SRC_MCLK (0x0 << 14) +#define RT5670_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5670_SCLK_SRC_RCCLK (0x2 << 14) /* 15MHz */ +#define RT5670_PLL1_SRC_MASK (0x3 << 12) +#define RT5670_PLL1_SRC_SFT 12 +#define RT5670_PLL1_SRC_MCLK (0x0 << 12) +#define RT5670_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5670_PLL1_SRC_BCLK2 (0x2 << 12) +#define RT5670_PLL1_SRC_BCLK3 (0x3 << 12) +#define RT5670_PLL1_PD_MASK (0x1 << 3) +#define RT5670_PLL1_PD_SFT 3 +#define RT5670_PLL1_PD_1 (0x0 << 3) +#define RT5670_PLL1_PD_2 (0x1 << 3) + +#define RT5670_PLL_INP_MAX 40000000 +#define RT5670_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5670_PLL_N_MAX 0x1ff +#define RT5670_PLL_N_MASK (RT5670_PLL_N_MAX << 7) +#define RT5670_PLL_N_SFT 7 +#define RT5670_PLL_K_MAX 0x1f +#define RT5670_PLL_K_MASK (RT5670_PLL_K_MAX) +#define RT5670_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5670_PLL_M_MAX 0xf +#define RT5670_PLL_M_MASK (RT5670_PLL_M_MAX << 12) +#define RT5670_PLL_M_SFT 12 +#define RT5670_PLL_M_BP (0x1 << 11) +#define RT5670_PLL_M_BP_SFT 11 + +/* ASRC Control 1 (0x83) */ +#define RT5670_STO_T_MASK (0x1 << 15) +#define RT5670_STO_T_SFT 15 +#define RT5670_STO_T_SCLK (0x0 << 15) +#define RT5670_STO_T_LRCK1 (0x1 << 15) +#define RT5670_M1_T_MASK (0x1 << 14) +#define RT5670_M1_T_SFT 14 +#define RT5670_M1_T_I2S2 (0x0 << 14) +#define RT5670_M1_T_I2S2_D3 (0x1 << 14) +#define RT5670_I2S2_F_MASK (0x1 << 12) +#define RT5670_I2S2_F_SFT 12 +#define RT5670_I2S2_F_I2S2_D2 (0x0 << 12) +#define RT5670_I2S2_F_I2S1_TCLK (0x1 << 12) +#define RT5670_DMIC_1_M_MASK (0x1 << 9) +#define RT5670_DMIC_1_M_SFT 9 +#define RT5670_DMIC_1_M_NOR (0x0 << 9) +#define RT5670_DMIC_1_M_ASYN (0x1 << 9) +#define RT5670_DMIC_2_M_MASK (0x1 << 8) +#define RT5670_DMIC_2_M_SFT 8 +#define RT5670_DMIC_2_M_NOR (0x0 << 8) +#define RT5670_DMIC_2_M_ASYN (0x1 << 8) + +/* ASRC Control 2 (0x84) */ +#define RT5670_MDA_L_M_MASK (0x1 << 15) +#define RT5670_MDA_L_M_SFT 15 +#define RT5670_MDA_L_M_NOR (0x0 << 15) +#define RT5670_MDA_L_M_ASYN (0x1 << 15) +#define RT5670_MDA_R_M_MASK (0x1 << 14) +#define RT5670_MDA_R_M_SFT 14 +#define RT5670_MDA_R_M_NOR (0x0 << 14) +#define RT5670_MDA_R_M_ASYN (0x1 << 14) +#define RT5670_MAD_L_M_MASK (0x1 << 13) +#define RT5670_MAD_L_M_SFT 13 +#define RT5670_MAD_L_M_NOR (0x0 << 13) +#define RT5670_MAD_L_M_ASYN (0x1 << 13) +#define RT5670_MAD_R_M_MASK (0x1 << 12) +#define RT5670_MAD_R_M_SFT 12 +#define RT5670_MAD_R_M_NOR (0x0 << 12) +#define RT5670_MAD_R_M_ASYN (0x1 << 12) +#define RT5670_ADC_M_MASK (0x1 << 11) +#define RT5670_ADC_M_SFT 11 +#define RT5670_ADC_M_NOR (0x0 << 11) +#define RT5670_ADC_M_ASYN (0x1 << 11) +#define RT5670_STO_DAC_M_MASK (0x1 << 5) +#define RT5670_STO_DAC_M_SFT 5 +#define RT5670_STO_DAC_M_NOR (0x0 << 5) +#define RT5670_STO_DAC_M_ASYN (0x1 << 5) +#define RT5670_I2S1_R_D_MASK (0x1 << 4) +#define RT5670_I2S1_R_D_SFT 4 +#define RT5670_I2S1_R_D_DIS (0x0 << 4) +#define RT5670_I2S1_R_D_EN (0x1 << 4) +#define RT5670_I2S2_R_D_MASK (0x1 << 3) +#define RT5670_I2S2_R_D_SFT 3 +#define RT5670_I2S2_R_D_DIS (0x0 << 3) +#define RT5670_I2S2_R_D_EN (0x1 << 3) +#define RT5670_PRE_SCLK_MASK (0x3) +#define RT5670_PRE_SCLK_SFT 0 +#define RT5670_PRE_SCLK_512 (0x0) +#define RT5670_PRE_SCLK_1024 (0x1) +#define RT5670_PRE_SCLK_2048 (0x2) + +/* ASRC Control 3 (0x85) */ +#define RT5670_I2S1_RATE_MASK (0xf << 12) +#define RT5670_I2S1_RATE_SFT 12 +#define RT5670_I2S2_RATE_MASK (0xf << 8) +#define RT5670_I2S2_RATE_SFT 8 + +/* ASRC Control 4 (0x89) */ +#define RT5670_I2S1_PD_MASK (0x7 << 12) +#define RT5670_I2S1_PD_SFT 12 +#define RT5670_I2S2_PD_MASK (0x7 << 8) +#define RT5670_I2S2_PD_SFT 8 + +/* HPOUT Over Current Detection (0x8b) */ +#define RT5670_HP_OVCD_MASK (0x1 << 10) +#define RT5670_HP_OVCD_SFT 10 +#define RT5670_HP_OVCD_DIS (0x0 << 10) +#define RT5670_HP_OVCD_EN (0x1 << 10) +#define RT5670_HP_OC_TH_MASK (0x3 << 8) +#define RT5670_HP_OC_TH_SFT 8 +#define RT5670_HP_OC_TH_90 (0x0 << 8) +#define RT5670_HP_OC_TH_105 (0x1 << 8) +#define RT5670_HP_OC_TH_120 (0x2 << 8) +#define RT5670_HP_OC_TH_135 (0x3 << 8) + +/* Class D Over Current Control (0x8c) */ +#define RT5670_CLSD_OC_MASK (0x1 << 9) +#define RT5670_CLSD_OC_SFT 9 +#define RT5670_CLSD_OC_PU (0x0 << 9) +#define RT5670_CLSD_OC_PD (0x1 << 9) +#define RT5670_AUTO_PD_MASK (0x1 << 8) +#define RT5670_AUTO_PD_SFT 8 +#define RT5670_AUTO_PD_DIS (0x0 << 8) +#define RT5670_AUTO_PD_EN (0x1 << 8) +#define RT5670_CLSD_OC_TH_MASK (0x3f) +#define RT5670_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5670_CLSD_RATIO_MASK (0xf << 12) +#define RT5670_CLSD_RATIO_SFT 12 +#define RT5670_CLSD_OM_MASK (0x1 << 11) +#define RT5670_CLSD_OM_SFT 11 +#define RT5670_CLSD_OM_MONO (0x0 << 11) +#define RT5670_CLSD_OM_STO (0x1 << 11) +#define RT5670_CLSD_SCH_MASK (0x1 << 10) +#define RT5670_CLSD_SCH_SFT 10 +#define RT5670_CLSD_SCH_L (0x0 << 10) +#define RT5670_CLSD_SCH_S (0x1 << 10) + +/* Depop Mode Control 1 (0x8e) */ +#define RT5670_SMT_TRIG_MASK (0x1 << 15) +#define RT5670_SMT_TRIG_SFT 15 +#define RT5670_SMT_TRIG_DIS (0x0 << 15) +#define RT5670_SMT_TRIG_EN (0x1 << 15) +#define RT5670_HP_L_SMT_MASK (0x1 << 9) +#define RT5670_HP_L_SMT_SFT 9 +#define RT5670_HP_L_SMT_DIS (0x0 << 9) +#define RT5670_HP_L_SMT_EN (0x1 << 9) +#define RT5670_HP_R_SMT_MASK (0x1 << 8) +#define RT5670_HP_R_SMT_SFT 8 +#define RT5670_HP_R_SMT_DIS (0x0 << 8) +#define RT5670_HP_R_SMT_EN (0x1 << 8) +#define RT5670_HP_CD_PD_MASK (0x1 << 7) +#define RT5670_HP_CD_PD_SFT 7 +#define RT5670_HP_CD_PD_DIS (0x0 << 7) +#define RT5670_HP_CD_PD_EN (0x1 << 7) +#define RT5670_RSTN_MASK (0x1 << 6) +#define RT5670_RSTN_SFT 6 +#define RT5670_RSTN_DIS (0x0 << 6) +#define RT5670_RSTN_EN (0x1 << 6) +#define RT5670_RSTP_MASK (0x1 << 5) +#define RT5670_RSTP_SFT 5 +#define RT5670_RSTP_DIS (0x0 << 5) +#define RT5670_RSTP_EN (0x1 << 5) +#define RT5670_HP_CO_MASK (0x1 << 4) +#define RT5670_HP_CO_SFT 4 +#define RT5670_HP_CO_DIS (0x0 << 4) +#define RT5670_HP_CO_EN (0x1 << 4) +#define RT5670_HP_CP_MASK (0x1 << 3) +#define RT5670_HP_CP_SFT 3 +#define RT5670_HP_CP_PD (0x0 << 3) +#define RT5670_HP_CP_PU (0x1 << 3) +#define RT5670_HP_SG_MASK (0x1 << 2) +#define RT5670_HP_SG_SFT 2 +#define RT5670_HP_SG_DIS (0x0 << 2) +#define RT5670_HP_SG_EN (0x1 << 2) +#define RT5670_HP_DP_MASK (0x1 << 1) +#define RT5670_HP_DP_SFT 1 +#define RT5670_HP_DP_PD (0x0 << 1) +#define RT5670_HP_DP_PU (0x1 << 1) +#define RT5670_HP_CB_MASK (0x1) +#define RT5670_HP_CB_SFT 0 +#define RT5670_HP_CB_PD (0x0) +#define RT5670_HP_CB_PU (0x1) + +/* Depop Mode Control 2 (0x8f) */ +#define RT5670_DEPOP_MASK (0x1 << 13) +#define RT5670_DEPOP_SFT 13 +#define RT5670_DEPOP_AUTO (0x0 << 13) +#define RT5670_DEPOP_MAN (0x1 << 13) +#define RT5670_RAMP_MASK (0x1 << 12) +#define RT5670_RAMP_SFT 12 +#define RT5670_RAMP_DIS (0x0 << 12) +#define RT5670_RAMP_EN (0x1 << 12) +#define RT5670_BPS_MASK (0x1 << 11) +#define RT5670_BPS_SFT 11 +#define RT5670_BPS_DIS (0x0 << 11) +#define RT5670_BPS_EN (0x1 << 11) +#define RT5670_FAST_UPDN_MASK (0x1 << 10) +#define RT5670_FAST_UPDN_SFT 10 +#define RT5670_FAST_UPDN_DIS (0x0 << 10) +#define RT5670_FAST_UPDN_EN (0x1 << 10) +#define RT5670_MRES_MASK (0x3 << 8) +#define RT5670_MRES_SFT 8 +#define RT5670_MRES_15MO (0x0 << 8) +#define RT5670_MRES_25MO (0x1 << 8) +#define RT5670_MRES_35MO (0x2 << 8) +#define RT5670_MRES_45MO (0x3 << 8) +#define RT5670_VLO_MASK (0x1 << 7) +#define RT5670_VLO_SFT 7 +#define RT5670_VLO_3V (0x0 << 7) +#define RT5670_VLO_32V (0x1 << 7) +#define RT5670_DIG_DP_MASK (0x1 << 6) +#define RT5670_DIG_DP_SFT 6 +#define RT5670_DIG_DP_DIS (0x0 << 6) +#define RT5670_DIG_DP_EN (0x1 << 6) +#define RT5670_DP_TH_MASK (0x3 << 4) +#define RT5670_DP_TH_SFT 4 + +/* Depop Mode Control 3 (0x90) */ +#define RT5670_CP_SYS_MASK (0x7 << 12) +#define RT5670_CP_SYS_SFT 12 +#define RT5670_CP_FQ1_MASK (0x7 << 8) +#define RT5670_CP_FQ1_SFT 8 +#define RT5670_CP_FQ2_MASK (0x7 << 4) +#define RT5670_CP_FQ2_SFT 4 +#define RT5670_CP_FQ3_MASK (0x7) +#define RT5670_CP_FQ3_SFT 0 +#define RT5670_CP_FQ_1_5_KHZ 0 +#define RT5670_CP_FQ_3_KHZ 1 +#define RT5670_CP_FQ_6_KHZ 2 +#define RT5670_CP_FQ_12_KHZ 3 +#define RT5670_CP_FQ_24_KHZ 4 +#define RT5670_CP_FQ_48_KHZ 5 +#define RT5670_CP_FQ_96_KHZ 6 +#define RT5670_CP_FQ_192_KHZ 7 + +/* HPOUT charge pump (0x91) */ +#define RT5670_OSW_L_MASK (0x1 << 11) +#define RT5670_OSW_L_SFT 11 +#define RT5670_OSW_L_DIS (0x0 << 11) +#define RT5670_OSW_L_EN (0x1 << 11) +#define RT5670_OSW_R_MASK (0x1 << 10) +#define RT5670_OSW_R_SFT 10 +#define RT5670_OSW_R_DIS (0x0 << 10) +#define RT5670_OSW_R_EN (0x1 << 10) +#define RT5670_PM_HP_MASK (0x3 << 8) +#define RT5670_PM_HP_SFT 8 +#define RT5670_PM_HP_LV (0x0 << 8) +#define RT5670_PM_HP_MV (0x1 << 8) +#define RT5670_PM_HP_HV (0x2 << 8) +#define RT5670_IB_HP_MASK (0x3 << 6) +#define RT5670_IB_HP_SFT 6 +#define RT5670_IB_HP_125IL (0x0 << 6) +#define RT5670_IB_HP_25IL (0x1 << 6) +#define RT5670_IB_HP_5IL (0x2 << 6) +#define RT5670_IB_HP_1IL (0x3 << 6) + +/* PV detection and SPK gain control (0x92) */ +#define RT5670_PVDD_DET_MASK (0x1 << 15) +#define RT5670_PVDD_DET_SFT 15 +#define RT5670_PVDD_DET_DIS (0x0 << 15) +#define RT5670_PVDD_DET_EN (0x1 << 15) +#define RT5670_SPK_AG_MASK (0x1 << 14) +#define RT5670_SPK_AG_SFT 14 +#define RT5670_SPK_AG_DIS (0x0 << 14) +#define RT5670_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5670_MIC1_BS_MASK (0x1 << 15) +#define RT5670_MIC1_BS_SFT 15 +#define RT5670_MIC1_BS_9AV (0x0 << 15) +#define RT5670_MIC1_BS_75AV (0x1 << 15) +#define RT5670_MIC2_BS_MASK (0x1 << 14) +#define RT5670_MIC2_BS_SFT 14 +#define RT5670_MIC2_BS_9AV (0x0 << 14) +#define RT5670_MIC2_BS_75AV (0x1 << 14) +#define RT5670_MIC1_CLK_MASK (0x1 << 13) +#define RT5670_MIC1_CLK_SFT 13 +#define RT5670_MIC1_CLK_DIS (0x0 << 13) +#define RT5670_MIC1_CLK_EN (0x1 << 13) +#define RT5670_MIC2_CLK_MASK (0x1 << 12) +#define RT5670_MIC2_CLK_SFT 12 +#define RT5670_MIC2_CLK_DIS (0x0 << 12) +#define RT5670_MIC2_CLK_EN (0x1 << 12) +#define RT5670_MIC1_OVCD_MASK (0x1 << 11) +#define RT5670_MIC1_OVCD_SFT 11 +#define RT5670_MIC1_OVCD_DIS (0x0 << 11) +#define RT5670_MIC1_OVCD_EN (0x1 << 11) +#define RT5670_MIC1_OVTH_MASK (0x3 << 9) +#define RT5670_MIC1_OVTH_SFT 9 +#define RT5670_MIC1_OVTH_600UA (0x0 << 9) +#define RT5670_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5670_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5670_MIC2_OVCD_MASK (0x1 << 8) +#define RT5670_MIC2_OVCD_SFT 8 +#define RT5670_MIC2_OVCD_DIS (0x0 << 8) +#define RT5670_MIC2_OVCD_EN (0x1 << 8) +#define RT5670_MIC2_OVTH_MASK (0x3 << 6) +#define RT5670_MIC2_OVTH_SFT 6 +#define RT5670_MIC2_OVTH_600UA (0x0 << 6) +#define RT5670_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5670_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5670_PWR_MB_MASK (0x1 << 5) +#define RT5670_PWR_MB_SFT 5 +#define RT5670_PWR_MB_PD (0x0 << 5) +#define RT5670_PWR_MB_PU (0x1 << 5) +#define RT5670_PWR_CLK25M_MASK (0x1 << 4) +#define RT5670_PWR_CLK25M_SFT 4 +#define RT5670_PWR_CLK25M_PD (0x0 << 4) +#define RT5670_PWR_CLK25M_PU (0x1 << 4) + +/* Analog JD Control 1 (0x94) */ +#define RT5670_JD1_MODE_MASK (0x3 << 0) +#define RT5670_JD1_MODE_0 (0x0 << 0) +#define RT5670_JD1_MODE_1 (0x1 << 0) +#define RT5670_JD1_MODE_2 (0x2 << 0) + +/* VAD Control 4 (0x9d) */ +#define RT5670_VAD_SEL_MASK (0x3 << 8) +#define RT5670_VAD_SEL_SFT 8 + +/* EQ Control 1 (0xb0) */ +#define RT5670_EQ_SRC_MASK (0x1 << 15) +#define RT5670_EQ_SRC_SFT 15 +#define RT5670_EQ_SRC_DAC (0x0 << 15) +#define RT5670_EQ_SRC_ADC (0x1 << 15) +#define RT5670_EQ_UPD (0x1 << 14) +#define RT5670_EQ_UPD_BIT 14 +#define RT5670_EQ_CD_MASK (0x1 << 13) +#define RT5670_EQ_CD_SFT 13 +#define RT5670_EQ_CD_DIS (0x0 << 13) +#define RT5670_EQ_CD_EN (0x1 << 13) +#define RT5670_EQ_DITH_MASK (0x3 << 8) +#define RT5670_EQ_DITH_SFT 8 +#define RT5670_EQ_DITH_NOR (0x0 << 8) +#define RT5670_EQ_DITH_LSB (0x1 << 8) +#define RT5670_EQ_DITH_LSB_1 (0x2 << 8) +#define RT5670_EQ_DITH_LSB_2 (0x3 << 8) + +/* EQ Control 2 (0xb1) */ +#define RT5670_EQ_HPF1_M_MASK (0x1 << 8) +#define RT5670_EQ_HPF1_M_SFT 8 +#define RT5670_EQ_HPF1_M_HI (0x0 << 8) +#define RT5670_EQ_HPF1_M_1ST (0x1 << 8) +#define RT5670_EQ_LPF1_M_MASK (0x1 << 7) +#define RT5670_EQ_LPF1_M_SFT 7 +#define RT5670_EQ_LPF1_M_LO (0x0 << 7) +#define RT5670_EQ_LPF1_M_1ST (0x1 << 7) +#define RT5670_EQ_HPF2_MASK (0x1 << 6) +#define RT5670_EQ_HPF2_SFT 6 +#define RT5670_EQ_HPF2_DIS (0x0 << 6) +#define RT5670_EQ_HPF2_EN (0x1 << 6) +#define RT5670_EQ_HPF1_MASK (0x1 << 5) +#define RT5670_EQ_HPF1_SFT 5 +#define RT5670_EQ_HPF1_DIS (0x0 << 5) +#define RT5670_EQ_HPF1_EN (0x1 << 5) +#define RT5670_EQ_BPF4_MASK (0x1 << 4) +#define RT5670_EQ_BPF4_SFT 4 +#define RT5670_EQ_BPF4_DIS (0x0 << 4) +#define RT5670_EQ_BPF4_EN (0x1 << 4) +#define RT5670_EQ_BPF3_MASK (0x1 << 3) +#define RT5670_EQ_BPF3_SFT 3 +#define RT5670_EQ_BPF3_DIS (0x0 << 3) +#define RT5670_EQ_BPF3_EN (0x1 << 3) +#define RT5670_EQ_BPF2_MASK (0x1 << 2) +#define RT5670_EQ_BPF2_SFT 2 +#define RT5670_EQ_BPF2_DIS (0x0 << 2) +#define RT5670_EQ_BPF2_EN (0x1 << 2) +#define RT5670_EQ_BPF1_MASK (0x1 << 1) +#define RT5670_EQ_BPF1_SFT 1 +#define RT5670_EQ_BPF1_DIS (0x0 << 1) +#define RT5670_EQ_BPF1_EN (0x1 << 1) +#define RT5670_EQ_LPF_MASK (0x1) +#define RT5670_EQ_LPF_SFT 0 +#define RT5670_EQ_LPF_DIS (0x0) +#define RT5670_EQ_LPF_EN (0x1) +#define RT5670_EQ_CTRL_MASK (0x7f) + +/* Memory Test (0xb2) */ +#define RT5670_MT_MASK (0x1 << 15) +#define RT5670_MT_SFT 15 +#define RT5670_MT_DIS (0x0 << 15) +#define RT5670_MT_EN (0x1 << 15) + +/* DRC/AGC Control 1 (0xb4) */ +#define RT5670_DRC_AGC_P_MASK (0x1 << 15) +#define RT5670_DRC_AGC_P_SFT 15 +#define RT5670_DRC_AGC_P_DAC (0x0 << 15) +#define RT5670_DRC_AGC_P_ADC (0x1 << 15) +#define RT5670_DRC_AGC_MASK (0x1 << 14) +#define RT5670_DRC_AGC_SFT 14 +#define RT5670_DRC_AGC_DIS (0x0 << 14) +#define RT5670_DRC_AGC_EN (0x1 << 14) +#define RT5670_DRC_AGC_UPD (0x1 << 13) +#define RT5670_DRC_AGC_UPD_BIT 13 +#define RT5670_DRC_AGC_AR_MASK (0x1f << 8) +#define RT5670_DRC_AGC_AR_SFT 8 +#define RT5670_DRC_AGC_R_MASK (0x7 << 5) +#define RT5670_DRC_AGC_R_SFT 5 +#define RT5670_DRC_AGC_R_48K (0x1 << 5) +#define RT5670_DRC_AGC_R_96K (0x2 << 5) +#define RT5670_DRC_AGC_R_192K (0x3 << 5) +#define RT5670_DRC_AGC_R_441K (0x5 << 5) +#define RT5670_DRC_AGC_R_882K (0x6 << 5) +#define RT5670_DRC_AGC_R_1764K (0x7 << 5) +#define RT5670_DRC_AGC_RC_MASK (0x1f) +#define RT5670_DRC_AGC_RC_SFT 0 + +/* DRC/AGC Control 2 (0xb5) */ +#define RT5670_DRC_AGC_POB_MASK (0x3f << 8) +#define RT5670_DRC_AGC_POB_SFT 8 +#define RT5670_DRC_AGC_CP_MASK (0x1 << 7) +#define RT5670_DRC_AGC_CP_SFT 7 +#define RT5670_DRC_AGC_CP_DIS (0x0 << 7) +#define RT5670_DRC_AGC_CP_EN (0x1 << 7) +#define RT5670_DRC_AGC_CPR_MASK (0x3 << 5) +#define RT5670_DRC_AGC_CPR_SFT 5 +#define RT5670_DRC_AGC_CPR_1_1 (0x0 << 5) +#define RT5670_DRC_AGC_CPR_1_2 (0x1 << 5) +#define RT5670_DRC_AGC_CPR_1_3 (0x2 << 5) +#define RT5670_DRC_AGC_CPR_1_4 (0x3 << 5) +#define RT5670_DRC_AGC_PRB_MASK (0x1f) +#define RT5670_DRC_AGC_PRB_SFT 0 + +/* DRC/AGC Control 3 (0xb6) */ +#define RT5670_DRC_AGC_NGB_MASK (0xf << 12) +#define RT5670_DRC_AGC_NGB_SFT 12 +#define RT5670_DRC_AGC_TAR_MASK (0x1f << 7) +#define RT5670_DRC_AGC_TAR_SFT 7 +#define RT5670_DRC_AGC_NG_MASK (0x1 << 6) +#define RT5670_DRC_AGC_NG_SFT 6 +#define RT5670_DRC_AGC_NG_DIS (0x0 << 6) +#define RT5670_DRC_AGC_NG_EN (0x1 << 6) +#define RT5670_DRC_AGC_NGH_MASK (0x1 << 5) +#define RT5670_DRC_AGC_NGH_SFT 5 +#define RT5670_DRC_AGC_NGH_DIS (0x0 << 5) +#define RT5670_DRC_AGC_NGH_EN (0x1 << 5) +#define RT5670_DRC_AGC_NGT_MASK (0x1f) +#define RT5670_DRC_AGC_NGT_SFT 0 + +/* Jack Detect Control (0xbb) */ +#define RT5670_JD_MASK (0x7 << 13) +#define RT5670_JD_SFT 13 +#define RT5670_JD_DIS (0x0 << 13) +#define RT5670_JD_GPIO1 (0x1 << 13) +#define RT5670_JD_JD1_IN4P (0x2 << 13) +#define RT5670_JD_JD2_IN4N (0x3 << 13) +#define RT5670_JD_GPIO2 (0x4 << 13) +#define RT5670_JD_GPIO3 (0x5 << 13) +#define RT5670_JD_GPIO4 (0x6 << 13) +#define RT5670_JD_HP_MASK (0x1 << 11) +#define RT5670_JD_HP_SFT 11 +#define RT5670_JD_HP_DIS (0x0 << 11) +#define RT5670_JD_HP_EN (0x1 << 11) +#define RT5670_JD_HP_TRG_MASK (0x1 << 10) +#define RT5670_JD_HP_TRG_SFT 10 +#define RT5670_JD_HP_TRG_LO (0x0 << 10) +#define RT5670_JD_HP_TRG_HI (0x1 << 10) +#define RT5670_JD_SPL_MASK (0x1 << 9) +#define RT5670_JD_SPL_SFT 9 +#define RT5670_JD_SPL_DIS (0x0 << 9) +#define RT5670_JD_SPL_EN (0x1 << 9) +#define RT5670_JD_SPL_TRG_MASK (0x1 << 8) +#define RT5670_JD_SPL_TRG_SFT 8 +#define RT5670_JD_SPL_TRG_LO (0x0 << 8) +#define RT5670_JD_SPL_TRG_HI (0x1 << 8) +#define RT5670_JD_SPR_MASK (0x1 << 7) +#define RT5670_JD_SPR_SFT 7 +#define RT5670_JD_SPR_DIS (0x0 << 7) +#define RT5670_JD_SPR_EN (0x1 << 7) +#define RT5670_JD_SPR_TRG_MASK (0x1 << 6) +#define RT5670_JD_SPR_TRG_SFT 6 +#define RT5670_JD_SPR_TRG_LO (0x0 << 6) +#define RT5670_JD_SPR_TRG_HI (0x1 << 6) +#define RT5670_JD_MO_MASK (0x1 << 5) +#define RT5670_JD_MO_SFT 5 +#define RT5670_JD_MO_DIS (0x0 << 5) +#define RT5670_JD_MO_EN (0x1 << 5) +#define RT5670_JD_MO_TRG_MASK (0x1 << 4) +#define RT5670_JD_MO_TRG_SFT 4 +#define RT5670_JD_MO_TRG_LO (0x0 << 4) +#define RT5670_JD_MO_TRG_HI (0x1 << 4) +#define RT5670_JD_LO_MASK (0x1 << 3) +#define RT5670_JD_LO_SFT 3 +#define RT5670_JD_LO_DIS (0x0 << 3) +#define RT5670_JD_LO_EN (0x1 << 3) +#define RT5670_JD_LO_TRG_MASK (0x1 << 2) +#define RT5670_JD_LO_TRG_SFT 2 +#define RT5670_JD_LO_TRG_LO (0x0 << 2) +#define RT5670_JD_LO_TRG_HI (0x1 << 2) +#define RT5670_JD1_IN4P_MASK (0x1 << 1) +#define RT5670_JD1_IN4P_SFT 1 +#define RT5670_JD1_IN4P_DIS (0x0 << 1) +#define RT5670_JD1_IN4P_EN (0x1 << 1) +#define RT5670_JD2_IN4N_MASK (0x1) +#define RT5670_JD2_IN4N_SFT 0 +#define RT5670_JD2_IN4N_DIS (0x0) +#define RT5670_JD2_IN4N_EN (0x1) + +/* IRQ Control 1 (0xbd) */ +#define RT5670_IRQ_JD_MASK (0x1 << 15) +#define RT5670_IRQ_JD_SFT 15 +#define RT5670_IRQ_JD_BP (0x0 << 15) +#define RT5670_IRQ_JD_NOR (0x1 << 15) +#define RT5670_IRQ_OT_MASK (0x1 << 14) +#define RT5670_IRQ_OT_SFT 14 +#define RT5670_IRQ_OT_BP (0x0 << 14) +#define RT5670_IRQ_OT_NOR (0x1 << 14) +#define RT5670_JD_STKY_MASK (0x1 << 13) +#define RT5670_JD_STKY_SFT 13 +#define RT5670_JD_STKY_DIS (0x0 << 13) +#define RT5670_JD_STKY_EN (0x1 << 13) +#define RT5670_OT_STKY_MASK (0x1 << 12) +#define RT5670_OT_STKY_SFT 12 +#define RT5670_OT_STKY_DIS (0x0 << 12) +#define RT5670_OT_STKY_EN (0x1 << 12) +#define RT5670_JD_P_MASK (0x1 << 11) +#define RT5670_JD_P_SFT 11 +#define RT5670_JD_P_NOR (0x0 << 11) +#define RT5670_JD_P_INV (0x1 << 11) +#define RT5670_OT_P_MASK (0x1 << 10) +#define RT5670_OT_P_SFT 10 +#define RT5670_OT_P_NOR (0x0 << 10) +#define RT5670_OT_P_INV (0x1 << 10) +#define RT5670_JD1_1_EN_MASK (0x1 << 9) +#define RT5670_JD1_1_EN_SFT 9 +#define RT5670_JD1_1_DIS (0x0 << 9) +#define RT5670_JD1_1_EN (0x1 << 9) + +/* IRQ Control 2 (0xbe) */ +#define RT5670_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5670_IRQ_MB1_OC_SFT 15 +#define RT5670_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5670_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5670_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5670_IRQ_MB2_OC_SFT 14 +#define RT5670_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5670_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5670_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5670_MB1_OC_STKY_SFT 11 +#define RT5670_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5670_MB1_OC_STKY_EN (0x1 << 11) +#define RT5670_MB2_OC_STKY_MASK (0x1 << 10) +#define RT5670_MB2_OC_STKY_SFT 10 +#define RT5670_MB2_OC_STKY_DIS (0x0 << 10) +#define RT5670_MB2_OC_STKY_EN (0x1 << 10) +#define RT5670_MB1_OC_P_MASK (0x1 << 7) +#define RT5670_MB1_OC_P_SFT 7 +#define RT5670_MB1_OC_P_NOR (0x0 << 7) +#define RT5670_MB1_OC_P_INV (0x1 << 7) +#define RT5670_MB2_OC_P_MASK (0x1 << 6) +#define RT5670_MB2_OC_P_SFT 6 +#define RT5670_MB2_OC_P_NOR (0x0 << 6) +#define RT5670_MB2_OC_P_INV (0x1 << 6) +#define RT5670_MB1_OC_CLR (0x1 << 3) +#define RT5670_MB1_OC_CLR_SFT 3 +#define RT5670_MB2_OC_CLR (0x1 << 2) +#define RT5670_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5670_GP1_PIN_MASK (0x1 << 15) +#define RT5670_GP1_PIN_SFT 15 +#define RT5670_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5670_GP1_PIN_IRQ (0x1 << 15) +#define RT5670_GP2_PIN_MASK (0x1 << 14) +#define RT5670_GP2_PIN_SFT 14 +#define RT5670_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5670_GP2_PIN_DMIC1_SCL (0x1 << 14) +#define RT5670_GP3_PIN_MASK (0x3 << 12) +#define RT5670_GP3_PIN_SFT 12 +#define RT5670_GP3_PIN_GPIO3 (0x0 << 12) +#define RT5670_GP3_PIN_DMIC1_SDA (0x1 << 12) +#define RT5670_GP3_PIN_IRQ (0x2 << 12) +#define RT5670_GP4_PIN_MASK (0x1 << 11) +#define RT5670_GP4_PIN_SFT 11 +#define RT5670_GP4_PIN_GPIO4 (0x0 << 11) +#define RT5670_GP4_PIN_DMIC2_SDA (0x1 << 11) +#define RT5670_DP_SIG_MASK (0x1 << 10) +#define RT5670_DP_SIG_SFT 10 +#define RT5670_DP_SIG_TEST (0x0 << 10) +#define RT5670_DP_SIG_AP (0x1 << 10) +#define RT5670_GPIO_M_MASK (0x1 << 9) +#define RT5670_GPIO_M_SFT 9 +#define RT5670_GPIO_M_FLT (0x0 << 9) +#define RT5670_GPIO_M_PH (0x1 << 9) +#define RT5670_I2S2_PIN_MASK (0x1 << 8) +#define RT5670_I2S2_PIN_SFT 8 +#define RT5670_I2S2_PIN_I2S (0x0 << 8) +#define RT5670_I2S2_PIN_GPIO (0x1 << 8) +#define RT5670_GP5_PIN_MASK (0x1 << 7) +#define RT5670_GP5_PIN_SFT 7 +#define RT5670_GP5_PIN_GPIO5 (0x0 << 7) +#define RT5670_GP5_PIN_DMIC3_SDA (0x1 << 7) +#define RT5670_GP6_PIN_MASK (0x1 << 6) +#define RT5670_GP6_PIN_SFT 6 +#define RT5670_GP6_PIN_GPIO6 (0x0 << 6) +#define RT5670_GP6_PIN_DMIC1_SDA (0x1 << 6) +#define RT5670_GP7_PIN_MASK (0x3 << 4) +#define RT5670_GP7_PIN_SFT 4 +#define RT5670_GP7_PIN_GPIO7 (0x0 << 4) +#define RT5670_GP7_PIN_DMIC1_SDA (0x1 << 4) +#define RT5670_GP7_PIN_PDM_SCL2 (0x2 << 4) +#define RT5670_GP8_PIN_MASK (0x1 << 3) +#define RT5670_GP8_PIN_SFT 3 +#define RT5670_GP8_PIN_GPIO8 (0x0 << 3) +#define RT5670_GP8_PIN_DMIC2_SDA (0x1 << 3) +#define RT5670_GP9_PIN_MASK (0x1 << 2) +#define RT5670_GP9_PIN_SFT 2 +#define RT5670_GP9_PIN_GPIO9 (0x0 << 2) +#define RT5670_GP9_PIN_DMIC3_SDA (0x1 << 2) +#define RT5670_GP10_PIN_MASK (0x3) +#define RT5670_GP10_PIN_SFT 0 +#define RT5670_GP10_PIN_GPIO9 (0x0) +#define RT5670_GP10_PIN_DMIC3_SDA (0x1) +#define RT5670_GP10_PIN_PDM_ADT2 (0x2) + +/* GPIO Control 2 (0xc1) */ +#define RT5670_GP4_PF_MASK (0x1 << 11) +#define RT5670_GP4_PF_SFT 11 +#define RT5670_GP4_PF_IN (0x0 << 11) +#define RT5670_GP4_PF_OUT (0x1 << 11) +#define RT5670_GP4_OUT_MASK (0x1 << 10) +#define RT5670_GP4_OUT_SFT 10 +#define RT5670_GP4_OUT_LO (0x0 << 10) +#define RT5670_GP4_OUT_HI (0x1 << 10) +#define RT5670_GP4_P_MASK (0x1 << 9) +#define RT5670_GP4_P_SFT 9 +#define RT5670_GP4_P_NOR (0x0 << 9) +#define RT5670_GP4_P_INV (0x1 << 9) +#define RT5670_GP3_PF_MASK (0x1 << 8) +#define RT5670_GP3_PF_SFT 8 +#define RT5670_GP3_PF_IN (0x0 << 8) +#define RT5670_GP3_PF_OUT (0x1 << 8) +#define RT5670_GP3_OUT_MASK (0x1 << 7) +#define RT5670_GP3_OUT_SFT 7 +#define RT5670_GP3_OUT_LO (0x0 << 7) +#define RT5670_GP3_OUT_HI (0x1 << 7) +#define RT5670_GP3_P_MASK (0x1 << 6) +#define RT5670_GP3_P_SFT 6 +#define RT5670_GP3_P_NOR (0x0 << 6) +#define RT5670_GP3_P_INV (0x1 << 6) +#define RT5670_GP2_PF_MASK (0x1 << 5) +#define RT5670_GP2_PF_SFT 5 +#define RT5670_GP2_PF_IN (0x0 << 5) +#define RT5670_GP2_PF_OUT (0x1 << 5) +#define RT5670_GP2_OUT_MASK (0x1 << 4) +#define RT5670_GP2_OUT_SFT 4 +#define RT5670_GP2_OUT_LO (0x0 << 4) +#define RT5670_GP2_OUT_HI (0x1 << 4) +#define RT5670_GP2_P_MASK (0x1 << 3) +#define RT5670_GP2_P_SFT 3 +#define RT5670_GP2_P_NOR (0x0 << 3) +#define RT5670_GP2_P_INV (0x1 << 3) +#define RT5670_GP1_PF_MASK (0x1 << 2) +#define RT5670_GP1_PF_SFT 2 +#define RT5670_GP1_PF_IN (0x0 << 2) +#define RT5670_GP1_PF_OUT (0x1 << 2) +#define RT5670_GP1_OUT_MASK (0x1 << 1) +#define RT5670_GP1_OUT_SFT 1 +#define RT5670_GP1_OUT_LO (0x0 << 1) +#define RT5670_GP1_OUT_HI (0x1 << 1) +#define RT5670_GP1_P_MASK (0x1) +#define RT5670_GP1_P_SFT 0 +#define RT5670_GP1_P_NOR (0x0) +#define RT5670_GP1_P_INV (0x1) + +/* Scramble Function (0xcd) */ +#define RT5670_SCB_KEY_MASK (0xff) +#define RT5670_SCB_KEY_SFT 0 + +/* Scramble Control (0xce) */ +#define RT5670_SCB_SWAP_MASK (0x1 << 15) +#define RT5670_SCB_SWAP_SFT 15 +#define RT5670_SCB_SWAP_DIS (0x0 << 15) +#define RT5670_SCB_SWAP_EN (0x1 << 15) +#define RT5670_SCB_MASK (0x1 << 14) +#define RT5670_SCB_SFT 14 +#define RT5670_SCB_DIS (0x0 << 14) +#define RT5670_SCB_EN (0x1 << 14) + +/* Baseback Control (0xcf) */ +#define RT5670_BB_MASK (0x1 << 15) +#define RT5670_BB_SFT 15 +#define RT5670_BB_DIS (0x0 << 15) +#define RT5670_BB_EN (0x1 << 15) +#define RT5670_BB_CT_MASK (0x7 << 12) +#define RT5670_BB_CT_SFT 12 +#define RT5670_BB_CT_A (0x0 << 12) +#define RT5670_BB_CT_B (0x1 << 12) +#define RT5670_BB_CT_C (0x2 << 12) +#define RT5670_BB_CT_D (0x3 << 12) +#define RT5670_M_BB_L_MASK (0x1 << 9) +#define RT5670_M_BB_L_SFT 9 +#define RT5670_M_BB_R_MASK (0x1 << 8) +#define RT5670_M_BB_R_SFT 8 +#define RT5670_M_BB_HPF_L_MASK (0x1 << 7) +#define RT5670_M_BB_HPF_L_SFT 7 +#define RT5670_M_BB_HPF_R_MASK (0x1 << 6) +#define RT5670_M_BB_HPF_R_SFT 6 +#define RT5670_G_BB_BST_MASK (0x3f) +#define RT5670_G_BB_BST_SFT 0 + +/* MP3 Plus Control 1 (0xd0) */ +#define RT5670_M_MP3_L_MASK (0x1 << 15) +#define RT5670_M_MP3_L_SFT 15 +#define RT5670_M_MP3_R_MASK (0x1 << 14) +#define RT5670_M_MP3_R_SFT 14 +#define RT5670_M_MP3_MASK (0x1 << 13) +#define RT5670_M_MP3_SFT 13 +#define RT5670_M_MP3_DIS (0x0 << 13) +#define RT5670_M_MP3_EN (0x1 << 13) +#define RT5670_EG_MP3_MASK (0x1f << 8) +#define RT5670_EG_MP3_SFT 8 +#define RT5670_MP3_HLP_MASK (0x1 << 7) +#define RT5670_MP3_HLP_SFT 7 +#define RT5670_MP3_HLP_DIS (0x0 << 7) +#define RT5670_MP3_HLP_EN (0x1 << 7) +#define RT5670_M_MP3_ORG_L_MASK (0x1 << 6) +#define RT5670_M_MP3_ORG_L_SFT 6 +#define RT5670_M_MP3_ORG_R_MASK (0x1 << 5) +#define RT5670_M_MP3_ORG_R_SFT 5 + +/* MP3 Plus Control 2 (0xd1) */ +#define RT5670_MP3_WT_MASK (0x1 << 13) +#define RT5670_MP3_WT_SFT 13 +#define RT5670_MP3_WT_1_4 (0x0 << 13) +#define RT5670_MP3_WT_1_2 (0x1 << 13) +#define RT5670_OG_MP3_MASK (0x1f << 8) +#define RT5670_OG_MP3_SFT 8 +#define RT5670_HG_MP3_MASK (0x3f) +#define RT5670_HG_MP3_SFT 0 + +/* 3D HP Control 1 (0xd2) */ +#define RT5670_3D_CF_MASK (0x1 << 15) +#define RT5670_3D_CF_SFT 15 +#define RT5670_3D_CF_DIS (0x0 << 15) +#define RT5670_3D_CF_EN (0x1 << 15) +#define RT5670_3D_HP_MASK (0x1 << 14) +#define RT5670_3D_HP_SFT 14 +#define RT5670_3D_HP_DIS (0x0 << 14) +#define RT5670_3D_HP_EN (0x1 << 14) +#define RT5670_3D_BT_MASK (0x1 << 13) +#define RT5670_3D_BT_SFT 13 +#define RT5670_3D_BT_DIS (0x0 << 13) +#define RT5670_3D_BT_EN (0x1 << 13) +#define RT5670_3D_1F_MIX_MASK (0x3 << 11) +#define RT5670_3D_1F_MIX_SFT 11 +#define RT5670_3D_HP_M_MASK (0x1 << 10) +#define RT5670_3D_HP_M_SFT 10 +#define RT5670_3D_HP_M_SUR (0x0 << 10) +#define RT5670_3D_HP_M_FRO (0x1 << 10) +#define RT5670_M_3D_HRTF_MASK (0x1 << 9) +#define RT5670_M_3D_HRTF_SFT 9 +#define RT5670_M_3D_D2H_MASK (0x1 << 8) +#define RT5670_M_3D_D2H_SFT 8 +#define RT5670_M_3D_D2R_MASK (0x1 << 7) +#define RT5670_M_3D_D2R_SFT 7 +#define RT5670_M_3D_REVB_MASK (0x1 << 6) +#define RT5670_M_3D_REVB_SFT 6 + +/* Adjustable high pass filter control 1 (0xd3) */ +#define RT5670_2ND_HPF_MASK (0x1 << 15) +#define RT5670_2ND_HPF_SFT 15 +#define RT5670_2ND_HPF_DIS (0x0 << 15) +#define RT5670_2ND_HPF_EN (0x1 << 15) +#define RT5670_HPF_CF_L_MASK (0x7 << 12) +#define RT5670_HPF_CF_L_SFT 12 +#define RT5670_1ST_HPF_MASK (0x1 << 11) +#define RT5670_1ST_HPF_SFT 11 +#define RT5670_1ST_HPF_DIS (0x0 << 11) +#define RT5670_1ST_HPF_EN (0x1 << 11) +#define RT5670_HPF_CF_R_MASK (0x7 << 8) +#define RT5670_HPF_CF_R_SFT 8 +#define RT5670_ZD_T_MASK (0x3 << 6) +#define RT5670_ZD_T_SFT 6 +#define RT5670_ZD_F_MASK (0x3 << 4) +#define RT5670_ZD_F_SFT 4 +#define RT5670_ZD_F_IM (0x0 << 4) +#define RT5670_ZD_F_ZC_IM (0x1 << 4) +#define RT5670_ZD_F_ZC_IOD (0x2 << 4) +#define RT5670_ZD_F_UN (0x3 << 4) + +/* HP calibration control and Amp detection (0xd6) */ +#define RT5670_SI_DAC_MASK (0x1 << 11) +#define RT5670_SI_DAC_SFT 11 +#define RT5670_SI_DAC_AUTO (0x0 << 11) +#define RT5670_SI_DAC_TEST (0x1 << 11) +#define RT5670_DC_CAL_M_MASK (0x1 << 10) +#define RT5670_DC_CAL_M_SFT 10 +#define RT5670_DC_CAL_M_CAL (0x0 << 10) +#define RT5670_DC_CAL_M_NOR (0x1 << 10) +#define RT5670_DC_CAL_MASK (0x1 << 9) +#define RT5670_DC_CAL_SFT 9 +#define RT5670_DC_CAL_DIS (0x0 << 9) +#define RT5670_DC_CAL_EN (0x1 << 9) +#define RT5670_HPD_RCV_MASK (0x7 << 6) +#define RT5670_HPD_RCV_SFT 6 +#define RT5670_HPD_PS_MASK (0x1 << 5) +#define RT5670_HPD_PS_SFT 5 +#define RT5670_HPD_PS_DIS (0x0 << 5) +#define RT5670_HPD_PS_EN (0x1 << 5) +#define RT5670_CAL_M_MASK (0x1 << 4) +#define RT5670_CAL_M_SFT 4 +#define RT5670_CAL_M_DEP (0x0 << 4) +#define RT5670_CAL_M_CAL (0x1 << 4) +#define RT5670_CAL_MASK (0x1 << 3) +#define RT5670_CAL_SFT 3 +#define RT5670_CAL_DIS (0x0 << 3) +#define RT5670_CAL_EN (0x1 << 3) +#define RT5670_CAL_TEST_MASK (0x1 << 2) +#define RT5670_CAL_TEST_SFT 2 +#define RT5670_CAL_TEST_DIS (0x0 << 2) +#define RT5670_CAL_TEST_EN (0x1 << 2) +#define RT5670_CAL_P_MASK (0x3) +#define RT5670_CAL_P_SFT 0 +#define RT5670_CAL_P_NONE (0x0) +#define RT5670_CAL_P_CAL (0x1) +#define RT5670_CAL_P_DAC_CAL (0x2) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5670_SV_MASK (0x1 << 15) +#define RT5670_SV_SFT 15 +#define RT5670_SV_DIS (0x0 << 15) +#define RT5670_SV_EN (0x1 << 15) +#define RT5670_SPO_SV_MASK (0x1 << 14) +#define RT5670_SPO_SV_SFT 14 +#define RT5670_SPO_SV_DIS (0x0 << 14) +#define RT5670_SPO_SV_EN (0x1 << 14) +#define RT5670_OUT_SV_MASK (0x1 << 13) +#define RT5670_OUT_SV_SFT 13 +#define RT5670_OUT_SV_DIS (0x0 << 13) +#define RT5670_OUT_SV_EN (0x1 << 13) +#define RT5670_HP_SV_MASK (0x1 << 12) +#define RT5670_HP_SV_SFT 12 +#define RT5670_HP_SV_DIS (0x0 << 12) +#define RT5670_HP_SV_EN (0x1 << 12) +#define RT5670_ZCD_DIG_MASK (0x1 << 11) +#define RT5670_ZCD_DIG_SFT 11 +#define RT5670_ZCD_DIG_DIS (0x0 << 11) +#define RT5670_ZCD_DIG_EN (0x1 << 11) +#define RT5670_ZCD_MASK (0x1 << 10) +#define RT5670_ZCD_SFT 10 +#define RT5670_ZCD_PD (0x0 << 10) +#define RT5670_ZCD_PU (0x1 << 10) +#define RT5670_M_ZCD_MASK (0x3f << 4) +#define RT5670_M_ZCD_SFT 4 +#define RT5670_M_ZCD_RM_L (0x1 << 9) +#define RT5670_M_ZCD_RM_R (0x1 << 8) +#define RT5670_M_ZCD_SM_L (0x1 << 7) +#define RT5670_M_ZCD_SM_R (0x1 << 6) +#define RT5670_M_ZCD_OM_L (0x1 << 5) +#define RT5670_M_ZCD_OM_R (0x1 << 4) +#define RT5670_SV_DLY_MASK (0xf) +#define RT5670_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5670_ZCD_HP_MASK (0x1 << 15) +#define RT5670_ZCD_HP_SFT 15 +#define RT5670_ZCD_HP_DIS (0x0 << 15) +#define RT5670_ZCD_HP_EN (0x1 << 15) + + +/* Codec Private Register definition */ +/* 3D Speaker Control (0x63) */ +#define RT5670_3D_SPK_MASK (0x1 << 15) +#define RT5670_3D_SPK_SFT 15 +#define RT5670_3D_SPK_DIS (0x0 << 15) +#define RT5670_3D_SPK_EN (0x1 << 15) +#define RT5670_3D_SPK_M_MASK (0x3 << 13) +#define RT5670_3D_SPK_M_SFT 13 +#define RT5670_3D_SPK_CG_MASK (0x1f << 8) +#define RT5670_3D_SPK_CG_SFT 8 +#define RT5670_3D_SPK_SG_MASK (0x1f) +#define RT5670_3D_SPK_SG_SFT 0 + +/* Wind Noise Detection Control 1 (0x6c) */ +#define RT5670_WND_MASK (0x1 << 15) +#define RT5670_WND_SFT 15 +#define RT5670_WND_DIS (0x0 << 15) +#define RT5670_WND_EN (0x1 << 15) + +/* Wind Noise Detection Control 2 (0x6d) */ +#define RT5670_WND_FC_NW_MASK (0x3f << 10) +#define RT5670_WND_FC_NW_SFT 10 +#define RT5670_WND_FC_WK_MASK (0x3f << 4) +#define RT5670_WND_FC_WK_SFT 4 + +/* Wind Noise Detection Control 3 (0x6e) */ +#define RT5670_HPF_FC_MASK (0x3f << 6) +#define RT5670_HPF_FC_SFT 6 +#define RT5670_WND_FC_ST_MASK (0x3f) +#define RT5670_WND_FC_ST_SFT 0 + +/* Wind Noise Detection Control 4 (0x6f) */ +#define RT5670_WND_TH_LO_MASK (0x3ff) +#define RT5670_WND_TH_LO_SFT 0 + +/* Wind Noise Detection Control 5 (0x70) */ +#define RT5670_WND_TH_HI_MASK (0x3ff) +#define RT5670_WND_TH_HI_SFT 0 + +/* Wind Noise Detection Control 8 (0x73) */ +#define RT5670_WND_WIND_MASK (0x1 << 13) /* Read-Only */ +#define RT5670_WND_WIND_SFT 13 +#define RT5670_WND_STRONG_MASK (0x1 << 12) /* Read-Only */ +#define RT5670_WND_STRONG_SFT 12 +enum { + RT5670_NO_WIND, + RT5670_BREEZE, + RT5670_STORM, +}; + +/* Dipole Speaker Interface (0x75) */ +#define RT5670_DP_ATT_MASK (0x3 << 14) +#define RT5670_DP_ATT_SFT 14 +#define RT5670_DP_SPK_MASK (0x1 << 10) +#define RT5670_DP_SPK_SFT 10 +#define RT5670_DP_SPK_DIS (0x0 << 10) +#define RT5670_DP_SPK_EN (0x1 << 10) + +/* EQ Pre Volume Control (0xb3) */ +#define RT5670_EQ_PRE_VOL_MASK (0xffff) +#define RT5670_EQ_PRE_VOL_SFT 0 + +/* EQ Post Volume Control (0xb4) */ +#define RT5670_EQ_PST_VOL_MASK (0xffff) +#define RT5670_EQ_PST_VOL_SFT 0 + +/* Jack Detect Control 3 (0xf8) */ +#define RT5670_CMP_MIC_IN_DET_MASK (0x7 << 12) +#define RT5670_JD_CBJ_EN (0x1 << 7) +#define RT5670_JD_CBJ_POL (0x1 << 6) +#define RT5670_JD_TRI_CBJ_SEL_MASK (0x7 << 3) +#define RT5670_JD_TRI_CBJ_SEL_SFT (3) +#define RT5670_JD_CBJ_GPIO_JD1 (0x0 << 3) +#define RT5670_JD_CBJ_JD1_1 (0x1 << 3) +#define RT5670_JD_CBJ_JD1_2 (0x2 << 3) +#define RT5670_JD_CBJ_JD2 (0x3 << 3) +#define RT5670_JD_CBJ_JD3 (0x4 << 3) +#define RT5670_JD_CBJ_GPIO_JD2 (0x5 << 3) +#define RT5670_JD_CBJ_MX0B_12 (0x6 << 3) +#define RT5670_JD_TRI_HPO_SEL_MASK (0x7 << 3) +#define RT5670_JD_TRI_HPO_SEL_SFT (0) +#define RT5670_JD_HPO_GPIO_JD1 (0x0) +#define RT5670_JD_HPO_JD1_1 (0x1) +#define RT5670_JD_HPO_JD1_2 (0x2) +#define RT5670_JD_HPO_JD2 (0x3) +#define RT5670_JD_HPO_JD3 (0x4) +#define RT5670_JD_HPO_GPIO_JD2 (0x5) +#define RT5670_JD_HPO_MX0B_12 (0x6) + +/* Digital Misc Control (0xfa) */ +#define RT5670_RST_DSP (0x1 << 13) +#define RT5670_IF1_ADC1_IN1_SEL (0x1 << 12) +#define RT5670_IF1_ADC1_IN1_SFT 12 +#define RT5670_IF1_ADC1_IN2_SEL (0x1 << 11) +#define RT5670_IF1_ADC1_IN2_SFT 11 +#define RT5670_IF1_ADC2_IN1_SEL (0x1 << 10) +#define RT5670_IF1_ADC2_IN1_SFT 10 + +/* General Control2 (0xfb) */ +#define RT5670_RXDC_SRC_MASK (0x1 << 7) +#define RT5670_RXDC_SRC_STO (0x0 << 7) +#define RT5670_RXDC_SRC_MONO (0x1 << 7) +#define RT5670_RXDC_SRC_SFT (7) +#define RT5670_RXDP2_SEL_MASK (0x1 << 3) +#define RT5670_RXDP2_SEL_IF2 (0x0 << 3) +#define RT5670_RXDP2_SEL_ADC (0x1 << 3) +#define RT5670_RXDP2_SEL_SFT (3) + +/* System Clock Source */ +enum { + RT5670_SCLK_S_MCLK, + RT5670_SCLK_S_PLL1, + RT5670_SCLK_S_RCCLK, +}; + +/* PLL1 Source */ +enum { + RT5670_PLL1_S_MCLK, + RT5670_PLL1_S_BCLK1, + RT5670_PLL1_S_BCLK2, + RT5670_PLL1_S_BCLK3, + RT5670_PLL1_S_BCLK4, +}; + +enum { + RT5670_AIF1, + RT5670_AIF2, + RT5670_AIF3, + RT5670_AIF4, + RT5670_AIFS, +}; + +enum { + RT5670_DMIC_DATA_GPIO6, + RT5670_DMIC_DATA_IN2P, + RT5670_DMIC_DATA_GPIO7, +}; + +enum { + RT5670_DMIC_DATA_GPIO8, + RT5670_DMIC_DATA_IN3N, +}; + +enum { + RT5670_DMIC_DATA_GPIO9, + RT5670_DMIC_DATA_GPIO10, + RT5670_DMIC_DATA_GPIO5, +}; + +struct rt5670_priv { + struct snd_soc_codec *codec; + struct rt5670_platform_data pdata; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck[RT5670_AIFS]; + int bclk[RT5670_AIFS]; + int master[RT5670_AIFS]; + + int pll_src; + int pll_in; + int pll_out; + + int dsp_sw; /* expected parameter setting */ + int dsp_rate; + int jack_type; +}; + +#endif /* __RT5670_H__ */ diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 833231e27340..67f14556462f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -27,6 +27,7 @@ #include <sound/initval.h> #include <sound/tlv.h> +#include "rl6231.h" #include "rt5677.h" #define RT5677_DEVICE_ID 0x6327 @@ -604,19 +605,19 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { adc_vol_tlv), /* ADC Boost Volume Control */ - SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5677_STO1_2_ADC_BST, + SOC_DOUBLE_TLV("STO1 ADC Boost Volume", RT5677_STO1_2_ADC_BST, RT5677_STO1_ADC_L_BST_SFT, RT5677_STO1_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), - SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5677_STO1_2_ADC_BST, + SOC_DOUBLE_TLV("STO2 ADC Boost Volume", RT5677_STO1_2_ADC_BST, RT5677_STO2_ADC_L_BST_SFT, RT5677_STO2_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), - SOC_DOUBLE_TLV("STO3 ADC Boost Gain", RT5677_STO3_4_ADC_BST, + SOC_DOUBLE_TLV("STO3 ADC Boost Volume", RT5677_STO3_4_ADC_BST, RT5677_STO3_ADC_L_BST_SFT, RT5677_STO3_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), - SOC_DOUBLE_TLV("STO4 ADC Boost Gain", RT5677_STO3_4_ADC_BST, + SOC_DOUBLE_TLV("STO4 ADC Boost Volume", RT5677_STO3_4_ADC_BST, RT5677_STO4_ADC_L_BST_SFT, RT5677_STO4_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), - SOC_DOUBLE_TLV("Mono ADC Boost Gain", RT5677_ADC_BST_CTRL2, + SOC_DOUBLE_TLV("Mono ADC Boost Volume", RT5677_ADC_BST_CTRL2, RT5677_MONO_ADC_L_BST_SFT, RT5677_MONO_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), }; @@ -636,21 +637,7 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - int div[] = {2, 3, 4, 6, 8, 12}, idx = -EINVAL, i; - int rate, red, bound, temp; - - rate = rt5677->sysclk; - red = 3000000 * 12; - for (i = 0; i < ARRAY_SIZE(div); i++) { - bound = div[i] * 3000000; - if (rate > bound) - continue; - temp = bound - rate; - if (temp < red) { - red = temp; - idx = i; - } - } + int idx = rl6231_calc_dmic_clk(rt5677->sysclk); if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); @@ -951,7 +938,7 @@ static const struct snd_kcontrol_new rt5677_ob_7_mix[] = { /* Mux */ -/* DAC1 L/R source */ /* MX-29 [10:8] */ +/* DAC1 L/R Source */ /* MX-29 [10:8] */ static const char * const rt5677_dac1_src[] = { "IF1 DAC 01", "IF2 DAC 01", "IF3 DAC LR", "IF4 DAC LR", "SLB DAC 01", "OB 01" @@ -962,9 +949,9 @@ static SOC_ENUM_SINGLE_DECL( RT5677_DAC1_L_SEL_SFT, rt5677_dac1_src); static const struct snd_kcontrol_new rt5677_dac1_mux = - SOC_DAPM_ENUM("DAC1 source", rt5677_dac1_enum); + SOC_DAPM_ENUM("DAC1 Source", rt5677_dac1_enum); -/* ADDA1 L/R source */ /* MX-29 [1:0] */ +/* ADDA1 L/R Source */ /* MX-29 [1:0] */ static const char * const rt5677_adda1_src[] = { "STO1 ADC MIX", "STO2 ADC MIX", "OB 67", }; @@ -974,10 +961,10 @@ static SOC_ENUM_SINGLE_DECL( RT5677_ADDA1_SEL_SFT, rt5677_adda1_src); static const struct snd_kcontrol_new rt5677_adda1_mux = - SOC_DAPM_ENUM("ADDA1 source", rt5677_adda1_enum); + SOC_DAPM_ENUM("ADDA1 Source", rt5677_adda1_enum); -/*DAC2 L/R source*/ /* MX-1B [6:4] [2:0] */ +/*DAC2 L/R Source*/ /* MX-1B [6:4] [2:0] */ static const char * const rt5677_dac2l_src[] = { "IF1 DAC 2", "IF2 DAC 2", "IF3 DAC L", "IF4 DAC L", "SLB DAC 2", "OB 2", @@ -988,7 +975,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_DAC2_L_SRC_SFT, rt5677_dac2l_src); static const struct snd_kcontrol_new rt5677_dac2_l_mux = - SOC_DAPM_ENUM("DAC2 L source", rt5677_dac2l_enum); + SOC_DAPM_ENUM("DAC2 L Source", rt5677_dac2l_enum); static const char * const rt5677_dac2r_src[] = { "IF1 DAC 3", "IF2 DAC 3", "IF3 DAC R", "IF4 DAC R", "SLB DAC 3", @@ -1000,9 +987,9 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_DAC2_R_SRC_SFT, rt5677_dac2r_src); static const struct snd_kcontrol_new rt5677_dac2_r_mux = - SOC_DAPM_ENUM("DAC2 R source", rt5677_dac2r_enum); + SOC_DAPM_ENUM("DAC2 R Source", rt5677_dac2r_enum); -/*DAC3 L/R source*/ /* MX-16 [6:4] [2:0] */ +/*DAC3 L/R Source*/ /* MX-16 [6:4] [2:0] */ static const char * const rt5677_dac3l_src[] = { "IF1 DAC 4", "IF2 DAC 4", "IF3 DAC L", "IF4 DAC L", "SLB DAC 4", "OB 4" @@ -1013,7 +1000,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_DAC3_L_SRC_SFT, rt5677_dac3l_src); static const struct snd_kcontrol_new rt5677_dac3_l_mux = - SOC_DAPM_ENUM("DAC3 L source", rt5677_dac3l_enum); + SOC_DAPM_ENUM("DAC3 L Source", rt5677_dac3l_enum); static const char * const rt5677_dac3r_src[] = { "IF1 DAC 5", "IF2 DAC 5", "IF3 DAC R", "IF4 DAC R", @@ -1025,9 +1012,9 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_DAC3_R_SRC_SFT, rt5677_dac3r_src); static const struct snd_kcontrol_new rt5677_dac3_r_mux = - SOC_DAPM_ENUM("DAC3 R source", rt5677_dac3r_enum); + SOC_DAPM_ENUM("DAC3 R Source", rt5677_dac3r_enum); -/*DAC4 L/R source*/ /* MX-16 [14:12] [10:8] */ +/*DAC4 L/R Source*/ /* MX-16 [14:12] [10:8] */ static const char * const rt5677_dac4l_src[] = { "IF1 DAC 6", "IF2 DAC 6", "IF3 DAC L", "IF4 DAC L", "SLB DAC 6", "OB 6" @@ -1038,7 +1025,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_DAC4_L_SRC_SFT, rt5677_dac4l_src); static const struct snd_kcontrol_new rt5677_dac4_l_mux = - SOC_DAPM_ENUM("DAC4 L source", rt5677_dac4l_enum); + SOC_DAPM_ENUM("DAC4 L Source", rt5677_dac4l_enum); static const char * const rt5677_dac4r_src[] = { "IF1 DAC 7", "IF2 DAC 7", "IF3 DAC R", "IF4 DAC R", @@ -1050,7 +1037,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_DAC4_R_SRC_SFT, rt5677_dac4r_src); static const struct snd_kcontrol_new rt5677_dac4_r_mux = - SOC_DAPM_ENUM("DAC4 R source", rt5677_dac4r_enum); + SOC_DAPM_ENUM("DAC4 R Source", rt5677_dac4r_enum); /* In/OutBound Source Pass SRC */ /* MX-A5 [3] [4] [0] [1] [2] */ static const char * const rt5677_iob_bypass_src[] = { @@ -1062,35 +1049,35 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_SRC_OB01_SFT, rt5677_iob_bypass_src); static const struct snd_kcontrol_new rt5677_ob01_bypass_src_mux = - SOC_DAPM_ENUM("OB01 Bypass source", rt5677_ob01_bypass_src_enum); + SOC_DAPM_ENUM("OB01 Bypass Source", rt5677_ob01_bypass_src_enum); static SOC_ENUM_SINGLE_DECL( rt5677_ob23_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL, RT5677_SEL_SRC_OB23_SFT, rt5677_iob_bypass_src); static const struct snd_kcontrol_new rt5677_ob23_bypass_src_mux = - SOC_DAPM_ENUM("OB23 Bypass source", rt5677_ob23_bypass_src_enum); + SOC_DAPM_ENUM("OB23 Bypass Source", rt5677_ob23_bypass_src_enum); static SOC_ENUM_SINGLE_DECL( rt5677_ib01_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL, RT5677_SEL_SRC_IB01_SFT, rt5677_iob_bypass_src); static const struct snd_kcontrol_new rt5677_ib01_bypass_src_mux = - SOC_DAPM_ENUM("IB01 Bypass source", rt5677_ib01_bypass_src_enum); + SOC_DAPM_ENUM("IB01 Bypass Source", rt5677_ib01_bypass_src_enum); static SOC_ENUM_SINGLE_DECL( rt5677_ib23_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL, RT5677_SEL_SRC_IB23_SFT, rt5677_iob_bypass_src); static const struct snd_kcontrol_new rt5677_ib23_bypass_src_mux = - SOC_DAPM_ENUM("IB23 Bypass source", rt5677_ib23_bypass_src_enum); + SOC_DAPM_ENUM("IB23 Bypass Source", rt5677_ib23_bypass_src_enum); static SOC_ENUM_SINGLE_DECL( rt5677_ib45_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL, RT5677_SEL_SRC_IB45_SFT, rt5677_iob_bypass_src); static const struct snd_kcontrol_new rt5677_ib45_bypass_src_mux = - SOC_DAPM_ENUM("IB45 Bypass source", rt5677_ib45_bypass_src_enum); + SOC_DAPM_ENUM("IB45 Bypass Source", rt5677_ib45_bypass_src_enum); /* Stereo ADC Source 2 */ /* MX-27 MX26 MX25 [11:10] */ static const char * const rt5677_stereo_adc2_src[] = { @@ -1102,21 +1089,21 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_STO1_ADC2_SFT, rt5677_stereo_adc2_src); static const struct snd_kcontrol_new rt5677_sto1_adc2_mux = - SOC_DAPM_ENUM("Stereo1 ADC2 source", rt5677_stereo1_adc2_enum); + SOC_DAPM_ENUM("Stereo1 ADC2 Source", rt5677_stereo1_adc2_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo2_adc2_enum, RT5677_STO2_ADC_MIXER, RT5677_SEL_STO2_ADC2_SFT, rt5677_stereo_adc2_src); static const struct snd_kcontrol_new rt5677_sto2_adc2_mux = - SOC_DAPM_ENUM("Stereo2 ADC2 source", rt5677_stereo2_adc2_enum); + SOC_DAPM_ENUM("Stereo2 ADC2 Source", rt5677_stereo2_adc2_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo3_adc2_enum, RT5677_STO3_ADC_MIXER, RT5677_SEL_STO3_ADC2_SFT, rt5677_stereo_adc2_src); static const struct snd_kcontrol_new rt5677_sto3_adc2_mux = - SOC_DAPM_ENUM("Stereo3 ADC2 source", rt5677_stereo3_adc2_enum); + SOC_DAPM_ENUM("Stereo3 ADC2 Source", rt5677_stereo3_adc2_enum); /* DMIC Source */ /* MX-28 [9:8][1:0] MX-27 MX-26 MX-25 MX-24 [9:8] */ static const char * const rt5677_dmic_src[] = { @@ -1128,44 +1115,44 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_MONO_DMIC_L_SFT, rt5677_dmic_src); static const struct snd_kcontrol_new rt5677_mono_dmic_l_mux = - SOC_DAPM_ENUM("Mono DMIC L source", rt5677_mono_dmic_l_enum); + SOC_DAPM_ENUM("Mono DMIC L Source", rt5677_mono_dmic_l_enum); static SOC_ENUM_SINGLE_DECL( rt5677_mono_dmic_r_enum, RT5677_MONO_ADC_MIXER, RT5677_SEL_MONO_DMIC_R_SFT, rt5677_dmic_src); static const struct snd_kcontrol_new rt5677_mono_dmic_r_mux = - SOC_DAPM_ENUM("Mono DMIC R source", rt5677_mono_dmic_r_enum); + SOC_DAPM_ENUM("Mono DMIC R Source", rt5677_mono_dmic_r_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo1_dmic_enum, RT5677_STO1_ADC_MIXER, RT5677_SEL_STO1_DMIC_SFT, rt5677_dmic_src); static const struct snd_kcontrol_new rt5677_sto1_dmic_mux = - SOC_DAPM_ENUM("Stereo1 DMIC source", rt5677_stereo1_dmic_enum); + SOC_DAPM_ENUM("Stereo1 DMIC Source", rt5677_stereo1_dmic_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo2_dmic_enum, RT5677_STO2_ADC_MIXER, RT5677_SEL_STO2_DMIC_SFT, rt5677_dmic_src); static const struct snd_kcontrol_new rt5677_sto2_dmic_mux = - SOC_DAPM_ENUM("Stereo2 DMIC source", rt5677_stereo2_dmic_enum); + SOC_DAPM_ENUM("Stereo2 DMIC Source", rt5677_stereo2_dmic_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo3_dmic_enum, RT5677_STO3_ADC_MIXER, RT5677_SEL_STO3_DMIC_SFT, rt5677_dmic_src); static const struct snd_kcontrol_new rt5677_sto3_dmic_mux = - SOC_DAPM_ENUM("Stereo3 DMIC source", rt5677_stereo3_dmic_enum); + SOC_DAPM_ENUM("Stereo3 DMIC Source", rt5677_stereo3_dmic_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo4_dmic_enum, RT5677_STO4_ADC_MIXER, RT5677_SEL_STO4_DMIC_SFT, rt5677_dmic_src); static const struct snd_kcontrol_new rt5677_sto4_dmic_mux = - SOC_DAPM_ENUM("Stereo4 DMIC source", rt5677_stereo4_dmic_enum); + SOC_DAPM_ENUM("Stereo4 DMIC Source", rt5677_stereo4_dmic_enum); -/* Stereo2 ADC source */ /* MX-26 [0] */ +/* Stereo2 ADC Source */ /* MX-26 [0] */ static const char * const rt5677_stereo2_adc_lr_src[] = { "L", "LR" }; @@ -1175,7 +1162,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_STO2_LR_MIX_SFT, rt5677_stereo2_adc_lr_src); static const struct snd_kcontrol_new rt5677_sto2_adc_lr_mux = - SOC_DAPM_ENUM("Stereo2 ADC LR source", rt5677_stereo2_adc_lr_enum); + SOC_DAPM_ENUM("Stereo2 ADC LR Source", rt5677_stereo2_adc_lr_enum); /* Stereo1 ADC Source 1 */ /* MX-27 MX26 MX25 [13:12] */ static const char * const rt5677_stereo_adc1_src[] = { @@ -1187,23 +1174,23 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_STO1_ADC1_SFT, rt5677_stereo_adc1_src); static const struct snd_kcontrol_new rt5677_sto1_adc1_mux = - SOC_DAPM_ENUM("Stereo1 ADC1 source", rt5677_stereo1_adc1_enum); + SOC_DAPM_ENUM("Stereo1 ADC1 Source", rt5677_stereo1_adc1_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo2_adc1_enum, RT5677_STO2_ADC_MIXER, RT5677_SEL_STO2_ADC1_SFT, rt5677_stereo_adc1_src); static const struct snd_kcontrol_new rt5677_sto2_adc1_mux = - SOC_DAPM_ENUM("Stereo2 ADC1 source", rt5677_stereo2_adc1_enum); + SOC_DAPM_ENUM("Stereo2 ADC1 Source", rt5677_stereo2_adc1_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo3_adc1_enum, RT5677_STO3_ADC_MIXER, RT5677_SEL_STO3_ADC1_SFT, rt5677_stereo_adc1_src); static const struct snd_kcontrol_new rt5677_sto3_adc1_mux = - SOC_DAPM_ENUM("Stereo3 ADC1 source", rt5677_stereo3_adc1_enum); + SOC_DAPM_ENUM("Stereo3 ADC1 Source", rt5677_stereo3_adc1_enum); -/* Mono ADC Left source 2 */ /* MX-28 [11:10] */ +/* Mono ADC Left Source 2 */ /* MX-28 [11:10] */ static const char * const rt5677_mono_adc2_l_src[] = { "DD MIX1L", "DMIC", "MONO DAC MIXL" }; @@ -1213,9 +1200,9 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_MONO_ADC_L2_SFT, rt5677_mono_adc2_l_src); static const struct snd_kcontrol_new rt5677_mono_adc2_l_mux = - SOC_DAPM_ENUM("Mono ADC2 L source", rt5677_mono_adc2_l_enum); + SOC_DAPM_ENUM("Mono ADC2 L Source", rt5677_mono_adc2_l_enum); -/* Mono ADC Left source 1 */ /* MX-28 [13:12] */ +/* Mono ADC Left Source 1 */ /* MX-28 [13:12] */ static const char * const rt5677_mono_adc1_l_src[] = { "DD MIX1L", "ADC1", "MONO DAC MIXL" }; @@ -1225,9 +1212,9 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_MONO_ADC_L1_SFT, rt5677_mono_adc1_l_src); static const struct snd_kcontrol_new rt5677_mono_adc1_l_mux = - SOC_DAPM_ENUM("Mono ADC1 L source", rt5677_mono_adc1_l_enum); + SOC_DAPM_ENUM("Mono ADC1 L Source", rt5677_mono_adc1_l_enum); -/* Mono ADC Right source 2 */ /* MX-28 [3:2] */ +/* Mono ADC Right Source 2 */ /* MX-28 [3:2] */ static const char * const rt5677_mono_adc2_r_src[] = { "DD MIX1R", "DMIC", "MONO DAC MIXR" }; @@ -1237,9 +1224,9 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_MONO_ADC_R2_SFT, rt5677_mono_adc2_r_src); static const struct snd_kcontrol_new rt5677_mono_adc2_r_mux = - SOC_DAPM_ENUM("Mono ADC2 R source", rt5677_mono_adc2_r_enum); + SOC_DAPM_ENUM("Mono ADC2 R Source", rt5677_mono_adc2_r_enum); -/* Mono ADC Right source 1 */ /* MX-28 [5:4] */ +/* Mono ADC Right Source 1 */ /* MX-28 [5:4] */ static const char * const rt5677_mono_adc1_r_src[] = { "DD MIX1R", "ADC2", "MONO DAC MIXR" }; @@ -1249,7 +1236,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_MONO_ADC_R1_SFT, rt5677_mono_adc1_r_src); static const struct snd_kcontrol_new rt5677_mono_adc1_r_mux = - SOC_DAPM_ENUM("Mono ADC1 R source", rt5677_mono_adc1_r_enum); + SOC_DAPM_ENUM("Mono ADC1 R Source", rt5677_mono_adc1_r_enum); /* Stereo4 ADC Source 2 */ /* MX-24 [11:10] */ static const char * const rt5677_stereo4_adc2_src[] = { @@ -1261,7 +1248,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_STO4_ADC2_SFT, rt5677_stereo4_adc2_src); static const struct snd_kcontrol_new rt5677_sto4_adc2_mux = - SOC_DAPM_ENUM("Stereo4 ADC2 source", rt5677_stereo4_adc2_enum); + SOC_DAPM_ENUM("Stereo4 ADC2 Source", rt5677_stereo4_adc2_enum); /* Stereo4 ADC Source 1 */ /* MX-24 [13:12] */ @@ -1274,7 +1261,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_STO4_ADC1_SFT, rt5677_stereo4_adc1_src); static const struct snd_kcontrol_new rt5677_sto4_adc1_mux = - SOC_DAPM_ENUM("Stereo4 ADC1 source", rt5677_stereo4_adc1_enum); + SOC_DAPM_ENUM("Stereo4 ADC1 Source", rt5677_stereo4_adc1_enum); /* InBound0/1 Source */ /* MX-A3 [14:12] */ static const char * const rt5677_inbound01_src[] = { @@ -1416,7 +1403,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_dac3_mux = SOC_DAPM_ENUM("Analog DAC3 Source", rt5677_dac3_enum); -/* PDM channel source */ /* MX-31 [13:12][9:8][5:4][1:0] */ +/* PDM channel Source */ /* MX-31 [13:12][9:8][5:4][1:0] */ static const char * const rt5677_pdm_src[] = { "STO1 DAC MIX", "MONO DAC MIX", "DD MIX1", "DD MIX2" }; @@ -1426,28 +1413,28 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_PDM1_L_SFT, rt5677_pdm_src); static const struct snd_kcontrol_new rt5677_pdm1_l_mux = - SOC_DAPM_ENUM("PDM1 source", rt5677_pdm1_l_enum); + SOC_DAPM_ENUM("PDM1 Source", rt5677_pdm1_l_enum); static SOC_ENUM_SINGLE_DECL( rt5677_pdm2_l_enum, RT5677_PDM_OUT_CTRL, RT5677_SEL_PDM2_L_SFT, rt5677_pdm_src); static const struct snd_kcontrol_new rt5677_pdm2_l_mux = - SOC_DAPM_ENUM("PDM2 source", rt5677_pdm2_l_enum); + SOC_DAPM_ENUM("PDM2 Source", rt5677_pdm2_l_enum); static SOC_ENUM_SINGLE_DECL( rt5677_pdm1_r_enum, RT5677_PDM_OUT_CTRL, RT5677_SEL_PDM1_R_SFT, rt5677_pdm_src); static const struct snd_kcontrol_new rt5677_pdm1_r_mux = - SOC_DAPM_ENUM("PDM1 source", rt5677_pdm1_r_enum); + SOC_DAPM_ENUM("PDM1 Source", rt5677_pdm1_r_enum); static SOC_ENUM_SINGLE_DECL( rt5677_pdm2_r_enum, RT5677_PDM_OUT_CTRL, RT5677_SEL_PDM2_R_SFT, rt5677_pdm_src); static const struct snd_kcontrol_new rt5677_pdm2_r_mux = - SOC_DAPM_ENUM("PDM2 source", rt5677_pdm2_r_enum); + SOC_DAPM_ENUM("PDM2 Source", rt5677_pdm2_r_enum); /* TDM IF1/2 SLB ADC1 Data Selection */ /* MX-3C MX-41 [5:4] MX-08 [1:0]*/ static const char * const rt5677_if12_adc1_src[] = { @@ -1459,21 +1446,21 @@ static SOC_ENUM_SINGLE_DECL( RT5677_IF1_ADC1_SFT, rt5677_if12_adc1_src); static const struct snd_kcontrol_new rt5677_if1_adc1_mux = - SOC_DAPM_ENUM("IF1 ADC1 source", rt5677_if1_adc1_enum); + SOC_DAPM_ENUM("IF1 ADC1 Source", rt5677_if1_adc1_enum); static SOC_ENUM_SINGLE_DECL( rt5677_if2_adc1_enum, RT5677_TDM2_CTRL2, RT5677_IF2_ADC1_SFT, rt5677_if12_adc1_src); static const struct snd_kcontrol_new rt5677_if2_adc1_mux = - SOC_DAPM_ENUM("IF2 ADC1 source", rt5677_if2_adc1_enum); + SOC_DAPM_ENUM("IF2 ADC1 Source", rt5677_if2_adc1_enum); static SOC_ENUM_SINGLE_DECL( rt5677_slb_adc1_enum, RT5677_SLIMBUS_RX, RT5677_SLB_ADC1_SFT, rt5677_if12_adc1_src); static const struct snd_kcontrol_new rt5677_slb_adc1_mux = - SOC_DAPM_ENUM("SLB ADC1 source", rt5677_slb_adc1_enum); + SOC_DAPM_ENUM("SLB ADC1 Source", rt5677_slb_adc1_enum); /* TDM IF1/2 SLB ADC2 Data Selection */ /* MX-3C MX-41 [7:6] MX-08 [3:2] */ static const char * const rt5677_if12_adc2_src[] = { @@ -1485,21 +1472,21 @@ static SOC_ENUM_SINGLE_DECL( RT5677_IF1_ADC2_SFT, rt5677_if12_adc2_src); static const struct snd_kcontrol_new rt5677_if1_adc2_mux = - SOC_DAPM_ENUM("IF1 ADC2 source", rt5677_if1_adc2_enum); + SOC_DAPM_ENUM("IF1 ADC2 Source", rt5677_if1_adc2_enum); static SOC_ENUM_SINGLE_DECL( rt5677_if2_adc2_enum, RT5677_TDM2_CTRL2, RT5677_IF2_ADC2_SFT, rt5677_if12_adc2_src); static const struct snd_kcontrol_new rt5677_if2_adc2_mux = - SOC_DAPM_ENUM("IF2 ADC2 source", rt5677_if2_adc2_enum); + SOC_DAPM_ENUM("IF2 ADC2 Source", rt5677_if2_adc2_enum); static SOC_ENUM_SINGLE_DECL( rt5677_slb_adc2_enum, RT5677_SLIMBUS_RX, RT5677_SLB_ADC2_SFT, rt5677_if12_adc2_src); static const struct snd_kcontrol_new rt5677_slb_adc2_mux = - SOC_DAPM_ENUM("SLB ADC2 source", rt5677_slb_adc2_enum); + SOC_DAPM_ENUM("SLB ADC2 Source", rt5677_slb_adc2_enum); /* TDM IF1/2 SLB ADC3 Data Selection */ /* MX-3C MX-41 [9:8] MX-08 [5:4] */ static const char * const rt5677_if12_adc3_src[] = { @@ -1511,21 +1498,21 @@ static SOC_ENUM_SINGLE_DECL( RT5677_IF1_ADC3_SFT, rt5677_if12_adc3_src); static const struct snd_kcontrol_new rt5677_if1_adc3_mux = - SOC_DAPM_ENUM("IF1 ADC3 source", rt5677_if1_adc3_enum); + SOC_DAPM_ENUM("IF1 ADC3 Source", rt5677_if1_adc3_enum); static SOC_ENUM_SINGLE_DECL( rt5677_if2_adc3_enum, RT5677_TDM2_CTRL2, RT5677_IF2_ADC3_SFT, rt5677_if12_adc3_src); static const struct snd_kcontrol_new rt5677_if2_adc3_mux = - SOC_DAPM_ENUM("IF2 ADC3 source", rt5677_if2_adc3_enum); + SOC_DAPM_ENUM("IF2 ADC3 Source", rt5677_if2_adc3_enum); static SOC_ENUM_SINGLE_DECL( rt5677_slb_adc3_enum, RT5677_SLIMBUS_RX, RT5677_SLB_ADC3_SFT, rt5677_if12_adc3_src); static const struct snd_kcontrol_new rt5677_slb_adc3_mux = - SOC_DAPM_ENUM("SLB ADC3 source", rt5677_slb_adc3_enum); + SOC_DAPM_ENUM("SLB ADC3 Source", rt5677_slb_adc3_enum); /* TDM IF1/2 SLB ADC4 Data Selection */ /* MX-3C MX-41 [11:10] MX-08 [7:6] */ static const char * const rt5677_if12_adc4_src[] = { @@ -1537,21 +1524,21 @@ static SOC_ENUM_SINGLE_DECL( RT5677_IF1_ADC4_SFT, rt5677_if12_adc4_src); static const struct snd_kcontrol_new rt5677_if1_adc4_mux = - SOC_DAPM_ENUM("IF1 ADC4 source", rt5677_if1_adc4_enum); + SOC_DAPM_ENUM("IF1 ADC4 Source", rt5677_if1_adc4_enum); static SOC_ENUM_SINGLE_DECL( rt5677_if2_adc4_enum, RT5677_TDM2_CTRL2, RT5677_IF2_ADC4_SFT, rt5677_if12_adc4_src); static const struct snd_kcontrol_new rt5677_if2_adc4_mux = - SOC_DAPM_ENUM("IF2 ADC4 source", rt5677_if2_adc4_enum); + SOC_DAPM_ENUM("IF2 ADC4 Source", rt5677_if2_adc4_enum); static SOC_ENUM_SINGLE_DECL( rt5677_slb_adc4_enum, RT5677_SLIMBUS_RX, RT5677_SLB_ADC4_SFT, rt5677_if12_adc4_src); static const struct snd_kcontrol_new rt5677_slb_adc4_mux = - SOC_DAPM_ENUM("SLB ADC4 source", rt5677_slb_adc4_enum); + SOC_DAPM_ENUM("SLB ADC4 Source", rt5677_slb_adc4_enum); /* Interface3/4 ADC Data Input */ /* MX-2F [3:0] MX-30 [7:4]*/ static const char * const rt5677_if34_adc_src[] = { @@ -1564,14 +1551,14 @@ static SOC_ENUM_SINGLE_DECL( RT5677_IF3_ADC_IN_SFT, rt5677_if34_adc_src); static const struct snd_kcontrol_new rt5677_if3_adc_mux = - SOC_DAPM_ENUM("IF3 ADC source", rt5677_if3_adc_enum); + SOC_DAPM_ENUM("IF3 ADC Source", rt5677_if3_adc_enum); static SOC_ENUM_SINGLE_DECL( rt5677_if4_adc_enum, RT5677_IF4_DATA, RT5677_IF4_ADC_IN_SFT, rt5677_if34_adc_src); static const struct snd_kcontrol_new rt5677_if4_adc_mux = - SOC_DAPM_ENUM("IF4 ADC source", rt5677_if4_adc_enum); + SOC_DAPM_ENUM("IF4 ADC Source", rt5677_if4_adc_enum); static int rt5677_bst1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) @@ -1670,6 +1657,13 @@ static int rt5677_set_micbias1_event(struct snd_soc_dapm_widget *w, RT5677_PWR_CLK_MB, RT5677_PWR_CLK_MB1 | RT5677_PWR_PP_MB1 | RT5677_PWR_CLK_MB); break; + + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, + RT5677_PWR_CLK_MB1 | RT5677_PWR_PP_MB1 | + RT5677_PWR_CLK_MB, 0); + break; + default: return 0; } @@ -1685,8 +1679,9 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { /* Input Side */ /* micbias */ - SND_SOC_DAPM_SUPPLY("micbias1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT, - 0, rt5677_set_micbias1_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT, + 0, rt5677_set_micbias1_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), /* Input Lines */ SND_SOC_DAPM_INPUT("DMIC L1"), @@ -2798,21 +2793,6 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "PDM2R", NULL, "PDM2 R Mux" }, }; -static int get_clk_info(int sclk, int rate) -{ - int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; - - if (sclk <= 0 || rate <= 0) - return -EINVAL; - - rate = rate << 8; - for (i = 0; i < ARRAY_SIZE(pd); i++) - if (sclk == rate * pd[i]) - return i; - - return -EINVAL; -} - static int rt5677_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -2822,7 +2802,7 @@ static int rt5677_hw_params(struct snd_pcm_substream *substream, int pre_div, bclk_ms, frame_size; rt5677->lrck[dai->id] = params_rate(params); - pre_div = get_clk_info(rt5677->sysclk, rt5677->lrck[dai->id]); + pre_div = rl6231_get_clk_info(rt5677->sysclk, rt5677->lrck[dai->id]); if (pre_div < 0) { dev_err(codec->dev, "Unsupported clock setting\n"); return -EINVAL; @@ -3016,62 +2996,12 @@ static int rt5677_set_dai_sysclk(struct snd_soc_dai *dai, * Returns 0 for success or negative error code. */ static int rt5677_pll_calc(const unsigned int freq_in, - const unsigned int freq_out, struct rt5677_pll_code *pll_code) + const unsigned int freq_out, struct rl6231_pll_code *pll_code) { - int max_n = RT5677_PLL_N_MAX, max_m = RT5677_PLL_M_MAX; - int k, red, n_t, pll_out, in_t; - int n = 0, m = 0, m_t = 0; - int out_t, red_t = abs(freq_out - freq_in); - bool m_bp = false, k_bp = false; - - if (RT5677_PLL_INP_MAX < freq_in || RT5677_PLL_INP_MIN > freq_in) + if (RT5677_PLL_INP_MIN > freq_in) return -EINVAL; - k = 100000000 / freq_out - 2; - if (k > RT5677_PLL_K_MAX) - k = RT5677_PLL_K_MAX; - for (n_t = 0; n_t <= max_n; n_t++) { - in_t = freq_in / (k + 2); - pll_out = freq_out / (n_t + 2); - if (in_t < 0) - continue; - if (in_t == pll_out) { - m_bp = true; - n = n_t; - goto code_find; - } - red = abs(in_t - pll_out); - if (red < red_t) { - m_bp = true; - n = n_t; - m = m_t; - if (red == 0) - goto code_find; - red_t = red; - } - for (m_t = 0; m_t <= max_m; m_t++) { - out_t = in_t / (m_t + 2); - red = abs(out_t - pll_out); - if (red < red_t) { - m_bp = false; - n = n_t; - m = m_t; - if (red == 0) - goto code_find; - red_t = red; - } - } - } - pr_debug("Only get approximation about PLL\n"); - -code_find: - - pll_code->m_bp = m_bp; - pll_code->k_bp = k_bp; - pll_code->m_code = m; - pll_code->n_code = n; - pll_code->k_code = k; - return 0; + return rl6231_pll_calc(freq_in, freq_out, pll_code); } static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, @@ -3079,7 +3009,7 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, { struct snd_soc_codec *codec = dai->codec; struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - struct rt5677_pll_code pll_code; + struct rl6231_pll_code pll_code; int ret; if (source == rt5677->pll_src && freq_in == rt5677->pll_in && @@ -3137,15 +3067,12 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, return ret; } - dev_dbg(codec->dev, "m_bypass=%d k_bypass=%d m=%d n=%d k=%d\n", - pll_code.m_bp, pll_code.k_bp, - (pll_code.m_bp ? 0 : pll_code.m_code), pll_code.n_code, - (pll_code.k_bp ? 0 : pll_code.k_code)); + dev_dbg(codec->dev, "m_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); regmap_write(rt5677->regmap, RT5677_PLL1_CTRL1, - pll_code.n_code << RT5677_PLL_N_SFT | - pll_code.k_bp << RT5677_PLL_K_BP_SFT | - (pll_code.k_bp ? 0 : pll_code.k_code)); + pll_code.n_code << RT5677_PLL_N_SFT | pll_code.k_code); regmap_write(rt5677->regmap, RT5677_PLL1_CTRL2, (pll_code.m_bp ? 0 : pll_code.m_code) << RT5677_PLL_M_SFT | pll_code.m_bp << RT5677_PLL_M_BP_SFT); @@ -3197,7 +3124,7 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec, regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x1, 0x0); regmap_write(rt5677->regmap, RT5677_PWR_DIG1, 0x0000); regmap_write(rt5677->regmap, RT5677_PWR_DIG2, 0x0000); - regmap_write(rt5677->regmap, RT5677_PWR_ANLG1, 0x0000); + regmap_write(rt5677->regmap, RT5677_PWR_ANLG1, 0x0022); regmap_write(rt5677->regmap, RT5677_PWR_ANLG2, 0x0000); regmap_update_bits(rt5677->regmap, RT5677_PR_BASE + RT5677_BIAS_CUR4, 0x0f00, 0x0000); @@ -3454,14 +3381,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5677->regmap, RT5677_IN1, RT5677_IN_DF2, RT5677_IN_DF2); - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677, - rt5677_dai, ARRAY_SIZE(rt5677_dai)); - if (ret < 0) - goto err; - - return 0; -err: - return ret; + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677, + rt5677_dai, ARRAY_SIZE(rt5677_dai)); } static int rt5677_i2c_remove(struct i2c_client *i2c) @@ -3480,18 +3401,7 @@ static struct i2c_driver rt5677_i2c_driver = { .remove = rt5677_i2c_remove, .id_table = rt5677_i2c_id, }; - -static int __init rt5677_modinit(void) -{ - return i2c_add_driver(&rt5677_i2c_driver); -} -module_init(rt5677_modinit); - -static void __exit rt5677_modexit(void) -{ - i2c_del_driver(&rt5677_i2c_driver); -} -module_exit(rt5677_modexit); +module_i2c_driver(rt5677_i2c_driver); MODULE_DESCRIPTION("ASoC RT5677 driver"); MODULE_AUTHOR("Oder Chiou <oder_chiou@realtek.com>"); diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index af4e9c797408..863393e62096 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1393,13 +1393,6 @@ #define RT5677_DSP_IB_9_L (0x1 << 1) #define RT5677_DSP_IB_9_L_SFT 1 -/* Debug String Length */ -#define RT5677_REG_DISP_LEN 23 - -#define RT5677_NO_JACK BIT(0) -#define RT5677_HEADSET_DET BIT(1) -#define RT5677_HEADPHO_DET BIT(2) - /* System Clock Source */ enum { RT5677_SCLK_S_MCLK, @@ -1425,14 +1418,6 @@ enum { RT5677_AIFS, }; -struct rt5677_pll_code { - bool m_bp; /* Indicates bypass m code or not. */ - bool k_bp; /* Indicates bypass k code or not. */ - int m_code; - int n_code; - int k_code; -}; - struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 3d39f0b5b4a8..e997d271728d 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -724,25 +724,25 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, return ret; /* set i2s data format */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) return -EINVAL; i2s_ctl |= SGTL5000_I2S_DLEN_16 << SGTL5000_I2S_DLEN_SHIFT; i2s_ctl |= SGTL5000_I2S_SCLKFREQ_32FS << SGTL5000_I2S_SCLKFREQ_SHIFT; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: i2s_ctl |= SGTL5000_I2S_DLEN_20 << SGTL5000_I2S_DLEN_SHIFT; i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << SGTL5000_I2S_SCLKFREQ_SHIFT; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: i2s_ctl |= SGTL5000_I2S_DLEN_24 << SGTL5000_I2S_DLEN_SHIFT; i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << SGTL5000_I2S_SCLKFREQ_SHIFT; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) return -EINVAL; i2s_ctl |= SGTL5000_I2S_DLEN_32 << SGTL5000_I2S_DLEN_SHIFT; @@ -843,10 +843,8 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL); - if (!ldo) { - dev_err(codec->dev, "failed to allocate ldo_regulator\n"); + if (!ldo) return -ENOMEM; - } ldo->desc.name = kstrdup(dev_name(codec->dev), GFP_KERNEL); if (!ldo->desc.name) { @@ -1277,7 +1275,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) return ret; } - ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) goto err_ldo_remove; @@ -1285,13 +1283,16 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) - goto err_ldo_remove; + goto err_regulator_free; /* wait for all power rails bring up */ udelay(10); return 0; +err_regulator_free: + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); err_ldo_remove: if (!external_vddd) ldo_regulator_remove(codec); @@ -1361,6 +1362,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) err: regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); ldo_regulator_remove(codec); return ret; @@ -1374,6 +1377,8 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); ldo_regulator_remove(codec); return 0; diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index f26befb0c297..cdf882fa7716 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -167,17 +167,17 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: + switch (params_width(params)) { + case 8: width = SI476X_PCM_FORMAT_S8; break; - case SNDRV_PCM_FORMAT_S16_LE: + case 16: width = SI476X_PCM_FORMAT_S16_LE; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = SI476X_PCM_FORMAT_S20_3LE; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: width = SI476X_PCM_FORMAT_S24_LE; break; default: diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index d90cb0fafcb2..06ba4923fd5a 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -471,8 +471,8 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev) mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); base = devm_ioremap_resource(&pdev->dev, mem_res); - if (base == NULL) - return -ENOMEM; + if (IS_ERR(base)) + return PTR_ERR(base); sirf_audio_codec->regmap = devm_regmap_init_mmio(&pdev->dev, base, &sirf_audio_codec_regmap_config); diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 42dff26b3a2a..cf8fa40662f0 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -661,12 +661,12 @@ static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, { unsigned int format, rate; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: format = BIT(4)|BIT(5); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: format = 0; break; default: diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c index a078aa31052a..e0df537dd4b7 100644 --- a/sound/soc/codecs/spdif_transmitter.c +++ b/sound/soc/codecs/spdif_transmitter.c @@ -24,7 +24,7 @@ #define DRV_NAME "spdif-dit" -#define STUB_RATES SNDRV_PCM_RATE_8000_96000 +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 56adb3e2def9..e8680bea5f86 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -361,11 +361,11 @@ static int ssm2518_hw_params(struct snd_pcm_substream *substream, return -EINVAL; if (ssm2518->right_j) { - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_16BIT; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; break; default: diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 97b0454eb346..484b3bbe8624 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -275,17 +275,17 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, regmap_write(ssm2602->regmap, SSM2602_SRATE, srate); /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: iface = 0x0; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface = 0x4; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface = 0x8; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface = 0xc; break; default: diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 0579d187135b..48740855566d 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -678,15 +678,11 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, confb = snd_soc_read(codec, STA32X_CONFB); confb &= ~(STA32X_CONFB_SAI_MASK | STA32X_CONFB_SAIFB); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S24_BE: - case SNDRV_PCM_FORMAT_S24_3LE: - case SNDRV_PCM_FORMAT_S24_3BE: + switch (params_width(params)) { + case 24: pr_debug("24bit\n"); /* fall through */ - case SNDRV_PCM_FORMAT_S32_LE: - case SNDRV_PCM_FORMAT_S32_BE: + case 32: pr_debug("24bit or 32bit\n"); switch (sta32x->format) { case SND_SOC_DAIFMT_I2S: @@ -701,8 +697,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, } break; - case SNDRV_PCM_FORMAT_S20_3LE: - case SNDRV_PCM_FORMAT_S20_3BE: + case 20: pr_debug("20bit\n"); switch (sta32x->format) { case SND_SOC_DAIFMT_I2S: @@ -717,8 +712,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, } break; - case SNDRV_PCM_FORMAT_S18_3LE: - case SNDRV_PCM_FORMAT_S18_3BE: + case 18: pr_debug("18bit\n"); switch (sta32x->format) { case SND_SOC_DAIFMT_I2S: @@ -733,8 +727,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, } break; - case SNDRV_PCM_FORMAT_S16_LE: - case SNDRV_PCM_FORMAT_S16_BE: + case 16: pr_debug("16bit\n"); switch (sta32x->format) { case SND_SOC_DAIFMT_I2S: diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index a40c4b0196a3..9aa1323fb2ab 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -197,16 +197,16 @@ static int sta529_hw_params(struct snd_pcm_substream *substream, int pdata, play_freq_val, record_freq_val; int bclk_to_fs_ratio; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: pdata = 1; bclk_to_fs_ratio = 0; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: pdata = 2; bclk_to_fs_ratio = 1; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: pdata = 3; bclk_to_fs_ratio = 2; break; @@ -380,10 +380,8 @@ static int sta529_i2c_probe(struct i2c_client *i2c, return -EINVAL; sta529 = devm_kzalloc(&i2c->dev, sizeof(struct sta529), GFP_KERNEL); - if (sta529 == NULL) { - dev_err(&i2c->dev, "Can not allocate memory\n"); + if (!sta529) return -ENOMEM; - } sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap); if (IS_ERR(sta529->regmap)) { diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c new file mode 100644 index 000000000000..23b32960ff1d --- /dev/null +++ b/sound/soc/codecs/tas2552.c @@ -0,0 +1,544 @@ +/* + * tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy <dmurphy@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. + * + * 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. + */ + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> + +#include <linux/gpio/consumer.h> +#include <linux/regulator/consumer.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/tas2552-plat.h> + +#include "tas2552.h" + +static struct reg_default tas2552_reg_defs[] = { + {TAS2552_CFG_1, 0x22}, + {TAS2552_CFG_3, 0x80}, + {TAS2552_DOUT, 0x00}, + {TAS2552_OUTPUT_DATA, 0xc0}, + {TAS2552_PDM_CFG, 0x01}, + {TAS2552_PGA_GAIN, 0x00}, + {TAS2552_BOOST_PT_CTRL, 0x0f}, + {TAS2552_RESERVED_0D, 0x00}, + {TAS2552_LIMIT_RATE_HYS, 0x08}, + {TAS2552_CFG_2, 0xef}, + {TAS2552_SER_CTRL_1, 0x00}, + {TAS2552_SER_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_1, 0x10}, + {TAS2552_PLL_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_3, 0x00}, + {TAS2552_BTIP, 0x8f}, + {TAS2552_BTS_CTRL, 0x80}, + {TAS2552_LIMIT_RELEASE, 0x04}, + {TAS2552_LIMIT_INT_COUNT, 0x00}, + {TAS2552_EDGE_RATE_CTRL, 0x40}, + {TAS2552_VBAT_DATA, 0x00}, +}; + +#define TAS2552_NUM_SUPPLIES 3 +static const char *tas2552_supply_names[TAS2552_NUM_SUPPLIES] = { + "vbat", /* vbat voltage */ + "iovdd", /* I/O Voltage */ + "avdd", /* Analog DAC Voltage */ +}; + +struct tas2552_data { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct i2c_client *tas2552_client; + struct regulator_bulk_data supplies[TAS2552_NUM_SUPPLIES]; + struct gpio_desc *enable_gpio; + unsigned char regs[TAS2552_VBAT_DATA]; + unsigned int mclk; +}; + +static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) +{ + u8 cfg1_reg; + + if (sw_shutdown) + cfg1_reg = 0; + else + cfg1_reg = TAS2552_SWS_MASK; + + snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1, + TAS2552_SWS_MASK, cfg1_reg); +} + +static int tas2552_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas2552_data *tas2552 = dev_get_drvdata(codec->dev); + int sample_rate, pll_clk; + int d; + u8 p, j; + + /* Turn on Class D amplifier */ + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK, + TAS2552_CLASSD_EN); + + if (!tas2552->mclk) + return -EINVAL; + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0); + + if (tas2552->mclk == TAS2552_245MHZ_CLK || + tas2552->mclk == TAS2552_225MHZ_CLK) { + /* By pass the PLL configuration */ + snd_soc_update_bits(codec, TAS2552_PLL_CTRL_2, + TAS2552_PLL_BYPASS_MASK, + TAS2552_PLL_BYPASS); + } else { + /* Fill in the PLL control registers for J & D + * PLL_CLK = (.5 * freq * J.D) / 2^p + * Need to fill in J and D here based on incoming freq + */ + p = snd_soc_read(codec, TAS2552_PLL_CTRL_1); + p = (p >> 7); + sample_rate = params_rate(params); + + if (sample_rate == 48000) + pll_clk = TAS2552_245MHZ_CLK; + else if (sample_rate == 44100) + pll_clk = TAS2552_225MHZ_CLK; + else { + dev_vdbg(codec->dev, "Substream sample rate is not found %i\n", + params_rate(params)); + return -EINVAL; + } + + j = (pll_clk * 2 * (1 << p)) / tas2552->mclk; + d = (pll_clk * 2 * (1 << p)) % tas2552->mclk; + + snd_soc_update_bits(codec, TAS2552_PLL_CTRL_1, + TAS2552_PLL_J_MASK, j); + snd_soc_write(codec, TAS2552_PLL_CTRL_2, + (d >> 7) & TAS2552_PLL_D_UPPER_MASK); + snd_soc_write(codec, TAS2552_PLL_CTRL_3, + d & TAS2552_PLL_D_LOWER_MASK); + + } + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, + TAS2552_PLL_ENABLE); + + return 0; +} + +static int tas2552_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 serial_format; + u8 serial_control_mask; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + serial_format = 0x00; + break; + case SND_SOC_DAIFMT_CBS_CFM: + serial_format = TAS2552_WORD_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFS: + serial_format = TAS2552_BIT_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFM: + serial_format = (TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK); + break; + default: + dev_vdbg(codec->dev, "DAI Format master is not found\n"); + return -EINVAL; + } + + serial_control_mask = TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + serial_format &= TAS2552_DAIFMT_I2S_MASK; + break; + case SND_SOC_DAIFMT_DSP_A: + serial_format |= TAS2552_DAIFMT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + serial_format |= TAS2552_DAIFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + serial_format |= TAS2552_DAIFMT_LEFT_J; + break; + default: + dev_vdbg(codec->dev, "DAI Format is not found\n"); + return -EINVAL; + } + + if (fmt & SND_SOC_DAIFMT_FORMAT_MASK) + serial_control_mask |= TAS2552_DATA_FORMAT_MASK; + + snd_soc_update_bits(codec, TAS2552_SER_CTRL_1, serial_control_mask, + serial_format); + + return 0; +} + +static int tas2552_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas2552_data *tas2552 = dev_get_drvdata(codec->dev); + + tas2552->mclk = freq; + + return 0; +} + +static int tas2552_mute(struct snd_soc_dai *dai, int mute) +{ + u8 cfg1_reg; + struct snd_soc_codec *codec = dai->codec; + + if (mute) + cfg1_reg = TAS2552_MUTE_MASK; + else + cfg1_reg = ~TAS2552_MUTE_MASK; + + snd_soc_update_bits(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK, cfg1_reg); + + return 0; +} + +#ifdef CONFIG_PM_RUNTIME +static int tas2552_runtime_suspend(struct device *dev) +{ + struct tas2552_data *tas2552 = dev_get_drvdata(dev); + + tas2552_sw_shutdown(tas2552, 0); + + regcache_cache_only(tas2552->regmap, true); + regcache_mark_dirty(tas2552->regmap); + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 0); + + return 0; +} + +static int tas2552_runtime_resume(struct device *dev) +{ + struct tas2552_data *tas2552 = dev_get_drvdata(dev); + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 1); + + tas2552_sw_shutdown(tas2552, 1); + + regcache_cache_only(tas2552->regmap, false); + regcache_sync(tas2552->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops tas2552_pm = { + SET_RUNTIME_PM_OPS(tas2552_runtime_suspend, tas2552_runtime_resume, + NULL) +}; + +static void tas2552_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0); +} + +static struct snd_soc_dai_ops tas2552_speaker_dai_ops = { + .hw_params = tas2552_hw_params, + .set_sysclk = tas2552_set_dai_sysclk, + .set_fmt = tas2552_set_dai_fmt, + .shutdown = tas2552_shutdown, + .digital_mute = tas2552_mute, +}; + +/* Formats supported by TAS2552 driver. */ +#define TAS2552_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* TAS2552 dai structure. */ +static struct snd_soc_dai_driver tas2552_dai[] = { + { + .name = "tas2552-amplifier", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2552_FORMATS, + }, + .ops = &tas2552_speaker_dai_ops, + }, +}; + +/* + * DAC digital volumes. From -7 to 24 dB in 1 dB steps + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 24); + +static const struct snd_kcontrol_new tas2552_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS2552_PGA_GAIN, 0, 0x1f, 1, dac_tlv), +}; + +static const struct reg_default tas2552_init_regs[] = { + { TAS2552_RESERVED_0D, 0xc0 }, +}; + +static int tas2552_codec_probe(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + tas2552->codec = codec; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 1); + + ret = pm_runtime_get_sync(codec->dev); + if (ret < 0) { + dev_err(codec->dev, "Enabling device failed: %d\n", + ret); + goto probe_fail; + } + + snd_soc_write(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK | + TAS2552_PLL_SRC_BCLK); + snd_soc_write(codec, TAS2552_CFG_3, TAS2552_I2S_OUT_SEL | + TAS2552_DIN_SRC_SEL_AVG_L_R | TAS2552_88_96KHZ); + snd_soc_write(codec, TAS2552_DOUT, TAS2552_PDM_DATA_I); + snd_soc_write(codec, TAS2552_OUTPUT_DATA, TAS2552_PDM_DATA_V_I | 0x8); + snd_soc_write(codec, TAS2552_PDM_CFG, TAS2552_PDM_BCLK_SEL); + snd_soc_write(codec, TAS2552_BOOST_PT_CTRL, TAS2552_APT_DELAY_200 | + TAS2552_APT_THRESH_2_1_7); + + ret = regmap_register_patch(tas2552->regmap, tas2552_init_regs, + ARRAY_SIZE(tas2552_init_regs)); + if (ret != 0) { + dev_err(codec->dev, "Failed to write init registers: %d\n", + ret); + goto patch_fail; + } + + snd_soc_write(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN | + TAS2552_BOOST_EN | TAS2552_APT_EN | + TAS2552_LIM_EN); + return 0; + +patch_fail: + pm_runtime_put(codec->dev); +probe_fail: + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 0); + + regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + return -EIO; +} + +static int tas2552_codec_remove(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + + pm_runtime_put(codec->dev); + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 0); + + return 0; +}; + +#ifdef CONFIG_PM +static int tas2552_suspend(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) + dev_err(codec->dev, "Failed to disable supplies: %d\n", + ret); + return 0; +} + +static int tas2552_resume(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", + ret); + } + + return 0; +} +#else +#define tas2552_suspend NULL +#define tas2552_resume NULL +#endif + +static struct snd_soc_codec_driver soc_codec_dev_tas2552 = { + .probe = tas2552_codec_probe, + .remove = tas2552_codec_remove, + .suspend = tas2552_suspend, + .resume = tas2552_resume, + .controls = tas2552_snd_controls, + .num_controls = ARRAY_SIZE(tas2552_snd_controls), +}; + +static const struct regmap_config tas2552_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS2552_MAX_REG, + .reg_defaults = tas2552_reg_defs, + .num_reg_defaults = ARRAY_SIZE(tas2552_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static int tas2552_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev; + struct tas2552_data *data; + int ret; + int i; + + dev = &client->dev; + data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; + + data->enable_gpio = devm_gpiod_get(dev, "enable"); + if (IS_ERR(data->enable_gpio)) { + ret = PTR_ERR(data->enable_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + data->enable_gpio = NULL; + } else { + gpiod_direction_output(data->enable_gpio, 0); + } + + data->tas2552_client = client; + data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tas2552_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) { + dev_err(dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + pm_runtime_set_active(&client->dev); + pm_runtime_set_autosuspend_delay(&client->dev, 1000); + pm_runtime_use_autosuspend(&client->dev); + pm_runtime_enable(&client->dev); + pm_runtime_mark_last_busy(&client->dev); + pm_runtime_put_sync_autosuspend(&client->dev); + + dev_set_drvdata(&client->dev, data); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_tas2552, + tas2552_dai, ARRAY_SIZE(tas2552_dai)); + if (ret < 0) + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + + return ret; +} + +static int tas2552_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id tas2552_id[] = { + { "tas2552", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2552_id); + +#if IS_ENABLED(CONFIG_OF) +static const struct of_device_id tas2552_of_match[] = { + { .compatible = "ti,tas2552", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2552_of_match); +#endif + +static struct i2c_driver tas2552_i2c_driver = { + .driver = { + .name = "tas2552", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas2552_of_match), + .pm = &tas2552_pm, + }, + .probe = tas2552_probe, + .remove = tas2552_i2c_remove, + .id_table = tas2552_id, +}; + +module_i2c_driver(tas2552_i2c_driver); + +MODULE_AUTHOR("Dan Muprhy <dmurphy@ti.com>"); +MODULE_DESCRIPTION("TAS2552 Audio amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2552.h b/sound/soc/codecs/tas2552.h new file mode 100644 index 000000000000..6cea8f31bf88 --- /dev/null +++ b/sound/soc/codecs/tas2552.h @@ -0,0 +1,129 @@ +/* + * tas2552.h - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy <dmurphy@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. + * + * 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. + */ + +#ifndef __TAS2552_H__ +#define __TAS2552_H__ + +/* Register Address Map */ +#define TAS2552_DEVICE_STATUS 0x00 +#define TAS2552_CFG_1 0x01 +#define TAS2552_CFG_2 0x02 +#define TAS2552_CFG_3 0x03 +#define TAS2552_DOUT 0x04 +#define TAS2552_SER_CTRL_1 0x05 +#define TAS2552_SER_CTRL_2 0x06 +#define TAS2552_OUTPUT_DATA 0x07 +#define TAS2552_PLL_CTRL_1 0x08 +#define TAS2552_PLL_CTRL_2 0x09 +#define TAS2552_PLL_CTRL_3 0x0a +#define TAS2552_BTIP 0x0b +#define TAS2552_BTS_CTRL 0x0c +#define TAS2552_RESERVED_0D 0x0d +#define TAS2552_LIMIT_RATE_HYS 0x0e +#define TAS2552_LIMIT_RELEASE 0x0f +#define TAS2552_LIMIT_INT_COUNT 0x10 +#define TAS2552_PDM_CFG 0x11 +#define TAS2552_PGA_GAIN 0x12 +#define TAS2552_EDGE_RATE_CTRL 0x13 +#define TAS2552_BOOST_PT_CTRL 0x14 +#define TAS2552_VER_NUM 0x16 +#define TAS2552_VBAT_DATA 0x19 +#define TAS2552_MAX_REG 0x20 + +/* CFG1 Register Masks */ +#define TAS2552_MUTE_MASK (1 << 2) +#define TAS2552_SWS_MASK (1 << 1) +#define TAS2552_WCLK_MASK 0x07 +#define TAS2552_CLASSD_EN_MASK (1 << 7) + +/* CFG2 Register Masks */ +#define TAS2552_CLASSD_EN (1 << 7) +#define TAS2552_BOOST_EN (1 << 6) +#define TAS2552_APT_EN (1 << 5) +#define TAS2552_PLL_ENABLE (1 << 3) +#define TAS2552_LIM_EN (1 << 2) +#define TAS2552_IVSENSE_EN (1 << 1) + +/* CFG3 Register Masks */ +#define TAS2552_WORD_CLK_MASK (1 << 7) +#define TAS2552_BIT_CLK_MASK (1 << 6) +#define TAS2552_DATA_FORMAT_MASK (0x11 << 2) + +#define TAS2552_DAIFMT_I2S_MASK 0xf3 +#define TAS2552_DAIFMT_DSP (1 << 3) +#define TAS2552_DAIFMT_RIGHT_J (1 << 4) +#define TAS2552_DAIFMT_LEFT_J (0x11 << 3) + +#define TAS2552_PLL_SRC_MCLK 0x00 +#define TAS2552_PLL_SRC_BCLK (1 << 3) +#define TAS2552_PLL_SRC_IVCLKIN (1 << 4) +#define TAS2552_PLL_SRC_1_8_FIXED (0x11 << 3) + +#define TAS2552_DIN_SRC_SEL_MUTED 0x00 +#define TAS2552_DIN_SRC_SEL_LEFT (1 << 4) +#define TAS2552_DIN_SRC_SEL_RIGHT (1 << 5) +#define TAS2552_DIN_SRC_SEL_AVG_L_R (0x11 << 4) + +#define TAS2552_PDM_IN_SEL (1 << 5) +#define TAS2552_I2S_OUT_SEL (1 << 6) +#define TAS2552_ANALOG_IN_SEL (1 << 7) + +/* CFG3 WCLK Dividers */ +#define TAS2552_8KHZ 0x00 +#define TAS2552_11_12KHZ (1 << 1) +#define TAS2552_16KHZ (1 << 2) +#define TAS2552_22_24KHZ (1 << 3) +#define TAS2552_32KHZ (1 << 4) +#define TAS2552_44_48KHZ (1 << 5) +#define TAS2552_88_96KHZ (1 << 6) +#define TAS2552_176_192KHZ (1 << 7) + +/* OUTPUT_DATA register */ +#define TAS2552_PDM_DATA_I 0x00 +#define TAS2552_PDM_DATA_V (1 << 6) +#define TAS2552_PDM_DATA_I_V (1 << 7) +#define TAS2552_PDM_DATA_V_I (0x11 << 6) + +/* PDM CFG Register */ +#define TAS2552_PDM_DATA_ES_RISE 0x4 + +#define TAS2552_PDM_PLL_CLK_SEL 0x00 +#define TAS2552_PDM_IV_CLK_SEL (1 << 1) +#define TAS2552_PDM_BCLK_SEL (1 << 2) +#define TAS2552_PDM_MCLK_SEL (1 << 3) + +/* Boost pass-through register */ +#define TAS2552_APT_DELAY_50 0x00 +#define TAS2552_APT_DELAY_75 (1 << 1) +#define TAS2552_APT_DELAY_125 (1 << 2) +#define TAS2552_APT_DELAY_200 (1 << 3) + +#define TAS2552_APT_THRESH_2_5 0x00 +#define TAS2552_APT_THRESH_1_7 (1 << 3) +#define TAS2552_APT_THRESH_1_4_1_1 (1 << 4) +#define TAS2552_APT_THRESH_2_1_7 (0x11 << 2) + +/* PLL Control Register */ +#define TAS2552_245MHZ_CLK 24576000 +#define TAS2552_225MHZ_CLK 22579200 +#define TAS2552_PLL_J_MASK 0x7f +#define TAS2552_PLL_D_UPPER_MASK 0x3f +#define TAS2552_PLL_D_LOWER_MASK 0xff +#define TAS2552_PLL_BYPASS_MASK 0x80 +#define TAS2552_PLL_BYPASS 0x80 + +#endif diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index d48491a4a19d..249ef5c4c762 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -36,6 +36,7 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/regmap.h> +#include <linux/regulator/consumer.h> #include <linux/spi/spi.h> #include <linux/of.h> #include <linux/of_device.h> @@ -240,6 +241,10 @@ static int tas5086_reg_read(void *context, unsigned int reg, return 0; } +static const char * const supply_names[] = { + "dvdd", "avdd" +}; + struct tas5086_private { struct regmap *regmap; unsigned int mclk, sclk; @@ -251,6 +256,7 @@ struct tas5086_private { int rate; /* GPIO driving Reset pin, if any */ int gpio_nreset; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static int tas5086_deemph[] = { 0, 32000, 44100, 48000 }; @@ -419,14 +425,14 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream, } /* ... then add the offset for the sample bit depth. */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: val += 0; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: val += 1; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: val += 2; break; default: @@ -773,6 +779,8 @@ static int tas5086_soc_suspend(struct snd_soc_codec *codec) if (ret < 0) return ret; + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; } @@ -781,6 +789,10 @@ static int tas5086_soc_resume(struct snd_soc_codec *codec) struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); int ret; + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); + if (ret < 0) + return ret; + tas5086_reset(priv); regcache_mark_dirty(priv->regmap); @@ -812,6 +824,12 @@ static int tas5086_probe(struct snd_soc_codec *codec) struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); int i, ret; + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + priv->pwm_start_mid_z = 0; priv->charge_period = 1300000; /* hardware default is 1300 ms */ @@ -832,16 +850,22 @@ static int tas5086_probe(struct snd_soc_codec *codec) } } + tas5086_reset(priv); ret = tas5086_init(codec->dev, priv); if (ret < 0) - return ret; + goto exit_disable_regulators; /* set master volume to 0 dB */ ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); if (ret < 0) - return ret; + goto exit_disable_regulators; return 0; + +exit_disable_regulators: + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + + return ret; } static int tas5086_remove(struct snd_soc_codec *codec) @@ -852,6 +876,8 @@ static int tas5086_remove(struct snd_soc_codec *codec) /* Set codec to the reset state */ gpio_set_value(priv->gpio_nreset, 0); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; }; @@ -900,6 +926,16 @@ static int tas5086_i2c_probe(struct i2c_client *i2c, if (!priv) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) { + dev_err(dev, "Failed to get regulators: %d\n", ret); + return ret; + } + priv->regmap = devm_regmap_init(dev, NULL, i2c, &tas5086_regmap); if (IS_ERR(priv->regmap)) { ret = PTR_ERR(priv->regmap); @@ -919,21 +955,34 @@ static int tas5086_i2c_probe(struct i2c_client *i2c, gpio_nreset = -EINVAL; priv->gpio_nreset = gpio_nreset; + + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); + if (ret < 0) { + dev_err(dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + tas5086_reset(priv); /* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */ ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i); - if (ret < 0) - return ret; - - if (i != 0x3) { + if (ret == 0 && i != 0x3) { dev_err(dev, "Failed to identify TAS5086 codec (got %02x)\n", i); - return -ENODEV; + ret = -ENODEV; } - return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086, - &tas5086_dai, 1); + /* + * The chip has been identified, so we can turn off the power + * again until the dai link is set up. + */ + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + + if (ret == 0) + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086, + &tas5086_dai, 1); + + return ret; } static int tas5086_i2c_remove(struct i2c_client *i2c) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 686b8b85b956..d67167920c2f 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -364,16 +364,16 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface_reg |= (0x01 << 2); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface_reg |= (0x02 << 2); break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface_reg |= (0x03 << 2); break; } diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 43069de3d56a..620ab9ea1ef0 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -71,8 +71,8 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, dev_dbg(&aic26->spi->dev, "aic26_hw_params(substream=%p, params=%p)\n", substream, params); - dev_dbg(&aic26->spi->dev, "rate=%i format=%i\n", params_rate(params), - params_format(params)); + dev_dbg(&aic26->spi->dev, "rate=%i width=%d\n", params_rate(params), + params_width(params)); switch (params_rate(params)) { case 8000: fsref = 48000; divisor = AIC26_DIV_6; break; @@ -89,11 +89,11 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, } /* select data word length */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: wlen = AIC26_WLEN_16; break; - case SNDRV_PCM_FORMAT_S16_BE: wlen = AIC26_WLEN_16; break; - case SNDRV_PCM_FORMAT_S24_BE: wlen = AIC26_WLEN_24; break; - case SNDRV_PCM_FORMAT_S32_BE: wlen = AIC26_WLEN_32; break; + switch (params_width(params)) { + case 8: wlen = AIC26_WLEN_16; break; + case 16: wlen = AIC26_WLEN_16; break; + case 24: wlen = AIC26_WLEN_24; break; + case 32: wlen = AIC26_WLEN_32; break; default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL; } diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 23419109ecac..0f64c7890eed 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -249,17 +249,16 @@ static const char * const mic_select_text[] = { "Off", "FFR 10 Ohm", "FFR 20 Ohm", "FFR 40 Ohm" }; -static const -SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, mic_select_text); -static const -SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, mic_select_text); -static const -SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, mic_select_text); - -static const -SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text); -static const -SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, mic_select_text); +static SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, + mic_select_text); +static SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, + mic_select_text); +static SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, + mic_select_text); + +static SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text); +static SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, + mic_select_text); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6350, 50, 0); static const DECLARE_TLV_DB_SCALE(adc_fgain_tlv, 0, 10, 0); @@ -329,6 +328,7 @@ static int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, unsigned int bits; int counter = count; int ret = regmap_read(aic31xx->regmap, reg, &bits); + while ((bits & mask) != wbits && counter && !ret) { usleep_range(sleep, sleep * 2); ret = regmap_read(aic31xx->regmap, reg, &bits); @@ -435,6 +435,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + switch (event) { case SND_SOC_DAPM_POST_PMU: /* change mic bias voltage to user defined */ @@ -759,8 +760,8 @@ static int aic31xx_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; u8 data = 0; - dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n", - __func__, params_format(params), params_width(params), + dev_dbg(codec->dev, "## %s: width %d rate %d\n", + __func__, params_width(params), params_rate(params)); switch (params_width(params)) { @@ -779,8 +780,8 @@ static int aic31xx_hw_params(struct snd_pcm_substream *substream, AIC31XX_IFACE1_DATALEN_SHIFT); break; default: - dev_err(codec->dev, "%s: Unsupported format %d\n", - __func__, params_format(params)); + dev_err(codec->dev, "%s: Unsupported width %d\n", + __func__, params_width(params)); return -EINVAL; } @@ -1178,7 +1179,7 @@ static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) } #endif /* CONFIG_OF */ -static void aic31xx_device_init(struct aic31xx_priv *aic31xx) +static int aic31xx_device_init(struct aic31xx_priv *aic31xx) { int ret, i; @@ -1197,7 +1198,7 @@ static void aic31xx_device_init(struct aic31xx_priv *aic31xx) "aic31xx-reset-pin"); if (ret < 0) { dev_err(aic31xx->dev, "not able to acquire gpio\n"); - return; + return ret; } } @@ -1210,6 +1211,7 @@ static void aic31xx_device_init(struct aic31xx_priv *aic31xx) if (ret != 0) dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret); + return ret; } static int aic31xx_i2c_probe(struct i2c_client *i2c, @@ -1239,7 +1241,9 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, aic31xx->pdata.codec_type = id->driver_data; - aic31xx_device_init(aic31xx); + ret = aic31xx_device_init(aic31xx); + if (ret) + return ret; return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, aic31xx_dai_driver, diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 1d9b117345a3..6ea662db2410 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -450,16 +450,16 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream, data = snd_soc_read(codec, AIC32X4_IFACE1); data = data & ~(3 << 4); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: data |= (AIC32X4_WORD_LEN_20BITS << AIC32X4_DOSRMSB_SHIFT); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: data |= (AIC32X4_WORD_LEN_24BITS << AIC32X4_DOSRMSB_SHIFT); break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: data |= (AIC32X4_WORD_LEN_32BITS << AIC32X4_DOSRMSB_SHIFT); break; } @@ -626,32 +626,33 @@ static int aic32x4_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN | AIC32X4_MICBIAS_2075V); } - if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) { + if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); - } tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ? AIC32X4_LDOCTLEN : 0; snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg); tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE); - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) { + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) tmp_reg |= AIC32X4_LDOIN_18_36; - } - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) { + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) tmp_reg |= AIC32X4_LDOIN2HP; - } snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg); /* Mic PGA routing */ if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) - snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K); + snd_soc_write(codec, AIC32X4_LMICPGANIN, + AIC32X4_LMICPGANIN_IN2R_10K); else - snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_CM1L_10K); + snd_soc_write(codec, AIC32X4_LMICPGANIN, + AIC32X4_LMICPGANIN_CM1L_10K); if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) - snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K); + snd_soc_write(codec, AIC32X4_RMICPGANIN, + AIC32X4_RMICPGANIN_IN1L_10K); else - snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_CM1R_10K); + snd_soc_write(codec, AIC32X4_RMICPGANIN, + AIC32X4_RMICPGANIN_CM1R_10K); aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e12fafbb1e09..64f179ee9834 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -873,16 +873,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, /* select data word length */ data = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4)); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: data |= (0x01 << 4); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: data |= (0x02 << 4); break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: data |= (0x03 << 4); break; } @@ -1194,7 +1194,8 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, #define AIC3X_RATES SNDRV_PCM_RATE_8000_96000 #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ - SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops aic3x_dai_ops = { .hw_params = aic3x_hw_params, @@ -1477,10 +1478,8 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, u32 value; aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL); - if (aic3x == NULL) { - dev_err(&i2c->dev, "failed to create private data\n"); + if (!aic3x) return -ENOMEM; - } aic3x->regmap = devm_regmap_init_i2c(i2c, &aic3x_regmap); if (IS_ERR(aic3x->regmap)) { @@ -1498,10 +1497,8 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, } else if (np) { ai3x_setup = devm_kzalloc(&i2c->dev, sizeof(*ai3x_setup), GFP_KERNEL); - if (ai3x_setup == NULL) { - dev_err(&i2c->dev, "failed to create private data\n"); + if (!ai3x_setup) return -ENOMEM; - } ret = of_get_named_gpio(np, "gpio-reset", 0); if (ret >= 0) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index df3a7506c023..e21ed934bdbf 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -832,18 +832,18 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: dac33->fifo_size = DAC33_FIFO_SIZE_16BIT; dac33->burst_rate = CALC_BURST_RATE(dac33->burst_bclkdiv, 32); break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: dac33->fifo_size = DAC33_FIFO_SIZE_24BIT; dac33->burst_rate = CALC_BURST_RATE(dac33->burst_bclkdiv, 64); break; default: - dev_err(codec->dev, "unsupported format %d\n", - params_format(params)); + dev_err(codec->dev, "unsupported width %d\n", + params_width(params)); return -EINVAL; } @@ -1404,7 +1404,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) if (dac33->irq >= 0) { ret = request_irq(dac33->irq, dac33_interrupt_handler, IRQF_TRIGGER_RISING, - codec->name, codec); + codec->component.name, codec); if (ret < 0) { dev_err(codec->dev, "Could not request IRQ%d (%d)\n", dac33->irq, ret); diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 8fc5a647453b..6fac9e034c48 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -381,10 +381,8 @@ static int tpa6130a2_probe(struct i2c_client *client, dev = &client->dev; data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); - if (data == NULL) { - dev_err(dev, "Can not allocate memory\n"); + if (!data) return -ENOMEM; - } if (pdata) { data->power_gpio = pdata->power_gpio; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 69e12a311ba2..b6b0cb399599 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -344,17 +344,16 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) { struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - int status = -1; if (enable) { twl4030->apll_enabled++; if (twl4030->apll_enabled == 1) - status = twl4030_audio_enable_resource( + twl4030_audio_enable_resource( TWL4030_AUDIO_RES_APLL); } else { twl4030->apll_enabled--; if (!twl4030->apll_enabled) - status = twl4030_audio_disable_resource( + twl4030_audio_disable_resource( TWL4030_AUDIO_RES_APLL); } } @@ -1764,16 +1763,16 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, old_format = twl4030_read(codec, TWL4030_REG_AUDIO_IF); format = old_format; format &= ~TWL4030_DATA_WIDTH; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: format |= TWL4030_DATA_WIDTH_16S_16W; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: format |= TWL4030_DATA_WIDTH_32S_24W; break; default: - dev_err(codec->dev, "%s: unknown format %d\n", __func__, - params_format(params)); + dev_err(codec->dev, "%s: unsupported bits/sample %d\n", + __func__, params_width(params)); return -EINVAL; } @@ -2162,10 +2161,8 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) twl4030 = devm_kzalloc(codec->dev, sizeof(struct twl4030_priv), GFP_KERNEL); - if (twl4030 == NULL) { - dev_err(codec->dev, "Can not allocate memory\n"); + if (!twl4030) return -ENOMEM; - } snd_soc_codec_set_drvdata(codec, twl4030); /* Set the defaults, and power up the codec */ twl4030->sysclk = twl4030_audio_get_mclk() / 1000; diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index edf27acc1d77..32b2f78aa62c 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -243,14 +243,14 @@ static int uda134x_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_I2S: break; case SND_SOC_DAIFMT_RIGHT_J: - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: hw_params |= (1<<1); break; - case SNDRV_PCM_FORMAT_S18_3LE: + case 18: hw_params |= (1<<2); break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: hw_params |= ((1<<2) | (1<<1)); break; default: @@ -479,7 +479,7 @@ static struct snd_soc_dai_driver uda134x_dai = { static int uda134x_soc_probe(struct snd_soc_codec *codec) { struct uda134x_priv *uda134x; - struct uda134x_platform_data *pd = codec->card->dev->platform_data; + struct uda134x_platform_data *pd = codec->component.card->dev->platform_data; const struct snd_soc_dapm_widget *widgets; unsigned num_widgets; diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 4ead0dc02b87..f3d4e88d0b7b 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -341,8 +341,9 @@ static int wl1273_hw_params(struct snd_pcm_substream *substream, struct wl1273_core *core = wl1273->core; unsigned int rate, width, r; - if (params_format(params) != SNDRV_PCM_FORMAT_S16_LE) { - pr_err("Only SNDRV_PCM_FORMAT_S16_LE supported.\n"); + if (params_width(params) != 16) { + dev_err(dai->dev, "%d bits/sample not supported\n", + params_width(params)); return -EINVAL; } @@ -461,10 +462,8 @@ static int wl1273_probe(struct snd_soc_codec *codec) } wl1273 = kzalloc(sizeof(struct wl1273_priv), GFP_KERNEL); - if (wl1273 == NULL) { - dev_err(codec->dev, "Cannot allocate memory.\n"); + if (!wl1273) return -ENOMEM; - } wl1273->mode = WL1273_MODE_BT; wl1273->core = *core; diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 71ce3159a62e..f37989ec7cba 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -144,7 +144,7 @@ static const struct snd_soc_dapm_route wm0010_dapm_routes[] = { static const char *wm0010_state_to_str(enum wm0010_state state) { - const char *state_to_str[] = { + static const char * const state_to_str[] = { "Power off", "Out of reset", "Boot ROM", @@ -413,7 +413,6 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); if (!xfer) { - dev_err(codec->dev, "Failed to allocate xfer\n"); ret = -ENOMEM; goto abort; } @@ -423,8 +422,6 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) out = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!out) { - dev_err(codec->dev, - "Failed to allocate RX buffer\n"); ret = -ENOMEM; goto abort1; } @@ -432,8 +429,6 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) img = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!img) { - dev_err(codec->dev, - "Failed to allocate image buffer\n"); ret = -ENOMEM; goto abort1; } @@ -526,14 +521,12 @@ static int wm0010_stage2_load(struct snd_soc_codec *codec) /* Copy to local buffer first as vmalloc causes problems for dma */ img = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); if (!img) { - dev_err(codec->dev, "Failed to allocate image buffer\n"); ret = -ENOMEM; goto abort2; } out = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); if (!out) { - dev_err(codec->dev, "Failed to allocate output buffer\n"); ret = -ENOMEM; goto abort1; } @@ -679,11 +672,8 @@ static int wm0010_boot(struct snd_soc_codec *codec) } img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA); - if (!img_swap) { - dev_err(codec->dev, - "Failed to allocate image buffer\n"); + if (!img_swap) goto abort; - } /* We need to re-order for 0010 */ byte_swap_64((u64 *)&pll_rec, img_swap, len); diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index 6e6b93d4696e..8011f75fb6cb 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -164,7 +164,6 @@ static int wm1250_ev1_pdata(struct i2c_client *i2c) wm1250 = devm_kzalloc(&i2c->dev, sizeof(*wm1250), GFP_KERNEL); if (!wm1250) { - dev_err(&i2c->dev, "Unable to allocate private data\n"); ret = -ENOMEM; goto err; } diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index a4c352cc3464..34ef65c52a7d 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -826,10 +826,8 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, wm2000 = devm_kzalloc(&i2c->dev, sizeof(struct wm2000_priv), GFP_KERNEL); - if (wm2000 == NULL) { - dev_err(&i2c->dev, "Unable to allocate private data\n"); + if (!wm2000) return -ENOMEM; - } mutex_init(&wm2000->lock); diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 91a9ea2a2056..7bb0d36d4c54 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -735,8 +735,7 @@ WM5100_MIXER_CONTROLS("LHPF4", WM5100_HPLP4MIX_INPUT_1_SOURCE), static void wm5100_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); u16 val, expect, i; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 289b64d89abd..f60234962527 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -612,6 +612,62 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, return 0; } +static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + uint16_t data; + + mutex_lock(&codec->mutex); + data = cpu_to_be16(arizona->dac_comp_coeff); + memcpy(ucontrol->value.bytes.data, &data, sizeof(data)); + mutex_unlock(&codec->mutex); + + return 0; +} + +static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + + mutex_lock(&codec->mutex); + memcpy(&arizona->dac_comp_coeff, ucontrol->value.bytes.data, + sizeof(arizona->dac_comp_coeff)); + arizona->dac_comp_coeff = be16_to_cpu(arizona->dac_comp_coeff); + mutex_unlock(&codec->mutex); + + return 0; +} + +static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + + mutex_lock(&codec->mutex); + ucontrol->value.integer.value[0] = arizona->dac_comp_enabled; + mutex_unlock(&codec->mutex); + + return 0; +} + +static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + + mutex_lock(&codec->mutex); + arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + mutex_unlock(&codec->mutex); + + return 0; +} + static const char *wm5102_osr_text[] = { "Low power", "Normal", "High performance", }; @@ -843,6 +899,12 @@ SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv), SOC_ENUM("Noise Gate Hold", arizona_ng_hold), +SND_SOC_BYTES_EXT("Output Compensation Coefficient", 2, + wm5102_out_comp_coeff_get, wm5102_out_comp_coeff_put), + +SOC_SINGLE_EXT("Output Compensation Switch", 0, 0, 1, 0, + wm5102_out_comp_switch_get, wm5102_out_comp_switch_put), + WM5102_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L), WM5102_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R), WM5102_NG_SRC("HPOUT2L", ARIZONA_NOISE_GATE_SELECT_2L), @@ -1653,6 +1715,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5102-aif2", @@ -1674,6 +1737,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5102-aif3", @@ -1695,6 +1759,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5102-slim1", diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2e5fcb559e90..2f2ec26d831c 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1485,6 +1485,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5110-aif2", @@ -1506,6 +1507,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5110-aif3", @@ -1527,6 +1529,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5110-slim1", @@ -1596,6 +1599,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_spk(codec); arizona_init_gpio(codec); + arizona_init_mono(codec); ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8); if (ret != 0) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 392285edb595..3dfdcc4197fa 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -918,16 +918,16 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, ~WM8350_AIF_WL_MASK; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x1 << 10; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x2 << 10; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x3 << 10; break; } @@ -1341,21 +1341,18 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, { struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); struct wm8350 *wm8350 = priv->wm8350; - int irq; int ena; switch (which) { case WM8350_JDL: priv->hpl.jack = jack; priv->hpl.report = report; - irq = WM8350_IRQ_CODEC_JCK_DET_L; ena = WM8350_JDL_ENA; break; case WM8350_JDR: priv->hpr.jack = jack; priv->hpr.report = report; - irq = WM8350_IRQ_CODEC_JCK_DET_R; ena = WM8350_JDR_ENA; break; diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 06e913d3fea1..72471bef2e9a 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1095,16 +1095,16 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream, audio1 &= ~WM8400_AIF_WL_MASK; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: audio1 |= WM8400_AIF_WL_20BITS; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: audio1 |= WM8400_AIF_WL_24BITS; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: audio1 |= WM8400_AIF_WL_32BITS; break; } diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 1c1e328feeb8..e11127f9069e 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -449,16 +449,16 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, u16 adn = snd_soc_read(codec, WM8510_ADD) & 0x1f1; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0020; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0040; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x0060; break; } diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 601ee8178af1..ec1f5740dbd0 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -163,16 +163,16 @@ static int wm8523_hw_params(struct snd_pcm_substream *substream, aifctrl2 |= lrclk_ratios[i].value; aifctrl1 &= ~WM8523_WL_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: aifctrl1 |= 0x8; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: aifctrl1 |= 0x10; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: aifctrl1 |= 0x18; break; } diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 7665ff6aea6d..911605ee25b0 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -511,19 +511,19 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, int i, ratio, osr; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: paifa |= 0x8; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: paifa |= 0x0; paifb |= WM8580_AIF_LENGTH_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: paifa |= 0x0; paifb |= WM8580_AIF_LENGTH_24; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: paifa |= 0x0; paifb |= WM8580_AIF_LENGTH_32; break; diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index b0fbcb377baf..32187e739b4f 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -169,13 +169,13 @@ static int wm8711_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, WM8711_SRATE, srate); /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0008; break; } diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index bac7fc28fe71..38ff826f589a 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -94,13 +94,13 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream, dac &= ~0x18; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: dac |= 0x10; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: dac |= 0x08; break; default: diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 5ada61611324..eebb3280bfad 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -348,13 +348,13 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, WM8731_SRATE, srate); /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0008; break; } diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index b27f26cdc049..744a422ecb05 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -367,16 +367,16 @@ static int wm8737_hw_params(struct snd_pcm_substream *substream, clocking |= coeff_div[i].usb | (coeff_div[i].sr << WM8737_SR_SHIFT); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: af |= 0x8; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: af |= 0x10; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: af |= 0x18; break; default: diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index b33542a04607..a237f1627f61 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -241,26 +241,26 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, } /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0001; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0002; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x0003; break; default: dev_dbg(codec->dev, "wm8741_hw_params: Unsupported bit size param = %d", - params_format(params)); + params_width(params)); return -EINVAL; } dev_dbg(codec->dev, "wm8741_hw_params: bit size param = %d", - params_format(params)); + params_width(params)); snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface); return 0; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 33990b63d214..67653a2db223 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -586,16 +586,16 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, int coeff = get_coeff(wm8750->sysclk, params_rate(params)); /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0008; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x000c; break; } diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 53e57b4049a8..e54e097f4fcb 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -937,16 +937,16 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x017f; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: voice |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: voice |= 0x0008; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: voice |= 0x000c; break; } @@ -1176,16 +1176,16 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, coeff_div[coeff].usb); /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: hifi |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: hifi |= 0x0008; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: hifi |= 0x000c; break; } diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index c61aeb38efb8..180e7a098726 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -426,16 +426,16 @@ static int wm8770_hw_params(struct snd_pcm_substream *substream, wm8770 = snd_soc_codec_get_drvdata(codec); iface = 0; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x10; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x20; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x30; break; } diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index d96e5963ee35..0ea01dfcb6e1 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -270,19 +270,19 @@ static int wm8804_hw_params(struct snd_pcm_substream *substream, codec = dai->codec; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: blen = 0x0; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: blen = 0x1; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: blen = 0x2; break; default: dev_err(dai->dev, "Unsupported word length: %u\n", - params_format(params)); + params_width(params)); return -EINVAL; } diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index d09fdce57f5a..44a5f1511f0f 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -640,16 +640,16 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream, reg = snd_soc_read(codec, WM8900_REG_AUDIO1) & ~0x60; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: reg |= 0x20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: reg |= 0x40; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: reg |= 0x60; break; default: diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b84940c359a1..aa0984864e76 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -281,8 +281,7 @@ static int wm8903_dcs_event(struct snd_soc_dapm_widget *w, static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int dcs_mode = WM8903_DCS_MODE_WRITE_STOP; int i, val; @@ -1477,19 +1476,19 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, aif1 &= ~WM8903_AIF_WL_MASK; bclk = 2 * fs; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: bclk *= 16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: bclk *= 20; aif1 |= 0x4; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: bclk *= 24; aif1 |= 0x8; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: bclk *= 32; aif1 |= 0xc; break; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index f7c549949c54..4d2d2b1380d5 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -11,6 +11,7 @@ * published by the Free Software Foundation. */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -49,6 +50,7 @@ static const char *wm8904_supply_names[WM8904_NUM_SUPPLIES] = { /* codec private data */ struct wm8904_priv { struct regmap *regmap; + struct clk *mclk; enum wm8904_type devtype; @@ -1290,16 +1292,16 @@ static int wm8904_hw_params(struct snd_pcm_substream *substream, wm8904->bclk = snd_soc_params_to_bclk(params); } - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: aif1 |= 0x40; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: aif1 |= 0x80; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: aif1 |= 0xc0; break; default: @@ -1828,6 +1830,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + clk_prepare_enable(wm8904->mclk); break; case SND_SOC_BIAS_PREPARE: @@ -1894,6 +1897,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); + clk_disable_unprepare(wm8904->mclk); break; } codec->dapm.bias_level = level; @@ -2013,12 +2017,8 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) /* We need an array of texts for the enum API */ wm8904->drc_texts = kmalloc(sizeof(char *) * pdata->num_drc_cfgs, GFP_KERNEL); - if (!wm8904->drc_texts) { - dev_err(codec->dev, - "Failed to allocate %d DRC config texts\n", - pdata->num_drc_cfgs); + if (!wm8904->drc_texts) return; - } for (i = 0; i < pdata->num_drc_cfgs; i++) wm8904->drc_texts[i] = pdata->drc_cfgs[i].name; @@ -2110,6 +2110,13 @@ static int wm8904_i2c_probe(struct i2c_client *i2c, if (wm8904 == NULL) return -ENOMEM; + wm8904->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8904->mclk)) { + ret = PTR_ERR(wm8904->mclk); + dev_err(&i2c->dev, "Failed to get MCLK\n"); + return ret; + } + wm8904->regmap = devm_regmap_init_i2c(i2c, &wm8904_regmap); if (IS_ERR(wm8904->regmap)) { ret = PTR_ERR(wm8904->regmap); diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index fc6eec9ad66b..52011043e54c 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -430,19 +430,19 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, if (ret) goto error_ret; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: + switch (params_width(params)) { + case 8: companding = companding | (1 << 5); break; - case SNDRV_PCM_FORMAT_S16_LE: + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= (1 << 5); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= (2 << 5); break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= (3 << 5); break; } diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 2a35108f233d..09d91d9dc4ee 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -597,17 +597,17 @@ static int wm8955_hw_params(struct snd_pcm_substream *substream, int ret; int wl; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: wl = 0; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: wl = 0x4; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: wl = 0x8; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: wl = 0xc; break; default: diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index b2ebb104d879..0dada7f0105e 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -934,12 +934,8 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) /* We need an array of texts for the enum API */ wm8994->mbc_texts = kmalloc(sizeof(char *) * pdata->num_mbc_cfgs, GFP_KERNEL); - if (!wm8994->mbc_texts) { - dev_err(wm8994->hubs.codec->dev, - "Failed to allocate %d MBC config texts\n", - pdata->num_mbc_cfgs); + if (!wm8994->mbc_texts) return; - } for (i = 0; i < pdata->num_mbc_cfgs; i++) wm8994->mbc_texts[i] = pdata->mbc_cfgs[i].name; @@ -963,12 +959,8 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) /* We need an array of texts for the enum API */ wm8994->vss_texts = kmalloc(sizeof(char *) * pdata->num_vss_cfgs, GFP_KERNEL); - if (!wm8994->vss_texts) { - dev_err(wm8994->hubs.codec->dev, - "Failed to allocate %d VSS config texts\n", - pdata->num_vss_cfgs); + if (!wm8994->vss_texts) return; - } for (i = 0; i < pdata->num_vss_cfgs; i++) wm8994->vss_texts[i] = pdata->vss_cfgs[i].name; @@ -993,12 +985,8 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) /* We need an array of texts for the enum API */ wm8994->vss_hpf_texts = kmalloc(sizeof(char *) * pdata->num_vss_hpf_cfgs, GFP_KERNEL); - if (!wm8994->vss_hpf_texts) { - dev_err(wm8994->hubs.codec->dev, - "Failed to allocate %d VSS HPF config texts\n", - pdata->num_vss_hpf_cfgs); + if (!wm8994->vss_hpf_texts) return; - } for (i = 0; i < pdata->num_vss_hpf_cfgs; i++) wm8994->vss_hpf_texts[i] = pdata->vss_hpf_cfgs[i].name; @@ -1024,12 +1012,8 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) /* We need an array of texts for the enum API */ wm8994->enh_eq_texts = kmalloc(sizeof(char *) * pdata->num_enh_eq_cfgs, GFP_KERNEL); - if (!wm8994->enh_eq_texts) { - dev_err(wm8994->hubs.codec->dev, - "Failed to allocate %d enhanced EQ config texts\n", - pdata->num_enh_eq_cfgs); + if (!wm8994->enh_eq_texts) return; - } for (i = 0; i < pdata->num_enh_eq_cfgs; i++) wm8994->enh_eq_texts[i] = pdata->enh_eq_cfgs[i].name; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index a145d0431b63..4dc4e85116cd 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -472,7 +472,7 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec) * list each time to find the desired power state do so now * and save the result. */ - list_for_each_entry(w, &codec->card->widgets, list) { + list_for_each_entry(w, &codec->component.card->widgets, list) { if (w->dapm != &codec->dapm) continue; if (strcmp(w->name, "LOUT1 PGA") == 0) @@ -567,24 +567,21 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; - snd_pcm_format_t format = params_format(params); int i; /* bit size */ - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: - case SNDRV_PCM_FORMAT_S16_BE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: - case SNDRV_PCM_FORMAT_S20_3BE: + case 20: iface |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S24_BE: + case 24: iface |= 0x0008; break; default: - dev_err(codec->dev, "unsupported format %i\n", format); + dev_err(codec->dev, "unsupported width %d\n", + params_width(params)); return -EINVAL; } diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 9c88f04442b3..41d23e920ad5 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -565,16 +565,16 @@ static int wm8961_hw_params(struct snd_pcm_substream *substream, reg = snd_soc_read(codec, WM8961_AUDIO_INTERFACE_0); reg &= ~WM8961_WL_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: reg |= 1 << WM8961_WL_SHIFT; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: reg |= 2 << WM8961_WL_SHIFT; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: reg |= 3 << WM8961_WL_SHIFT; break; default: diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index ca2fda9d72be..1098ae32f1f9 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -14,6 +14,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/gcd.h> @@ -2586,16 +2587,16 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, if (wm8962->lrclk % 8000 == 0) adctl3 |= WM8962_SAMPLE_RATE_INT_MODE; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: aif0 |= 0x4; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: aif0 |= 0x8; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: aif0 |= 0xc; break; default: @@ -3541,6 +3542,8 @@ static int wm8962_set_pdata_from_of(struct i2c_client *i2c, pdata->gpio_init[i] = 0x0; } + pdata->mclk = devm_clk_get(&i2c->dev, NULL); + return 0; } @@ -3572,6 +3575,14 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, return ret; } + /* Mark the mclk pointer to NULL if no mclk assigned */ + if (IS_ERR(wm8962->pdata.mclk)) { + /* But do not ignore the request for probe defer */ + if (PTR_ERR(wm8962->pdata.mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + wm8962->pdata.mclk = NULL; + } + for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) wm8962->supplies[i].supply = wm8962_supply_names[i]; @@ -3780,6 +3791,12 @@ static int wm8962_runtime_resume(struct device *dev) struct wm8962_priv *wm8962 = dev_get_drvdata(dev); int ret; + ret = clk_prepare_enable(wm8962->pdata.mclk); + if (ret) { + dev_err(dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); if (ret != 0) { @@ -3839,6 +3856,8 @@ static int wm8962_runtime_suspend(struct device *dev) regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); + clk_disable_unprepare(wm8962->pdata.mclk); + return 0; } #endif diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 09b7b4200221..0499cd4cfb71 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -517,16 +517,16 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, int coeff = get_coeff(wm8971->sysclk, params_rate(params)); /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0008; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x000c; break; } diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 0627c56fa44e..682e9eda1019 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -445,16 +445,16 @@ static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream, u16 adn = snd_soc_read(codec, WM8974_ADD) & 0x1f1; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0020; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0040; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x0060; break; } diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 28ef46c91f62..ee2ba574952b 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -736,16 +736,16 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, return -EINVAL; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface_ctl |= 0x20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface_ctl |= 0x40; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface_ctl |= 0x60; break; } @@ -817,8 +817,8 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, wm8978->sysclk == WM8978_MCLK ? ", consider using PLL" : ""); - dev_dbg(codec->dev, "%s: fmt %d, rate %u, MCLK divisor #%d\n", __func__, - params_format(params), params_rate(params), best); + dev_dbg(codec->dev, "%s: width %d, rate %u, MCLK divisor #%d\n", __func__, + params_width(params), params_rate(params), best); /* MCLK divisor mask = 0xe0 */ snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, best << 5); diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 19d5baa38f5c..ac5defda8824 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -719,22 +719,22 @@ static int wm8983_hw_params(struct snd_pcm_substream *substream, wm8983->bclk = ret; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: blen = 0x0; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: blen = 0x1; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: blen = 0x2; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: blen = 0x3; break; default: dev_err(dai->dev, "Unsupported word length %u\n", - params_format(params)); + params_width(params)); return -EINVAL; } diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 0f5780c09f3a..ee380190399f 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -698,22 +698,22 @@ static int wm8985_hw_params(struct snd_pcm_substream *substream, if ((int)wm8985->bclk < 0) return wm8985->bclk; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: blen = 0x0; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: blen = 0x1; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: blen = 0x2; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: blen = 0x3; break; default: dev_err(dai->dev, "Unsupported word length %u\n", - params_format(params)); + params_width(params)); return -EINVAL; } @@ -980,9 +980,6 @@ static int wm8985_resume(struct snd_soc_codec *codec) static int wm8985_remove(struct snd_soc_codec *codec) { - struct wm8985_priv *wm8985; - - wm8985 = snd_soc_codec_get_drvdata(codec); wm8985_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index d3fea46d58e8..a5130d965146 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -687,16 +687,16 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, } /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0008; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x000c; break; } diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index b5c1f0f07058..03e43e3f395e 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1073,16 +1073,16 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream, audio1 &= ~WM8990_AIF_WL_MASK; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: audio1 |= WM8990_AIF_WL_20BITS; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: audio1 |= WM8990_AIF_WL_24BITS; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: audio1 |= WM8990_AIF_WL_32BITS; break; } diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index b8fd284fc0c0..d0be89731cdb 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1081,16 +1081,16 @@ static int wm8991_hw_params(struct snd_pcm_substream *substream, audio1 &= ~WM8991_AIF_WL_MASK; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: audio1 |= WM8991_AIF_WL_20BITS; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: audio1 |= WM8991_AIF_WL_24BITS; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: audio1 |= WM8991_AIF_WL_32BITS; break; } diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index f825dc04ebe1..93b14eda355a 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1214,19 +1214,19 @@ static int wm8993_hw_params(struct snd_pcm_substream *substream, wm8993->tdm_slots, wm8993->tdm_width); wm8993->bclk *= wm8993->tdm_width * wm8993->tdm_slots; } else { - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: wm8993->bclk *= 16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: wm8993->bclk *= 20; aif1 |= 0x8; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: wm8993->bclk *= 24; aif1 |= 0x10; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: wm8993->bclk *= 32; aif1 |= 0x18; break; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247b39013fba..6cc0566dc29a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2815,19 +2815,19 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, } bclk_rate = params_rate(params); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: bclk_rate *= 16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: bclk_rate *= 20; aif1 |= 0x20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: bclk_rate *= 24; aif1 |= 0x40; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: bclk_rate *= 32; aif1 |= 0x60; break; @@ -2966,16 +2966,16 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, return 0; } - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: aif1 |= 0x20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: aif1 |= 0x40; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: aif1 |= 0x60; break; default: @@ -3296,12 +3296,8 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) /* We need an array of texts for the enum API */ wm8994->drc_texts = devm_kzalloc(wm8994->hubs.codec->dev, sizeof(char *) * pdata->num_drc_cfgs, GFP_KERNEL); - if (!wm8994->drc_texts) { - dev_err(wm8994->hubs.codec->dev, - "Failed to allocate %d DRC config texts\n", - pdata->num_drc_cfgs); + if (!wm8994->drc_texts) return; - } for (i = 0; i < pdata->num_drc_cfgs; i++) wm8994->drc_texts[i] = pdata->drc_cfgs[i].name; @@ -3505,6 +3501,7 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) return IRQ_HANDLED; } +/* Should be called with accdet_lock held */ static void wm1811_micd_stop(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -3512,14 +3509,10 @@ static void wm1811_micd_stop(struct snd_soc_codec *codec) if (!wm8994->jackdet) return; - mutex_lock(&wm8994->accdet_lock); - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_JACK); - mutex_unlock(&wm8994->accdet_lock); - if (wm8994->wm8994->pdata.jd_ext_cap) snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); @@ -3560,10 +3553,10 @@ static void wm8958_open_circuit_work(struct work_struct *work) open_circuit_work.work); struct device *dev = wm8994->wm8994->dev; - wm1811_micd_stop(wm8994->hubs.codec); - mutex_lock(&wm8994->accdet_lock); + wm1811_micd_stop(wm8994->hubs.codec); + dev_dbg(dev, "Reporting open circuit\n"); wm8994->jack_mic = false; diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 863a2c38bcb5..cae4ac5a5730 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1597,21 +1597,21 @@ static int wm8995_hw_params(struct snd_pcm_substream *substream, return bclk_rate; aif1 = 0; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: aif1 |= (0x1 << WM8995_AIF1_WL_SHIFT); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: aif1 |= (0x2 << WM8995_AIF1_WL_SHIFT); break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: aif1 |= (0x3 << WM8995_AIF1_WL_SHIFT); break; default: dev_err(dai->dev, "Unsupported word length %u\n", - params_format(params)); + params_width(params)); return -EINVAL; } diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 69266332760e..f16ff4f56923 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -620,15 +620,12 @@ static int bg_event(struct snd_soc_dapm_widget *w, static int cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - int ret = 0; - switch (event) { case SND_SOC_DAPM_POST_PMU: msleep(5); break; default: WARN(1, "Invalid event %d\n", event); - ret = -EINVAL; } return 0; @@ -690,8 +687,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); u16 val, mask; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index bb9b47b956aa..ab33fe596519 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -967,6 +967,7 @@ static struct snd_soc_dai_driver wm8997_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm8997-aif2", @@ -988,6 +989,7 @@ static struct snd_soc_dai_driver wm8997_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm8997-slim1", diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 185eb97769e7..0cdc9e2184ab 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1029,19 +1029,19 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream, /* Otherwise work out a BCLK from the sample size */ wm9081->bclk = 2 * wm9081->fs; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: wm9081->bclk *= 16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: wm9081->bclk *= 20; aif2 |= 0x4; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: wm9081->bclk *= 24; aif2 |= 0x8; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: wm9081->bclk *= 32; aif2 |= 0xc; break; diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 87934171f063..a13f0725611a 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -613,10 +613,8 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, int ret; wm9090 = devm_kzalloc(&i2c->dev, sizeof(*wm9090), GFP_KERNEL); - if (wm9090 == NULL) { - dev_err(&i2c->dev, "Can not allocate memory\n"); + if (!wm9090) return -ENOMEM; - } wm9090->regmap = devm_regmap_init_i2c(i2c, &wm9090_regmap); if (IS_ERR(wm9090->regmap)) { diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 2a9c6d11330c..bddee30a4bc7 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -953,16 +953,16 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: reg |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: reg |= 0x0008; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: reg |= 0x000c; break; } diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 060027182dcb..f412a9911a75 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1382,7 +1382,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, int ret; int val; - dsp->card = codec->card; + dsp->card = codec->component.card; switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -1617,7 +1617,7 @@ int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; - dsp->card = codec->card; + dsp->card = codec->component.card; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -1758,3 +1758,5 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_init); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 916817fe6632..374537d5e179 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -183,10 +183,8 @@ static void wm_hubs_dcs_cache_set(struct snd_soc_codec *codec, u16 dcs_cfg) return; cache = devm_kzalloc(codec->dev, sizeof(*cache), GFP_KERNEL); - if (!cache) { - dev_err(codec->dev, "Failed to allocate DCS cache entry\n"); + if (!cache) return; - } cache->left = snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME); cache->left &= WM8993_HPOUT1L_VOL_MASK; diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 50a098749b9e..d69510c53239 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,12 +1,29 @@ config SND_DAVINCI_SOC - tristate "SoC Audio for TI DAVINCI or AM33XX/AM43XX chips" - depends on ARCH_DAVINCI || SOC_AM33XX || SOC_AM43XX + tristate "SoC Audio for TI DAVINCI" + depends on ARCH_DAVINCI + +config SND_EDMA_SOC + tristate "SoC Audio for Texas Instruments chips using eDMA (AM33XX/43XX)" + depends on SOC_AM33XX || SOC_AM43XX + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M here if you want audio support for TI SoC which uses eDMA. + The following line of SoCs are supported by this platform driver: + - AM335x + - AM437x/AM438x config SND_DAVINCI_SOC_I2S tristate config SND_DAVINCI_SOC_MCASP - tristate + tristate "Multichannel Audio Serial Port (McASP) support" + depends on SND_DAVINCI_SOC || SND_OMAP_SOC || SND_EDMA_SOC + help + Say Y or M here if you want to have support for McASP IP found in + various Texas Instruments SoCs like: + - daVinci devices + - Sitara line of SoCs (AM335x, AM438x, etc) + - DRA7x devices config SND_DAVINCI_SOC_VCIF tristate @@ -18,7 +35,7 @@ config SND_DAVINCI_SOC_GENERIC_EVM config SND_AM33XX_SOC_EVM tristate "SoC Audio for the AM33XX chip based boards" - depends on SND_DAVINCI_SOC && SOC_AM33XX && I2C + depends on SND_EDMA_SOC && SOC_AM33XX && I2C select SND_DAVINCI_SOC_GENERIC_EVM help Say Y or M if you want to add support for SoC audio on AM33XX diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 744d4d9a0184..09bf2ba92d38 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,10 +1,12 @@ # DAVINCI Platform Support snd-soc-davinci-objs := davinci-pcm.o +snd-soc-edma-objs := edma-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o snd-soc-davinci-vcif-objs:= davinci-vcif.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o +obj-$(CONFIG_SND_EDMA_SOC) += snd-soc-edma.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9afb14629a17..c28508da34cf 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include <linux/of_platform.h> #include <linux/of_device.h> +#include <sound/asoundef.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -36,6 +37,7 @@ #include <sound/omap-pcm.h> #include "davinci-pcm.h" +#include "edma-pcm.h" #include "davinci-mcasp.h" #define MCASP_MAX_AFIFO_DEPTH 64 @@ -63,6 +65,7 @@ struct davinci_mcasp { u8 num_serializer; u8 *serial_dir; u8 version; + u8 bclk_div; u16 bclk_lrclk_ratio; int streams; @@ -417,6 +420,7 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div ACLKXDIV(div - 1), ACLKXDIV_MASK); mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRDIV(div - 1), ACLKRDIV_MASK); + mcasp->bclk_div = div; break; case 2: /* BCLK/LRCLK ratio */ @@ -637,8 +641,12 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) } /* S/PDIF */ -static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) +static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, + unsigned int rate) { + u32 cs_value = 0; + u8 *cs_bytes = (u8*) &cs_value; + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); @@ -660,6 +668,46 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) /* Enable the DIT */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); + /* Set S/PDIF channel status bits */ + cs_bytes[0] = IEC958_AES0_CON_NOT_COPYRIGHT; + cs_bytes[1] = IEC958_AES1_CON_PCM_CODER; + + switch (rate) { + case 22050: + cs_bytes[3] |= IEC958_AES3_CON_FS_22050; + break; + case 24000: + cs_bytes[3] |= IEC958_AES3_CON_FS_24000; + break; + case 32000: + cs_bytes[3] |= IEC958_AES3_CON_FS_32000; + break; + case 44100: + cs_bytes[3] |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + cs_bytes[3] |= IEC958_AES3_CON_FS_48000; + break; + case 88200: + cs_bytes[3] |= IEC958_AES3_CON_FS_88200; + break; + case 96000: + cs_bytes[3] |= IEC958_AES3_CON_FS_96000; + break; + case 176400: + cs_bytes[3] |= IEC958_AES3_CON_FS_176400; + break; + case 192000: + cs_bytes[3] |= IEC958_AES3_CON_FS_192000; + break; + default: + printk(KERN_WARNING "unsupported sampling rate: %d\n", rate); + return -EINVAL; + } + + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, cs_value); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, cs_value); + return 0; } @@ -675,15 +723,22 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int period_size = params_period_size(params); int ret; - /* If mcasp is BCLK master we need to set BCLK divider */ - if (mcasp->bclk_master) { + /* + * If mcasp is BCLK master, and a BCLK divider was not provided by + * the machine driver, we need to calculate the ratio. + */ + if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { unsigned int bclk_freq = snd_soc_params_to_bclk(params); + unsigned int div = mcasp->sysclk_freq / bclk_freq; if (mcasp->sysclk_freq % bclk_freq != 0) { - dev_err(mcasp->dev, "Can't produce required BCLK\n"); - return -EINVAL; + if (((mcasp->sysclk_freq / div) - bclk_freq) > + (bclk_freq - (mcasp->sysclk_freq / (div+1)))) + div++; + dev_warn(mcasp->dev, + "Inaccurate BCLK: %u Hz / %u != %u Hz\n", + mcasp->sysclk_freq, div, bclk_freq); } - davinci_mcasp_set_clkdiv( - cpu_dai, 1, mcasp->sysclk_freq / bclk_freq); + davinci_mcasp_set_clkdiv(cpu_dai, 1, div); } ret = mcasp_common_hw_param(mcasp, substream->stream, @@ -692,7 +747,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return ret; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - ret = mcasp_dit_hw_param(mcasp); + ret = mcasp_dit_hw_param(mcasp, params_rate(params)); else ret = mcasp_i2s_hw_param(mcasp, substream->stream); @@ -720,6 +775,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_U24_LE: case SNDRV_PCM_FORMAT_S24_LE: + dma_params->data_type = 4; + word_length = 24; + break; + case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: dma_params->data_type = 4; @@ -778,7 +837,7 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - if (mcasp->version == MCASP_VERSION_4) { + if (mcasp->version >= MCASP_VERSION_3) { /* Using dmaengine PCM */ dai->playback_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; @@ -1223,14 +1282,28 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err; switch (mcasp->version) { +#if IS_BUILTIN(CONFIG_SND_DAVINCI_SOC) || \ + (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ + IS_MODULE(CONFIG_SND_DAVINCI_SOC)) case MCASP_VERSION_1: case MCASP_VERSION_2: - case MCASP_VERSION_3: ret = davinci_soc_platform_register(&pdev->dev); break; +#endif +#if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \ + (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ + IS_MODULE(CONFIG_SND_EDMA_SOC)) + case MCASP_VERSION_3: + ret = edma_pcm_platform_register(&pdev->dev); + break; +#endif +#if IS_BUILTIN(CONFIG_SND_OMAP_SOC) || \ + (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ + IS_MODULE(CONFIG_SND_OMAP_SOC)) case MCASP_VERSION_4: ret = omap_pcm_platform_register(&pdev->dev); break; +#endif default: dev_err(&pdev->dev, "Invalid McASP version: %d\n", mcasp->version); diff --git a/sound/soc/davinci/edma-pcm.c b/sound/soc/davinci/edma-pcm.c index d38afb1c61ae..605e643133db 100644 --- a/sound/soc/davinci/edma-pcm.c +++ b/sound/soc/davinci/edma-pcm.c @@ -28,8 +28,8 @@ static const struct snd_pcm_hardware edma_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP | SNDRV_PCM_INFO_INTERLEAVED, .buffer_bytes_max = 128 * 1024, .period_bytes_min = 32, diff --git a/sound/soc/davinci/edma-pcm.h b/sound/soc/davinci/edma-pcm.h index 894c378c0f74..b0957744851c 100644 --- a/sound/soc/davinci/edma-pcm.h +++ b/sound/soc/davinci/edma-pcm.h @@ -20,6 +20,13 @@ #ifndef __EDMA_PCM_H__ #define __EDMA_PCM_H__ +#if IS_ENABLED(CONFIG_SND_EDMA_SOC) int edma_pcm_platform_register(struct device *dev); +#else +static inline int edma_pcm_platform_register(struct device *dev) +{ + return 0; +} +#endif /* CONFIG_SND_EDMA_SOC */ #endif /* __EDMA_PCM_H__ */ diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 37933629cbed..f54a8fc99291 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -2,9 +2,20 @@ menu "SoC Audio for Freescale CPUs" comment "Common SoC Audio options for Freescale CPUs:" +config SND_SOC_FSL_ASRC + tristate "Asynchronous Sample Rate Converter (ASRC) module support" + select REGMAP_MMIO + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y if you want to add Asynchronous Sample Rate Converter (ASRC) + support for the Freescale CPUs. + This option is only useful for out-of-tree drivers since + in-tree drivers select it automatically. + config SND_SOC_FSL_SAI tristate "Synchronous Audio Interface (SAI) module support" select REGMAP_MMIO + select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y if you want to add Synchronous Audio Interface (SAI) @@ -15,7 +26,7 @@ config SND_SOC_FSL_SAI config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC + select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) select REGMAP_MMIO help Say Y if you want to add Synchronous Serial Interface (SSI) @@ -27,7 +38,7 @@ config SND_SOC_FSL_SPDIF tristate "Sony/Philips Digital Interface module support" select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC + select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) help Say Y if you want to add Sony/Philips Digital Interface (SPDIF) support for the Freescale CPUs. @@ -37,6 +48,7 @@ config SND_SOC_FSL_SPDIF config SND_SOC_FSL_ESAI tristate "Enhanced Serial Audio Interface (ESAI) module support" select REGMAP_MMIO + select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_FSL_UTILS help Say Y if you want to add Enhanced Synchronous Audio Interface diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index db254e358c18..9ff59267eac9 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -11,6 +11,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-y := fsl_ssi.o snd-soc-fsl-ssi-$(CONFIG_DEBUG_FS) += fsl_ssi_dbg.o @@ -18,6 +19,7 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c new file mode 100644 index 000000000000..822110420b71 --- /dev/null +++ b/sound/soc/fsl/fsl_asrc.c @@ -0,0 +1,995 @@ +/* + * 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. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/dma-mapping.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/platform_data/dma-imx.h> +#include <linux/pm_runtime.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> + +#include "fsl_asrc.h" + +#define IDEAL_RATIO_DECIMAL_DEPTH 26 + +#define pair_err(fmt, ...) \ + dev_err(&asrc_priv->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) + +#define pair_dbg(fmt, ...) \ + dev_dbg(&asrc_priv->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) + +/* Sample rates are aligned with that defined in pcm.h file */ +static const u8 process_option[][8][2] = { + /* 32kHz 44.1kHz 48kHz 64kHz 88.2kHz 96kHz 176kHz 192kHz */ + {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 5512Hz */ + {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 8kHz */ + {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 11025Hz */ + {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 16kHz */ + {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 22050Hz */ + {{0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0},}, /* 32kHz */ + {{0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 44.1kHz */ + {{0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 48kHz */ + {{1, 2}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0},}, /* 64kHz */ + {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 88.2kHz */ + {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 96kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 176kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 192kHz */ +}; + +/* Corresponding to process_option */ +static int supported_input_rate[] = { + 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, + 96000, 176400, 192000, +}; + +static int supported_asrc_rate[] = { + 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, +}; + +/** + * The following tables map the relationship between asrc_inclk/asrc_outclk in + * fsl_asrc.h and the registers of ASRCSR + */ +static unsigned char input_clk_map_imx35[] = { + 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf, +}; + +static unsigned char output_clk_map_imx35[] = { + 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf, +}; + +/* i.MX53 uses the same map for input and output */ +static unsigned char input_clk_map_imx53[] = { +/* 0x0 0x1 0x2 0x3 0x4 0x5 0x6 0x7 0x8 0x9 0xa 0xb 0xc 0xd 0xe 0xf */ + 0x0, 0x1, 0x2, 0x7, 0x4, 0x5, 0x6, 0x3, 0x8, 0x9, 0xa, 0xb, 0xc, 0xf, 0xe, 0xd, +}; + +static unsigned char output_clk_map_imx53[] = { +/* 0x0 0x1 0x2 0x3 0x4 0x5 0x6 0x7 0x8 0x9 0xa 0xb 0xc 0xd 0xe 0xf */ + 0x8, 0x9, 0xa, 0x7, 0xc, 0x5, 0x6, 0xb, 0x0, 0x1, 0x2, 0x3, 0x4, 0xf, 0xe, 0xd, +}; + +static unsigned char *clk_map[2]; + +/** + * Request ASRC pair + * + * It assigns pair by the order of A->C->B because allocation of pair B, + * within range [ANCA, ANCA+ANCB-1], depends on the channels of pair A + * while pair A and pair C are comparatively independent. + */ +static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair) +{ + enum asrc_pair_index index = ASRC_INVALID_PAIR; + struct fsl_asrc *asrc_priv = pair->asrc_priv; + struct device *dev = &asrc_priv->pdev->dev; + unsigned long lock_flags; + int i, ret = 0; + + spin_lock_irqsave(&asrc_priv->lock, lock_flags); + + for (i = ASRC_PAIR_A; i < ASRC_PAIR_MAX_NUM; i++) { + if (asrc_priv->pair[i] != NULL) + continue; + + index = i; + + if (i != ASRC_PAIR_B) + break; + } + + if (index == ASRC_INVALID_PAIR) { + dev_err(dev, "all pairs are busy now\n"); + ret = -EBUSY; + } else if (asrc_priv->channel_avail < channels) { + dev_err(dev, "can't afford required channels: %d\n", channels); + ret = -EINVAL; + } else { + asrc_priv->channel_avail -= channels; + asrc_priv->pair[index] = pair; + pair->channels = channels; + pair->index = index; + } + + spin_unlock_irqrestore(&asrc_priv->lock, lock_flags); + + return ret; +} + +/** + * Release ASRC pair + * + * It clears the resource from asrc_priv and releases the occupied channels. + */ +static void fsl_asrc_release_pair(struct fsl_asrc_pair *pair) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + unsigned long lock_flags; + + /* Make sure the pair is disabled */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ASRCEi_MASK(index), 0); + + spin_lock_irqsave(&asrc_priv->lock, lock_flags); + + asrc_priv->channel_avail += pair->channels; + asrc_priv->pair[index] = NULL; + pair->error = 0; + + spin_unlock_irqrestore(&asrc_priv->lock, lock_flags); +} + +/** + * Configure input and output thresholds + */ +static void fsl_asrc_set_watermarks(struct fsl_asrc_pair *pair, u32 in, u32 out) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + + regmap_update_bits(asrc_priv->regmap, REG_ASRMCR(index), + ASRMCRi_EXTTHRSHi_MASK | + ASRMCRi_INFIFO_THRESHOLD_MASK | + ASRMCRi_OUTFIFO_THRESHOLD_MASK, + ASRMCRi_EXTTHRSHi | + ASRMCRi_INFIFO_THRESHOLD(in) | + ASRMCRi_OUTFIFO_THRESHOLD(out)); +} + +/** + * Calculate the total divisor between asrck clock rate and sample rate + * + * It follows the formula clk_rate = samplerate * (2 ^ prescaler) * divider + */ +static u32 fsl_asrc_cal_asrck_divisor(struct fsl_asrc_pair *pair, u32 div) +{ + u32 ps; + + /* Calculate the divisors: prescaler [2^0, 2^7], divder [1, 8] */ + for (ps = 0; div > 8; ps++) + div >>= 1; + + return ((div - 1) << ASRCDRi_AxCPi_WIDTH) | ps; +} + +/** + * Calculate and set the ratio for Ideal Ratio mode only + * + * The ratio is a 32-bit fixed point value with 26 fractional bits. + */ +static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair, + int inrate, int outrate) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + unsigned long ratio; + int i; + + if (!outrate) { + pair_err("output rate should not be zero\n"); + return -EINVAL; + } + + /* Calculate the intergal part of the ratio */ + ratio = (inrate / outrate) << IDEAL_RATIO_DECIMAL_DEPTH; + + /* ... and then the 26 depth decimal part */ + inrate %= outrate; + + for (i = 1; i <= IDEAL_RATIO_DECIMAL_DEPTH; i++) { + inrate <<= 1; + + if (inrate < outrate) + continue; + + ratio |= 1 << (IDEAL_RATIO_DECIMAL_DEPTH - i); + inrate -= outrate; + + if (!inrate) + break; + } + + regmap_write(asrc_priv->regmap, REG_ASRIDRL(index), ratio); + regmap_write(asrc_priv->regmap, REG_ASRIDRH(index), ratio >> 24); + + return 0; +} + +/** + * Configure the assigned ASRC pair + * + * It configures those ASRC registers according to a configuration instance + * of struct asrc_config which includes in/output sample rate, width, channel + * and clock settings. + */ +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) +{ + struct asrc_config *config = pair->config; + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + u32 inrate, outrate, indiv, outdiv; + u32 clk_index[2], div[2]; + int in, out, channels; + struct clk *clk; + bool ideal; + + if (!config) { + pair_err("invalid pair config\n"); + return -EINVAL; + } + + /* Validate channels */ + if (config->channel_num < 1 || config->channel_num > 10) { + pair_err("does not support %d channels\n", config->channel_num); + return -EINVAL; + } + + /* Validate output width */ + if (config->output_word_width == ASRC_WIDTH_8_BIT) { + pair_err("does not support 8bit width output\n"); + return -EINVAL; + } + + inrate = config->input_sample_rate; + outrate = config->output_sample_rate; + ideal = config->inclk == INCLK_NONE; + + /* Validate input and output sample rates */ + for (in = 0; in < ARRAY_SIZE(supported_input_rate); in++) + if (inrate == supported_input_rate[in]) + break; + + if (in == ARRAY_SIZE(supported_input_rate)) { + pair_err("unsupported input sample rate: %dHz\n", inrate); + return -EINVAL; + } + + for (out = 0; out < ARRAY_SIZE(supported_asrc_rate); out++) + if (outrate == supported_asrc_rate[out]) + break; + + if (out == ARRAY_SIZE(supported_asrc_rate)) { + pair_err("unsupported output sample rate: %dHz\n", outrate); + return -EINVAL; + } + + /* Validate input and output clock sources */ + clk_index[IN] = clk_map[IN][config->inclk]; + clk_index[OUT] = clk_map[OUT][config->outclk]; + + /* We only have output clock for ideal ratio mode */ + clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; + + div[IN] = clk_get_rate(clk) / inrate; + if (div[IN] == 0) { + pair_err("failed to support input sample rate %dHz by asrck_%x\n", + inrate, clk_index[ideal ? OUT : IN]); + return -EINVAL; + } + + clk = asrc_priv->asrck_clk[clk_index[OUT]]; + + /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ + if (ideal) + div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; + else + div[OUT] = clk_get_rate(clk) / outrate; + + if (div[OUT] == 0) { + pair_err("failed to support output sample rate %dHz by asrck_%x\n", + outrate, clk_index[OUT]); + return -EINVAL; + } + + /* Set the channel number */ + channels = config->channel_num; + + if (asrc_priv->channel_bits < 4) + channels /= 2; + + /* Update channels for current pair */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCNCR, + ASRCNCR_ANCi_MASK(index, asrc_priv->channel_bits), + ASRCNCR_ANCi(index, channels, asrc_priv->channel_bits)); + + /* Default setting: Automatic selection for processing mode */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ATSi_MASK(index), ASRCTR_ATS(index)); + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_USRi_MASK(index), 0); + + /* Set the input and output clock sources */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCSR, + ASRCSR_AICSi_MASK(index) | ASRCSR_AOCSi_MASK(index), + ASRCSR_AICS(index, clk_index[IN]) | + ASRCSR_AOCS(index, clk_index[OUT])); + + /* Calculate the input clock divisors */ + indiv = fsl_asrc_cal_asrck_divisor(pair, div[IN]); + outdiv = fsl_asrc_cal_asrck_divisor(pair, div[OUT]); + + /* Suppose indiv and outdiv includes prescaler, so add its MASK too */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCDR(index), + ASRCDRi_AOCPi_MASK(index) | ASRCDRi_AICPi_MASK(index) | + ASRCDRi_AOCDi_MASK(index) | ASRCDRi_AICDi_MASK(index), + ASRCDRi_AOCP(index, outdiv) | ASRCDRi_AICP(index, indiv)); + + /* Implement word_width configurations */ + regmap_update_bits(asrc_priv->regmap, REG_ASRMCR1(index), + ASRMCR1i_OW16_MASK | ASRMCR1i_IWD_MASK, + ASRMCR1i_OW16(config->output_word_width) | + ASRMCR1i_IWD(config->input_word_width)); + + /* Enable BUFFER STALL */ + regmap_update_bits(asrc_priv->regmap, REG_ASRMCR(index), + ASRMCRi_BUFSTALLi_MASK, ASRMCRi_BUFSTALLi); + + /* Set default thresholds for input and output FIFO */ + fsl_asrc_set_watermarks(pair, ASRC_INPUTFIFO_THRESHOLD, + ASRC_INPUTFIFO_THRESHOLD); + + /* Configure the followings only for Ideal Ratio mode */ + if (!ideal) + return 0; + + /* Clear ASTSx bit to use Ideal Ratio mode */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ATSi_MASK(index), 0); + + /* Enable Ideal Ratio mode */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_IDRi_MASK(index) | ASRCTR_USRi_MASK(index), + ASRCTR_IDR(index) | ASRCTR_USR(index)); + + /* Apply configurations for pre- and post-processing */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCFG, + ASRCFG_PREMODi_MASK(index) | ASRCFG_POSTMODi_MASK(index), + ASRCFG_PREMOD(index, process_option[in][out][0]) | + ASRCFG_POSTMOD(index, process_option[in][out][1])); + + return fsl_asrc_set_ideal_ratio(pair, inrate, outrate); +} + +/** + * Start the assigned ASRC pair + * + * It enables the assigned pair and makes it stopped at the stall level. + */ +static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + int reg, retry = 10, i; + + /* Enable the current pair */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ASRCEi_MASK(index), ASRCTR_ASRCE(index)); + + /* Wait for status of initialization */ + do { + udelay(5); + regmap_read(asrc_priv->regmap, REG_ASRCFG, ®); + reg &= ASRCFG_INIRQi_MASK(index); + } while (!reg && --retry); + + /* Make the input fifo to ASRC STALL level */ + regmap_read(asrc_priv->regmap, REG_ASRCNCR, ®); + for (i = 0; i < pair->channels * 4; i++) + regmap_write(asrc_priv->regmap, REG_ASRDI(index), 0); + + /* Enable overload interrupt */ + regmap_write(asrc_priv->regmap, REG_ASRIER, ASRIER_AOLIE); +} + +/** + * Stop the assigned ASRC pair + */ +static void fsl_asrc_stop_pair(struct fsl_asrc_pair *pair) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + + /* Stop the current pair */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ASRCEi_MASK(index), 0); +} + +/** + * Get DMA channel according to the pair and direction. + */ +struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + char name[4]; + + sprintf(name, "%cx%c", dir == IN ? 'r' : 't', index + 'a'); + + return dma_request_slave_channel(&asrc_priv->pdev->dev, name); +} +EXPORT_SYMBOL_GPL(fsl_asrc_get_dma_channel); + +static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai); + int width = snd_pcm_format_width(params_format(params)); + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + unsigned int channels = params_channels(params); + unsigned int rate = params_rate(params); + struct asrc_config config; + int word_width, ret; + + ret = fsl_asrc_request_pair(channels, pair); + if (ret) { + dev_err(dai->dev, "fail to request asrc pair\n"); + return ret; + } + + pair->config = &config; + + if (width == 16) + width = ASRC_WIDTH_16_BIT; + else + width = ASRC_WIDTH_24_BIT; + + if (asrc_priv->asrc_width == 16) + word_width = ASRC_WIDTH_16_BIT; + else + word_width = ASRC_WIDTH_24_BIT; + + config.pair = pair->index; + config.channel_num = channels; + config.inclk = INCLK_NONE; + config.outclk = OUTCLK_ASRCK1_CLK; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + config.input_word_width = width; + config.output_word_width = word_width; + config.input_sample_rate = rate; + config.output_sample_rate = asrc_priv->asrc_rate; + } else { + config.input_word_width = word_width; + config.output_word_width = width; + config.input_sample_rate = asrc_priv->asrc_rate; + config.output_sample_rate = rate; + } + + ret = fsl_asrc_config_pair(pair); + if (ret) { + dev_err(dai->dev, "fail to config asrc pair\n"); + return ret; + } + + return 0; +} + +static int fsl_asrc_dai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + + if (pair) + fsl_asrc_release_pair(pair); + + return 0; +} + +static int fsl_asrc_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + fsl_asrc_start_pair(pair); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + fsl_asrc_stop_pair(pair); + break; + default: + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops fsl_asrc_dai_ops = { + .hw_params = fsl_asrc_dai_hw_params, + .hw_free = fsl_asrc_dai_hw_free, + .trigger = fsl_asrc_dai_trigger, +}; + +static int fsl_asrc_dai_probe(struct snd_soc_dai *dai) +{ + struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &asrc_priv->dma_params_tx, + &asrc_priv->dma_params_rx); + + return 0; +} + +#define FSL_ASRC_RATES SNDRV_PCM_RATE_8000_192000 +#define FSL_ASRC_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE) + +static struct snd_soc_dai_driver fsl_asrc_dai = { + .probe = fsl_asrc_dai_probe, + .playback = { + .stream_name = "ASRC-Playback", + .channels_min = 1, + .channels_max = 10, + .rates = FSL_ASRC_RATES, + .formats = FSL_ASRC_FORMATS, + }, + .capture = { + .stream_name = "ASRC-Capture", + .channels_min = 1, + .channels_max = 10, + .rates = FSL_ASRC_RATES, + .formats = FSL_ASRC_FORMATS, + }, + .ops = &fsl_asrc_dai_ops, +}; + +static const struct snd_soc_component_driver fsl_asrc_component = { + .name = "fsl-asrc-dai", +}; + +static bool fsl_asrc_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ASRCTR: + case REG_ASRIER: + case REG_ASRCNCR: + case REG_ASRCFG: + case REG_ASRCSR: + case REG_ASRCDR1: + case REG_ASRCDR2: + case REG_ASRSTR: + case REG_ASRPM1: + case REG_ASRPM2: + case REG_ASRPM3: + case REG_ASRPM4: + case REG_ASRPM5: + case REG_ASRTFR1: + case REG_ASRCCR: + case REG_ASRDOA: + case REG_ASRDOB: + case REG_ASRDOC: + case REG_ASRIDRHA: + case REG_ASRIDRLA: + case REG_ASRIDRHB: + case REG_ASRIDRLB: + case REG_ASRIDRHC: + case REG_ASRIDRLC: + case REG_ASR76K: + case REG_ASR56K: + case REG_ASRMCRA: + case REG_ASRFSTA: + case REG_ASRMCRB: + case REG_ASRFSTB: + case REG_ASRMCRC: + case REG_ASRFSTC: + case REG_ASRMCR1A: + case REG_ASRMCR1B: + case REG_ASRMCR1C: + return true; + default: + return false; + } +} + +static bool fsl_asrc_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ASRSTR: + case REG_ASRDIA: + case REG_ASRDIB: + case REG_ASRDIC: + case REG_ASRDOA: + case REG_ASRDOB: + case REG_ASRDOC: + case REG_ASRFSTA: + case REG_ASRFSTB: + case REG_ASRFSTC: + case REG_ASRCFG: + return true; + default: + return false; + } +} + +static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ASRCTR: + case REG_ASRIER: + case REG_ASRCNCR: + case REG_ASRCFG: + case REG_ASRCSR: + case REG_ASRCDR1: + case REG_ASRCDR2: + case REG_ASRSTR: + case REG_ASRPM1: + case REG_ASRPM2: + case REG_ASRPM3: + case REG_ASRPM4: + case REG_ASRPM5: + case REG_ASRTFR1: + case REG_ASRCCR: + case REG_ASRDIA: + case REG_ASRDIB: + case REG_ASRDIC: + case REG_ASRIDRHA: + case REG_ASRIDRLA: + case REG_ASRIDRHB: + case REG_ASRIDRLB: + case REG_ASRIDRHC: + case REG_ASRIDRLC: + case REG_ASR76K: + case REG_ASR56K: + case REG_ASRMCRA: + case REG_ASRMCRB: + case REG_ASRMCRC: + case REG_ASRMCR1A: + case REG_ASRMCR1B: + case REG_ASRMCR1C: + return true; + default: + return false; + } +} + +static struct regmap_config fsl_asrc_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + + .max_register = REG_ASRMCR1C, + .readable_reg = fsl_asrc_readable_reg, + .volatile_reg = fsl_asrc_volatile_reg, + .writeable_reg = fsl_asrc_writeable_reg, + .cache_type = REGCACHE_RBTREE, +}; + +/** + * Initialize ASRC registers with a default configurations + */ +static int fsl_asrc_init(struct fsl_asrc *asrc_priv) +{ + /* Halt ASRC internal FP when input FIFO needs data for pair A, B, C */ + regmap_write(asrc_priv->regmap, REG_ASRCTR, ASRCTR_ASRCEN); + + /* Disable interrupt by default */ + regmap_write(asrc_priv->regmap, REG_ASRIER, 0x0); + + /* Apply recommended settings for parameters from Reference Manual */ + regmap_write(asrc_priv->regmap, REG_ASRPM1, 0x7fffff); + regmap_write(asrc_priv->regmap, REG_ASRPM2, 0x255555); + regmap_write(asrc_priv->regmap, REG_ASRPM3, 0xff7280); + regmap_write(asrc_priv->regmap, REG_ASRPM4, 0xff7280); + regmap_write(asrc_priv->regmap, REG_ASRPM5, 0xff7280); + + /* Base address for task queue FIFO. Set to 0x7C */ + regmap_update_bits(asrc_priv->regmap, REG_ASRTFR1, + ASRTFR1_TF_BASE_MASK, ASRTFR1_TF_BASE(0xfc)); + + /* Set the processing clock for 76KHz to 133M */ + regmap_write(asrc_priv->regmap, REG_ASR76K, 0x06D6); + + /* Set the processing clock for 56KHz to 133M */ + return regmap_write(asrc_priv->regmap, REG_ASR56K, 0x0947); +} + +/** + * Interrupt handler for ASRC + */ +static irqreturn_t fsl_asrc_isr(int irq, void *dev_id) +{ + struct fsl_asrc *asrc_priv = (struct fsl_asrc *)dev_id; + struct device *dev = &asrc_priv->pdev->dev; + enum asrc_pair_index index; + u32 status; + + regmap_read(asrc_priv->regmap, REG_ASRSTR, &status); + + /* Clean overload error */ + regmap_write(asrc_priv->regmap, REG_ASRSTR, ASRSTR_AOLE); + + /* + * We here use dev_dbg() for all exceptions because ASRC itself does + * not care if FIFO overflowed or underrun while a warning in the + * interrupt would result a ridged conversion. + */ + for (index = ASRC_PAIR_A; index < ASRC_PAIR_MAX_NUM; index++) { + if (!asrc_priv->pair[index]) + continue; + + if (status & ASRSTR_ATQOL) { + asrc_priv->pair[index]->error |= ASRC_TASK_Q_OVERLOAD; + dev_dbg(dev, "ASRC Task Queue FIFO overload\n"); + } + + if (status & ASRSTR_AOOL(index)) { + asrc_priv->pair[index]->error |= ASRC_OUTPUT_TASK_OVERLOAD; + pair_dbg("Output Task Overload\n"); + } + + if (status & ASRSTR_AIOL(index)) { + asrc_priv->pair[index]->error |= ASRC_INPUT_TASK_OVERLOAD; + pair_dbg("Input Task Overload\n"); + } + + if (status & ASRSTR_AODO(index)) { + asrc_priv->pair[index]->error |= ASRC_OUTPUT_BUFFER_OVERFLOW; + pair_dbg("Output Data Buffer has overflowed\n"); + } + + if (status & ASRSTR_AIDU(index)) { + asrc_priv->pair[index]->error |= ASRC_INPUT_BUFFER_UNDERRUN; + pair_dbg("Input Data Buffer has underflowed\n"); + } + } + + return IRQ_HANDLED; +} + +static int fsl_asrc_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct fsl_asrc *asrc_priv; + struct resource *res; + void __iomem *regs; + int irq, ret, i; + char tmp[16]; + + asrc_priv = devm_kzalloc(&pdev->dev, sizeof(*asrc_priv), GFP_KERNEL); + if (!asrc_priv) + return -ENOMEM; + + asrc_priv->pdev = pdev; + strcpy(asrc_priv->name, np->name); + + /* Get the addresses and IRQ */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + asrc_priv->paddr = res->start; + + /* Register regmap and let it prepare core clock */ + if (of_property_read_bool(np, "big-endian")) + fsl_asrc_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; + + asrc_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "mem", regs, + &fsl_asrc_regmap_config); + if (IS_ERR(asrc_priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap\n"); + return PTR_ERR(asrc_priv->regmap); + } + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, fsl_asrc_isr, 0, + asrc_priv->name, asrc_priv); + if (ret) { + dev_err(&pdev->dev, "failed to claim irq %u: %d\n", irq, ret); + return ret; + } + + asrc_priv->mem_clk = devm_clk_get(&pdev->dev, "mem"); + if (IS_ERR(asrc_priv->mem_clk)) { + dev_err(&pdev->dev, "failed to get mem clock\n"); + return PTR_ERR(asrc_priv->mem_clk); + } + + asrc_priv->ipg_clk = devm_clk_get(&pdev->dev, "ipg"); + if (IS_ERR(asrc_priv->ipg_clk)) { + dev_err(&pdev->dev, "failed to get ipg clock\n"); + return PTR_ERR(asrc_priv->ipg_clk); + } + + for (i = 0; i < ASRC_CLK_MAX_NUM; i++) { + sprintf(tmp, "asrck_%x", i); + asrc_priv->asrck_clk[i] = devm_clk_get(&pdev->dev, tmp); + if (IS_ERR(asrc_priv->asrck_clk[i])) { + dev_err(&pdev->dev, "failed to get %s clock\n", tmp); + return PTR_ERR(asrc_priv->asrck_clk[i]); + } + } + + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx35-asrc")) { + asrc_priv->channel_bits = 3; + clk_map[IN] = input_clk_map_imx35; + clk_map[OUT] = output_clk_map_imx35; + } else { + asrc_priv->channel_bits = 4; + clk_map[IN] = input_clk_map_imx53; + clk_map[OUT] = output_clk_map_imx53; + } + + ret = fsl_asrc_init(asrc_priv); + if (ret) { + dev_err(&pdev->dev, "failed to init asrc %d\n", ret); + return -EINVAL; + } + + asrc_priv->channel_avail = 10; + + ret = of_property_read_u32(np, "fsl,asrc-rate", + &asrc_priv->asrc_rate); + if (ret) { + dev_err(&pdev->dev, "failed to get output rate\n"); + return -EINVAL; + } + + ret = of_property_read_u32(np, "fsl,asrc-width", + &asrc_priv->asrc_width); + if (ret) { + dev_err(&pdev->dev, "failed to get output width\n"); + return -EINVAL; + } + + if (asrc_priv->asrc_width != 16 && asrc_priv->asrc_width != 24) { + dev_warn(&pdev->dev, "unsupported width, switching to 24bit\n"); + asrc_priv->asrc_width = 24; + } + + platform_set_drvdata(pdev, asrc_priv); + pm_runtime_enable(&pdev->dev); + spin_lock_init(&asrc_priv->lock); + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_asrc_component, + &fsl_asrc_dai, 1); + if (ret) { + dev_err(&pdev->dev, "failed to register ASoC DAI\n"); + return ret; + } + + ret = devm_snd_soc_register_platform(&pdev->dev, &fsl_asrc_platform); + if (ret) { + dev_err(&pdev->dev, "failed to register ASoC platform\n"); + return ret; + } + + dev_info(&pdev->dev, "driver registered\n"); + + return 0; +} + +#ifdef CONFIG_PM_RUNTIME +static int fsl_asrc_runtime_resume(struct device *dev) +{ + struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + int i; + + clk_prepare_enable(asrc_priv->mem_clk); + clk_prepare_enable(asrc_priv->ipg_clk); + for (i = 0; i < ASRC_CLK_MAX_NUM; i++) + clk_prepare_enable(asrc_priv->asrck_clk[i]); + + return 0; +} + +static int fsl_asrc_runtime_suspend(struct device *dev) +{ + struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + int i; + + for (i = 0; i < ASRC_CLK_MAX_NUM; i++) + clk_disable_unprepare(asrc_priv->asrck_clk[i]); + clk_disable_unprepare(asrc_priv->ipg_clk); + clk_disable_unprepare(asrc_priv->mem_clk); + + return 0; +} +#endif /* CONFIG_PM_RUNTIME */ + +#ifdef CONFIG_PM_SLEEP +static int fsl_asrc_suspend(struct device *dev) +{ + struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + + regcache_cache_only(asrc_priv->regmap, true); + regcache_mark_dirty(asrc_priv->regmap); + + return 0; +} + +static int fsl_asrc_resume(struct device *dev) +{ + struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + u32 asrctr; + + /* Stop all pairs provisionally */ + regmap_read(asrc_priv->regmap, REG_ASRCTR, &asrctr); + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ASRCEi_ALL_MASK, 0); + + /* Restore all registers */ + regcache_cache_only(asrc_priv->regmap, false); + regcache_sync(asrc_priv->regmap); + + /* Restart enabled pairs */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ASRCEi_ALL_MASK, asrctr); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_asrc_pm = { + SET_RUNTIME_PM_OPS(fsl_asrc_runtime_suspend, fsl_asrc_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(fsl_asrc_suspend, fsl_asrc_resume) +}; + +static const struct of_device_id fsl_asrc_ids[] = { + { .compatible = "fsl,imx35-asrc", }, + { .compatible = "fsl,imx53-asrc", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_asrc_ids); + +static struct platform_driver fsl_asrc_driver = { + .probe = fsl_asrc_probe, + .driver = { + .name = "fsl-asrc", + .of_match_table = fsl_asrc_ids, + .pm = &fsl_asrc_pm, + }, +}; +module_platform_driver(fsl_asrc_driver); + +MODULE_DESCRIPTION("Freescale ASRC ASoC driver"); +MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>"); +MODULE_ALIAS("platform:fsl-asrc"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h new file mode 100644 index 000000000000..a3f211f53c23 --- /dev/null +++ b/sound/soc/fsl/fsl_asrc.h @@ -0,0 +1,461 @@ +/* + * 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 +#define _FSL_ASRC_H + +#define IN 0 +#define OUT 1 + +#define ASRC_DMA_BUFFER_NUM 2 +#define ASRC_INPUTFIFO_THRESHOLD 32 +#define ASRC_OUTPUTFIFO_THRESHOLD 32 +#define ASRC_FIFO_THRESHOLD_MIN 0 +#define ASRC_FIFO_THRESHOLD_MAX 63 +#define ASRC_DMA_BUFFER_SIZE (1024 * 48 * 4) +#define ASRC_MAX_BUFFER_SIZE (1024 * 48) +#define ASRC_OUTPUT_LAST_SAMPLE 8 + +#define IDEAL_RATIO_RATE 1000000 + +#define REG_ASRCTR 0x00 +#define REG_ASRIER 0x04 +#define REG_ASRCNCR 0x0C +#define REG_ASRCFG 0x10 +#define REG_ASRCSR 0x14 + +#define REG_ASRCDR1 0x18 +#define REG_ASRCDR2 0x1C +#define REG_ASRCDR(i) ((i < 2) ? REG_ASRCDR1 : REG_ASRCDR2) + +#define REG_ASRSTR 0x20 +#define REG_ASRRA 0x24 +#define REG_ASRRB 0x28 +#define REG_ASRRC 0x2C +#define REG_ASRPM1 0x40 +#define REG_ASRPM2 0x44 +#define REG_ASRPM3 0x48 +#define REG_ASRPM4 0x4C +#define REG_ASRPM5 0x50 +#define REG_ASRTFR1 0x54 +#define REG_ASRCCR 0x5C + +#define REG_ASRDIA 0x60 +#define REG_ASRDOA 0x64 +#define REG_ASRDIB 0x68 +#define REG_ASRDOB 0x6C +#define REG_ASRDIC 0x70 +#define REG_ASRDOC 0x74 +#define REG_ASRDI(i) (REG_ASRDIA + (i << 3)) +#define REG_ASRDO(i) (REG_ASRDOA + (i << 3)) +#define REG_ASRDx(x, i) (x == IN ? REG_ASRDI(i) : REG_ASRDO(i)) + +#define REG_ASRIDRHA 0x80 +#define REG_ASRIDRLA 0x84 +#define REG_ASRIDRHB 0x88 +#define REG_ASRIDRLB 0x8C +#define REG_ASRIDRHC 0x90 +#define REG_ASRIDRLC 0x94 +#define REG_ASRIDRH(i) (REG_ASRIDRHA + (i << 3)) +#define REG_ASRIDRL(i) (REG_ASRIDRLA + (i << 3)) + +#define REG_ASR76K 0x98 +#define REG_ASR56K 0x9C + +#define REG_ASRMCRA 0xA0 +#define REG_ASRFSTA 0xA4 +#define REG_ASRMCRB 0xA8 +#define REG_ASRFSTB 0xAC +#define REG_ASRMCRC 0xB0 +#define REG_ASRFSTC 0xB4 +#define REG_ASRMCR(i) (REG_ASRMCRA + (i << 3)) +#define REG_ASRFST(i) (REG_ASRFSTA + (i << 3)) + +#define REG_ASRMCR1A 0xC0 +#define REG_ASRMCR1B 0xC4 +#define REG_ASRMCR1C 0xC8 +#define REG_ASRMCR1(i) (REG_ASRMCR1A + (i << 2)) + + +/* REG0 0x00 REG_ASRCTR */ +#define ASRCTR_ATSi_SHIFT(i) (20 + i) +#define ASRCTR_ATSi_MASK(i) (1 << ASRCTR_ATSi_SHIFT(i)) +#define ASRCTR_ATS(i) (1 << ASRCTR_ATSi_SHIFT(i)) +#define ASRCTR_USRi_SHIFT(i) (14 + (i << 1)) +#define ASRCTR_USRi_MASK(i) (1 << ASRCTR_USRi_SHIFT(i)) +#define ASRCTR_USR(i) (1 << ASRCTR_USRi_SHIFT(i)) +#define ASRCTR_IDRi_SHIFT(i) (13 + (i << 1)) +#define ASRCTR_IDRi_MASK(i) (1 << ASRCTR_IDRi_SHIFT(i)) +#define ASRCTR_IDR(i) (1 << ASRCTR_IDRi_SHIFT(i)) +#define ASRCTR_SRST_SHIFT 4 +#define ASRCTR_SRST_MASK (1 << ASRCTR_SRST_SHIFT) +#define ASRCTR_SRST (1 << ASRCTR_SRST_SHIFT) +#define ASRCTR_ASRCEi_SHIFT(i) (1 + i) +#define ASRCTR_ASRCEi_MASK(i) (1 << ASRCTR_ASRCEi_SHIFT(i)) +#define ASRCTR_ASRCE(i) (1 << ASRCTR_ASRCEi_SHIFT(i)) +#define ASRCTR_ASRCEi_ALL_MASK (0x7 << ASRCTR_ASRCEi_SHIFT(0)) +#define ASRCTR_ASRCEN_SHIFT 0 +#define ASRCTR_ASRCEN_MASK (1 << ASRCTR_ASRCEN_SHIFT) +#define ASRCTR_ASRCEN (1 << ASRCTR_ASRCEN_SHIFT) + +/* REG1 0x04 REG_ASRIER */ +#define ASRIER_AFPWE_SHIFT 7 +#define ASRIER_AFPWE_MASK (1 << ASRIER_AFPWE_SHIFT) +#define ASRIER_AFPWE (1 << ASRIER_AFPWE_SHIFT) +#define ASRIER_AOLIE_SHIFT 6 +#define ASRIER_AOLIE_MASK (1 << ASRIER_AOLIE_SHIFT) +#define ASRIER_AOLIE (1 << ASRIER_AOLIE_SHIFT) +#define ASRIER_ADOEi_SHIFT(i) (3 + i) +#define ASRIER_ADOEi_MASK(i) (1 << ASRIER_ADOEi_SHIFT(i)) +#define ASRIER_ADOE(i) (1 << ASRIER_ADOEi_SHIFT(i)) +#define ASRIER_ADIEi_SHIFT(i) (0 + i) +#define ASRIER_ADIEi_MASK(i) (1 << ASRIER_ADIEi_SHIFT(i)) +#define ASRIER_ADIE(i) (1 << ASRIER_ADIEi_SHIFT(i)) + +/* REG2 0x0C REG_ASRCNCR */ +#define ASRCNCR_ANCi_SHIFT(i, b) (b * i) +#define ASRCNCR_ANCi_MASK(i, b) (((1 << b) - 1) << ASRCNCR_ANCi_SHIFT(i, b)) +#define ASRCNCR_ANCi(i, v, b) ((v << ASRCNCR_ANCi_SHIFT(i, b)) & ASRCNCR_ANCi_MASK(i, b)) + +/* REG3 0x10 REG_ASRCFG */ +#define ASRCFG_INIRQi_SHIFT(i) (21 + i) +#define ASRCFG_INIRQi_MASK(i) (1 << ASRCFG_INIRQi_SHIFT(i)) +#define ASRCFG_INIRQi (1 << ASRCFG_INIRQi_SHIFT(i)) +#define ASRCFG_NDPRi_SHIFT(i) (18 + i) +#define ASRCFG_NDPRi_MASK(i) (1 << ASRCFG_NDPRi_SHIFT(i)) +#define ASRCFG_NDPRi (1 << ASRCFG_NDPRi_SHIFT(i)) +#define ASRCFG_POSTMODi_SHIFT(i) (8 + (i << 2)) +#define ASRCFG_POSTMODi_WIDTH 2 +#define ASRCFG_POSTMODi_MASK(i) (((1 << ASRCFG_POSTMODi_WIDTH) - 1) << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_POSTMOD(i, v) ((v) << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_POSTMODi_UP(i) (0 << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_POSTMODi_DCON(i) (1 << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_POSTMODi_DOWN(i) (2 << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_PREMODi_SHIFT(i) (6 + (i << 2)) +#define ASRCFG_PREMODi_WIDTH 2 +#define ASRCFG_PREMODi_MASK(i) (((1 << ASRCFG_PREMODi_WIDTH) - 1) << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMOD(i, v) ((v) << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMODi_UP(i) (0 << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMODi_DCON(i) (1 << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMODi_DOWN(i) (2 << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMODi_BYPASS(i) (3 << ASRCFG_PREMODi_SHIFT(i)) + +/* REG4 0x14 REG_ASRCSR */ +#define ASRCSR_AxCSi_WIDTH 4 +#define ASRCSR_AxCSi_MASK ((1 << ASRCSR_AxCSi_WIDTH) - 1) +#define ASRCSR_AOCSi_SHIFT(i) (12 + (i << 2)) +#define ASRCSR_AOCSi_MASK(i) (((1 << ASRCSR_AxCSi_WIDTH) - 1) << ASRCSR_AOCSi_SHIFT(i)) +#define ASRCSR_AOCS(i, v) ((v) << ASRCSR_AOCSi_SHIFT(i)) +#define ASRCSR_AICSi_SHIFT(i) (i << 2) +#define ASRCSR_AICSi_MASK(i) (((1 << ASRCSR_AxCSi_WIDTH) - 1) << ASRCSR_AICSi_SHIFT(i)) +#define ASRCSR_AICS(i, v) ((v) << ASRCSR_AICSi_SHIFT(i)) + +/* REG5&6 0x18 & 0x1C REG_ASRCDR1 & ASRCDR2 */ +#define ASRCDRi_AxCPi_WIDTH 3 +#define ASRCDRi_AICPi_SHIFT(i) (0 + (i % 2) * 6) +#define ASRCDRi_AICPi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AICPi_SHIFT(i)) +#define ASRCDRi_AICP(i, v) ((v) << ASRCDRi_AICPi_SHIFT(i)) +#define ASRCDRi_AICDi_SHIFT(i) (3 + (i % 2) * 6) +#define ASRCDRi_AICDi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AICDi_SHIFT(i)) +#define ASRCDRi_AICD(i, v) ((v) << ASRCDRi_AICDi_SHIFT(i)) +#define ASRCDRi_AOCPi_SHIFT(i) ((i < 2) ? 12 + i * 6 : 6) +#define ASRCDRi_AOCPi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AOCPi_SHIFT(i)) +#define ASRCDRi_AOCP(i, v) ((v) << ASRCDRi_AOCPi_SHIFT(i)) +#define ASRCDRi_AOCDi_SHIFT(i) ((i < 2) ? 15 + i * 6 : 9) +#define ASRCDRi_AOCDi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AOCDi_SHIFT(i)) +#define ASRCDRi_AOCD(i, v) ((v) << ASRCDRi_AOCDi_SHIFT(i)) + +/* REG7 0x20 REG_ASRSTR */ +#define ASRSTR_DSLCNT_SHIFT 21 +#define ASRSTR_DSLCNT_MASK (1 << ASRSTR_DSLCNT_SHIFT) +#define ASRSTR_DSLCNT (1 << ASRSTR_DSLCNT_SHIFT) +#define ASRSTR_ATQOL_SHIFT 20 +#define ASRSTR_ATQOL_MASK (1 << ASRSTR_ATQOL_SHIFT) +#define ASRSTR_ATQOL (1 << ASRSTR_ATQOL_SHIFT) +#define ASRSTR_AOOLi_SHIFT(i) (17 + i) +#define ASRSTR_AOOLi_MASK(i) (1 << ASRSTR_AOOLi_SHIFT(i)) +#define ASRSTR_AOOL(i) (1 << ASRSTR_AOOLi_SHIFT(i)) +#define ASRSTR_AIOLi_SHIFT(i) (14 + i) +#define ASRSTR_AIOLi_MASK(i) (1 << ASRSTR_AIOLi_SHIFT(i)) +#define ASRSTR_AIOL(i) (1 << ASRSTR_AIOLi_SHIFT(i)) +#define ASRSTR_AODOi_SHIFT(i) (11 + i) +#define ASRSTR_AODOi_MASK(i) (1 << ASRSTR_AODOi_SHIFT(i)) +#define ASRSTR_AODO(i) (1 << ASRSTR_AODOi_SHIFT(i)) +#define ASRSTR_AIDUi_SHIFT(i) (8 + i) +#define ASRSTR_AIDUi_MASK(i) (1 << ASRSTR_AIDUi_SHIFT(i)) +#define ASRSTR_AIDU(i) (1 << ASRSTR_AIDUi_SHIFT(i)) +#define ASRSTR_FPWT_SHIFT 7 +#define ASRSTR_FPWT_MASK (1 << ASRSTR_FPWT_SHIFT) +#define ASRSTR_FPWT (1 << ASRSTR_FPWT_SHIFT) +#define ASRSTR_AOLE_SHIFT 6 +#define ASRSTR_AOLE_MASK (1 << ASRSTR_AOLE_SHIFT) +#define ASRSTR_AOLE (1 << ASRSTR_AOLE_SHIFT) +#define ASRSTR_AODEi_SHIFT(i) (3 + i) +#define ASRSTR_AODFi_MASK(i) (1 << ASRSTR_AODEi_SHIFT(i)) +#define ASRSTR_AODF(i) (1 << ASRSTR_AODEi_SHIFT(i)) +#define ASRSTR_AIDEi_SHIFT(i) (0 + i) +#define ASRSTR_AIDEi_MASK(i) (1 << ASRSTR_AIDEi_SHIFT(i)) +#define ASRSTR_AIDE(i) (1 << ASRSTR_AIDEi_SHIFT(i)) + +/* REG10 0x54 REG_ASRTFR1 */ +#define ASRTFR1_TF_BASE_WIDTH 7 +#define ASRTFR1_TF_BASE_SHIFT 6 +#define ASRTFR1_TF_BASE_MASK (((1 << ASRTFR1_TF_BASE_WIDTH) - 1) << ASRTFR1_TF_BASE_SHIFT) +#define ASRTFR1_TF_BASE(i) ((i) << ASRTFR1_TF_BASE_SHIFT) + +/* + * REG22 0xA0 REG_ASRMCRA + * REG24 0xA8 REG_ASRMCRB + * REG26 0xB0 REG_ASRMCRC + */ +#define ASRMCRi_ZEROBUFi_SHIFT 23 +#define ASRMCRi_ZEROBUFi_MASK (1 << ASRMCRi_ZEROBUFi_SHIFT) +#define ASRMCRi_ZEROBUFi (1 << ASRMCRi_ZEROBUFi_SHIFT) +#define ASRMCRi_EXTTHRSHi_SHIFT 22 +#define ASRMCRi_EXTTHRSHi_MASK (1 << ASRMCRi_EXTTHRSHi_SHIFT) +#define ASRMCRi_EXTTHRSHi (1 << ASRMCRi_EXTTHRSHi_SHIFT) +#define ASRMCRi_BUFSTALLi_SHIFT 21 +#define ASRMCRi_BUFSTALLi_MASK (1 << ASRMCRi_BUFSTALLi_SHIFT) +#define ASRMCRi_BUFSTALLi (1 << ASRMCRi_BUFSTALLi_SHIFT) +#define ASRMCRi_BYPASSPOLYi_SHIFT 20 +#define ASRMCRi_BYPASSPOLYi_MASK (1 << ASRMCRi_BYPASSPOLYi_SHIFT) +#define ASRMCRi_BYPASSPOLYi (1 << ASRMCRi_BYPASSPOLYi_SHIFT) +#define ASRMCRi_OUTFIFO_THRESHOLD_WIDTH 6 +#define ASRMCRi_OUTFIFO_THRESHOLD_SHIFT 12 +#define ASRMCRi_OUTFIFO_THRESHOLD_MASK (((1 << ASRMCRi_OUTFIFO_THRESHOLD_WIDTH) - 1) << ASRMCRi_OUTFIFO_THRESHOLD_SHIFT) +#define ASRMCRi_OUTFIFO_THRESHOLD(v) (((v) << ASRMCRi_OUTFIFO_THRESHOLD_SHIFT) & ASRMCRi_OUTFIFO_THRESHOLD_MASK) +#define ASRMCRi_RSYNIFi_SHIFT 11 +#define ASRMCRi_RSYNIFi_MASK (1 << ASRMCRi_RSYNIFi_SHIFT) +#define ASRMCRi_RSYNIFi (1 << ASRMCRi_RSYNIFi_SHIFT) +#define ASRMCRi_RSYNOFi_SHIFT 10 +#define ASRMCRi_RSYNOFi_MASK (1 << ASRMCRi_RSYNOFi_SHIFT) +#define ASRMCRi_RSYNOFi (1 << ASRMCRi_RSYNOFi_SHIFT) +#define ASRMCRi_INFIFO_THRESHOLD_WIDTH 6 +#define ASRMCRi_INFIFO_THRESHOLD_SHIFT 0 +#define ASRMCRi_INFIFO_THRESHOLD_MASK (((1 << ASRMCRi_INFIFO_THRESHOLD_WIDTH) - 1) << ASRMCRi_INFIFO_THRESHOLD_SHIFT) +#define ASRMCRi_INFIFO_THRESHOLD(v) (((v) << ASRMCRi_INFIFO_THRESHOLD_SHIFT) & ASRMCRi_INFIFO_THRESHOLD_MASK) + +/* + * REG23 0xA4 REG_ASRFSTA + * REG25 0xAC REG_ASRFSTB + * REG27 0xB4 REG_ASRFSTC + */ +#define ASRFSTi_OAFi_SHIFT 23 +#define ASRFSTi_OAFi_MASK (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_OAFi (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_OUTPUT_FIFO_WIDTH 7 +#define ASRFSTi_OUTPUT_FIFO_SHIFT 12 +#define ASRFSTi_OUTPUT_FIFO_MASK (((1 << ASRFSTi_OUTPUT_FIFO_WIDTH) - 1) << ASRFSTi_OUTPUT_FIFO_SHIFT) +#define ASRFSTi_IAEi_SHIFT 11 +#define ASRFSTi_IAEi_MASK (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_IAEi (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_INPUT_FIFO_WIDTH 7 +#define ASRFSTi_INPUT_FIFO_SHIFT 0 +#define ASRFSTi_INPUT_FIFO_MASK ((1 << ASRFSTi_INPUT_FIFO_WIDTH) - 1) + +/* REG28 0xC0 & 0xC4 & 0xC8 REG_ASRMCR1i */ +#define ASRMCR1i_IWD_WIDTH 3 +#define ASRMCR1i_IWD_SHIFT 9 +#define ASRMCR1i_IWD_MASK (((1 << ASRMCR1i_IWD_WIDTH) - 1) << ASRMCR1i_IWD_SHIFT) +#define ASRMCR1i_IWD(v) ((v) << ASRMCR1i_IWD_SHIFT) +#define ASRMCR1i_IMSB_SHIFT 8 +#define ASRMCR1i_IMSB_MASK (1 << ASRMCR1i_IMSB_SHIFT) +#define ASRMCR1i_IMSB_MSB (1 << ASRMCR1i_IMSB_SHIFT) +#define ASRMCR1i_IMSB_LSB (0 << ASRMCR1i_IMSB_SHIFT) +#define ASRMCR1i_OMSB_SHIFT 2 +#define ASRMCR1i_OMSB_MASK (1 << ASRMCR1i_OMSB_SHIFT) +#define ASRMCR1i_OMSB_MSB (1 << ASRMCR1i_OMSB_SHIFT) +#define ASRMCR1i_OMSB_LSB (0 << ASRMCR1i_OMSB_SHIFT) +#define ASRMCR1i_OSGN_SHIFT 1 +#define ASRMCR1i_OSGN_MASK (1 << ASRMCR1i_OSGN_SHIFT) +#define ASRMCR1i_OSGN (1 << ASRMCR1i_OSGN_SHIFT) +#define ASRMCR1i_OW16_SHIFT 0 +#define ASRMCR1i_OW16_MASK (1 << ASRMCR1i_OW16_SHIFT) +#define ASRMCR1i_OW16(v) ((v) << ASRMCR1i_OW16_SHIFT) + + +enum asrc_pair_index { + ASRC_INVALID_PAIR = -1, + ASRC_PAIR_A = 0, + ASRC_PAIR_B = 1, + ASRC_PAIR_C = 2, +}; + +#define ASRC_PAIR_MAX_NUM (ASRC_PAIR_C + 1) + +enum asrc_inclk { + INCLK_NONE = 0x03, + INCLK_ESAI_RX = 0x00, + INCLK_SSI1_RX = 0x01, + INCLK_SSI2_RX = 0x02, + INCLK_SSI3_RX = 0x07, + INCLK_SPDIF_RX = 0x04, + INCLK_MLB_CLK = 0x05, + INCLK_PAD = 0x06, + INCLK_ESAI_TX = 0x08, + INCLK_SSI1_TX = 0x09, + INCLK_SSI2_TX = 0x0a, + INCLK_SSI3_TX = 0x0b, + INCLK_SPDIF_TX = 0x0c, + INCLK_ASRCK1_CLK = 0x0f, +}; + +enum asrc_outclk { + OUTCLK_NONE = 0x03, + OUTCLK_ESAI_TX = 0x00, + OUTCLK_SSI1_TX = 0x01, + OUTCLK_SSI2_TX = 0x02, + OUTCLK_SSI3_TX = 0x07, + OUTCLK_SPDIF_TX = 0x04, + OUTCLK_MLB_CLK = 0x05, + OUTCLK_PAD = 0x06, + OUTCLK_ESAI_RX = 0x08, + OUTCLK_SSI1_RX = 0x09, + OUTCLK_SSI2_RX = 0x0a, + OUTCLK_SSI3_RX = 0x0b, + OUTCLK_SPDIF_RX = 0x0c, + OUTCLK_ASRCK1_CLK = 0x0f, +}; + +#define ASRC_CLK_MAX_NUM 16 + +enum asrc_word_width { + ASRC_WIDTH_24_BIT = 0, + ASRC_WIDTH_16_BIT = 1, + ASRC_WIDTH_8_BIT = 2, +}; + +struct asrc_config { + enum asrc_pair_index pair; + unsigned int channel_num; + unsigned int buffer_num; + unsigned int dma_buffer_size; + unsigned int input_sample_rate; + unsigned int output_sample_rate; + enum asrc_word_width input_word_width; + enum asrc_word_width output_word_width; + enum asrc_inclk inclk; + enum asrc_outclk outclk; +}; + +struct asrc_req { + unsigned int chn_num; + enum asrc_pair_index index; +}; + +struct asrc_querybuf { + unsigned int buffer_index; + unsigned int input_length; + unsigned int output_length; + unsigned long input_offset; + unsigned long output_offset; +}; + +struct asrc_convert_buffer { + void *input_buffer_vaddr; + void *output_buffer_vaddr; + unsigned int input_buffer_length; + unsigned int output_buffer_length; +}; + +struct asrc_status_flags { + enum asrc_pair_index index; + unsigned int overload_error; +}; + +enum asrc_error_status { + ASRC_TASK_Q_OVERLOAD = 0x01, + ASRC_OUTPUT_TASK_OVERLOAD = 0x02, + ASRC_INPUT_TASK_OVERLOAD = 0x04, + ASRC_OUTPUT_BUFFER_OVERFLOW = 0x08, + ASRC_INPUT_BUFFER_UNDERRUN = 0x10, +}; + +struct dma_block { + dma_addr_t dma_paddr; + void *dma_vaddr; + unsigned int length; +}; + +/** + * fsl_asrc_pair: ASRC Pair private data + * + * @asrc_priv: pointer to its parent module + * @config: configuration profile + * @error: error record + * @index: pair index (ASRC_PAIR_A, ASRC_PAIR_B, ASRC_PAIR_C) + * @channels: occupied channel number + * @desc: input and output dma descriptors + * @dma_chan: inputer and output DMA channels + * @dma_data: private dma data + * @pos: hardware pointer position + * @private: pair private area + */ +struct fsl_asrc_pair { + struct fsl_asrc *asrc_priv; + struct asrc_config *config; + unsigned int error; + + enum asrc_pair_index index; + unsigned int channels; + + struct dma_async_tx_descriptor *desc[2]; + struct dma_chan *dma_chan[2]; + struct imx_dma_data dma_data; + unsigned int pos; + + void *private; +}; + +/** + * fsl_asrc_pair: ASRC private data + * + * @dma_params_rx: DMA parameters for receive channel + * @dma_params_tx: DMA parameters for transmit channel + * @pdev: platform device pointer + * @regmap: regmap handler + * @paddr: physical address to the base address of registers + * @mem_clk: clock source to access register + * @ipg_clk: clock source to drive peripheral + * @asrck_clk: clock sources to driver ASRC internal logic + * @lock: spin lock for resource protection + * @pair: pair pointers + * @channel_bits: width of ASRCNCR register for each pair + * @channel_avail: non-occupied channel numbers + * @asrc_rate: default sample rate for ASoC Back-Ends + * @asrc_width: default sample width for ASoC Back-Ends + * @name: driver name + */ +struct fsl_asrc { + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct platform_device *pdev; + struct regmap *regmap; + unsigned long paddr; + struct clk *mem_clk; + struct clk *ipg_clk; + struct clk *asrck_clk[ASRC_CLK_MAX_NUM]; + spinlock_t lock; + + struct fsl_asrc_pair *pair[ASRC_PAIR_MAX_NUM]; + unsigned int channel_bits; + unsigned int channel_avail; + + int asrc_rate; + int asrc_width; + + char name[32]; +}; + +extern struct snd_soc_platform_driver fsl_asrc_platform; +struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir); +#endif /* _FSL_ASRC_H */ diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c new file mode 100644 index 000000000000..ffc000bc1f15 --- /dev/null +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -0,0 +1,391 @@ +/* + * 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. + */ + +#include <linux/dma-mapping.h> +#include <linux/module.h> +#include <linux/platform_data/dma-imx.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> + +#include "fsl_asrc.h" + +#define FSL_ASRC_DMABUF_SIZE (256 * 1024) + +static struct snd_pcm_hardware snd_imx_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .buffer_bytes_max = FSL_ASRC_DMABUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = 65535, /* Limited by SDMA engine */ + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +static bool filter(struct dma_chan *chan, void *param) +{ + if (!imx_dma_is_general_purpose(chan)) + return false; + + chan->private = param; + + return true; +} + +static void fsl_asrc_dma_complete(void *arg) +{ + struct snd_pcm_substream *substream = arg; + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + + pair->pos += snd_pcm_lib_period_bytes(substream); + if (pair->pos >= snd_pcm_lib_buffer_bytes(substream)) + pair->pos = 0; + + snd_pcm_period_elapsed(substream); +} + +static int fsl_asrc_dma_prepare_and_submit(struct snd_pcm_substream *substream) +{ + u8 dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? OUT : IN; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + struct device *dev = rtd->platform->dev; + unsigned long flags = DMA_CTRL_ACK; + + /* Prepare and submit Front-End DMA channel */ + if (!substream->runtime->no_period_wakeup) + flags |= DMA_PREP_INTERRUPT; + + pair->pos = 0; + pair->desc[!dir] = dmaengine_prep_dma_cyclic( + pair->dma_chan[!dir], runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), + dir == OUT ? DMA_TO_DEVICE : DMA_FROM_DEVICE, flags); + if (!pair->desc[!dir]) { + dev_err(dev, "failed to prepare slave DMA for Front-End\n"); + return -ENOMEM; + } + + pair->desc[!dir]->callback = fsl_asrc_dma_complete; + pair->desc[!dir]->callback_param = substream; + + dmaengine_submit(pair->desc[!dir]); + + /* Prepare and submit Back-End DMA channel */ + pair->desc[dir] = dmaengine_prep_dma_cyclic( + pair->dma_chan[dir], 0xffff, 64, 64, DMA_DEV_TO_DEV, 0); + if (!pair->desc[dir]) { + dev_err(dev, "failed to prepare slave DMA for Back-End\n"); + return -ENOMEM; + } + + dmaengine_submit(pair->desc[dir]); + + return 0; +} + +static int fsl_asrc_dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = fsl_asrc_dma_prepare_and_submit(substream); + if (ret) + return ret; + dma_async_issue_pending(pair->dma_chan[IN]); + dma_async_issue_pending(pair->dma_chan[OUT]); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dmaengine_terminate_all(pair->dma_chan[OUT]); + dmaengine_terminate_all(pair->dma_chan[IN]); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_dmaengine_dai_dma_data *dma_params_fe = NULL; + struct snd_dmaengine_dai_dma_data *dma_params_be = NULL; + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + struct fsl_asrc *asrc_priv = pair->asrc_priv; + struct dma_slave_config config_fe, config_be; + enum asrc_pair_index index = pair->index; + struct device *dev = rtd->platform->dev; + int stream = substream->stream; + struct imx_dma_data *tmp_data; + struct snd_soc_dpcm *dpcm; + struct dma_chan *tmp_chan; + struct device *dev_be; + u8 dir = tx ? OUT : IN; + dma_cap_mask_t mask; + int ret; + + /* Fetch the Back-End dma_data from DPCM */ + list_for_each_entry(dpcm, &rtd->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *substream_be; + struct snd_soc_dai *dai = be->cpu_dai; + + if (dpcm->fe != rtd) + continue; + + substream_be = snd_soc_dpcm_get_substream(be, stream); + dma_params_be = snd_soc_dai_get_dma_data(dai, substream_be); + dev_be = dai->dev; + break; + } + + if (!dma_params_be) { + dev_err(dev, "failed to get the substream of Back-End\n"); + return -EINVAL; + } + + /* Override dma_data of the Front-End and config its dmaengine */ + dma_params_fe = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_params_fe->addr = asrc_priv->paddr + REG_ASRDx(!dir, index); + dma_params_fe->maxburst = dma_params_be->maxburst; + + pair->dma_chan[!dir] = fsl_asrc_get_dma_channel(pair, !dir); + if (!pair->dma_chan[!dir]) { + dev_err(dev, "failed to request DMA channel\n"); + return -EINVAL; + } + + memset(&config_fe, 0, sizeof(config_fe)); + ret = snd_dmaengine_pcm_prepare_slave_config(substream, params, &config_fe); + if (ret) { + dev_err(dev, "failed to prepare DMA config for Front-End\n"); + return ret; + } + + ret = dmaengine_slave_config(pair->dma_chan[!dir], &config_fe); + if (ret) { + dev_err(dev, "failed to config DMA channel for Front-End\n"); + return ret; + } + + /* Request and config DMA channel for Back-End */ + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + dma_cap_set(DMA_CYCLIC, mask); + + /* Get DMA request of Back-End */ + tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx"); + tmp_data = tmp_chan->private; + pair->dma_data.dma_request = tmp_data->dma_request; + dma_release_channel(tmp_chan); + + /* Get DMA request of Front-End */ + tmp_chan = fsl_asrc_get_dma_channel(pair, dir); + tmp_data = tmp_chan->private; + pair->dma_data.dma_request2 = tmp_data->dma_request; + pair->dma_data.peripheral_type = tmp_data->peripheral_type; + pair->dma_data.priority = tmp_data->priority; + dma_release_channel(tmp_chan); + + pair->dma_chan[dir] = dma_request_channel(mask, filter, &pair->dma_data); + if (!pair->dma_chan[dir]) { + dev_err(dev, "failed to request DMA channel for Back-End\n"); + return -EINVAL; + } + + if (asrc_priv->asrc_width == 16) + buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + else + buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; + + config_be.direction = DMA_DEV_TO_DEV; + config_be.src_addr_width = buswidth; + config_be.src_maxburst = dma_params_be->maxburst; + config_be.dst_addr_width = buswidth; + config_be.dst_maxburst = dma_params_be->maxburst; + + if (tx) { + config_be.src_addr = asrc_priv->paddr + REG_ASRDO(index); + config_be.dst_addr = dma_params_be->addr; + } else { + config_be.dst_addr = asrc_priv->paddr + REG_ASRDI(index); + config_be.src_addr = dma_params_be->addr; + } + + ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); + if (ret) { + dev_err(dev, "failed to config DMA channel for Back-End\n"); + return ret; + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int fsl_asrc_dma_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + + snd_pcm_set_runtime_buffer(substream, NULL); + + if (pair->dma_chan[IN]) + dma_release_channel(pair->dma_chan[IN]); + + if (pair->dma_chan[OUT]) + dma_release_channel(pair->dma_chan[OUT]); + + pair->dma_chan[IN] = NULL; + pair->dma_chan[OUT] = NULL; + + return 0; +} + +static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = rtd->platform->dev; + struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + struct fsl_asrc_pair *pair; + + pair = kzalloc(sizeof(struct fsl_asrc_pair), GFP_KERNEL); + if (!pair) { + dev_err(dev, "failed to allocate pair\n"); + return -ENOMEM; + } + + pair->asrc_priv = asrc_priv; + + runtime->private_data = pair; + + snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + + return 0; +} + +static int fsl_asrc_dma_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + struct fsl_asrc *asrc_priv; + + if (!pair) + return 0; + + asrc_priv = pair->asrc_priv; + + if (asrc_priv->pair[pair->index] == pair) + asrc_priv->pair[pair->index] = NULL; + + kfree(pair); + + return 0; +} + +static snd_pcm_uframes_t fsl_asrc_dma_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + + return bytes_to_frames(substream->runtime, pair->pos); +} + +static struct snd_pcm_ops fsl_asrc_dma_pcm_ops = { + .ioctl = snd_pcm_lib_ioctl, + .hw_params = fsl_asrc_dma_hw_params, + .hw_free = fsl_asrc_dma_hw_free, + .trigger = fsl_asrc_dma_trigger, + .open = fsl_asrc_dma_startup, + .close = fsl_asrc_dma_shutdown, + .pointer = fsl_asrc_dma_pcm_pointer, +}; + +static int fsl_asrc_dma_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm_substream *substream; + struct snd_pcm *pcm = rtd->pcm; + int ret, i; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) { + dev_err(card->dev, "failed to set DMA mask\n"); + return ret; + } + + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { + substream = pcm->streams[i].substream; + if (!substream) + continue; + + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, + FSL_ASRC_DMABUF_SIZE, &substream->dma_buffer); + if (ret) { + dev_err(card->dev, "failed to allocate DMA buffer\n"); + goto err; + } + } + + return 0; + +err: + if (--i == 0 && pcm->streams[i].substream) + snd_dma_free_pages(&pcm->streams[i].substream->dma_buffer); + + return ret; +} + +static void fsl_asrc_dma_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + int i; + + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { + substream = pcm->streams[i].substream; + if (!substream) + continue; + + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; + substream->dma_buffer.addr = 0; + } +} + +struct snd_soc_platform_driver fsl_asrc_platform = { + .ops = &fsl_asrc_dma_pcm_ops, + .pcm_new = fsl_asrc_dma_pcm_new, + .pcm_free = fsl_asrc_dma_pcm_free, +}; +EXPORT_SYMBOL_GPL(fsl_asrc_platform); diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index d719caf26dc2..72d154e7dd03 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -624,12 +624,14 @@ static int fsl_esai_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver fsl_esai_dai = { .probe = fsl_esai_dai_probe, .playback = { + .stream_name = "CPU-Playback", .channels_min = 1, .channels_max = 12, .rates = FSL_ESAI_RATES, .formats = FSL_ESAI_FORMATS, }, .capture = { + .stream_name = "CPU-Capture", .channels_min = 1, .channels_max = 8, .rates = FSL_ESAI_RATES, diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index c5a0e8af8226..faa049797897 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -106,7 +106,7 @@ irq_rx: xcsr &= ~FSL_SAI_CSR_xF_MASK; if (flags) - regmap_write(sai->regmap, FSL_SAI_TCSR, flags | xcsr); + regmap_write(sai->regmap, FSL_SAI_RCSR, flags | xcsr); out: if (irq_none) @@ -327,7 +327,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - u32 tcsr, rcsr; + u32 xcsr, count = 100; /* * The transmitter bit clock and frame sync are to be @@ -338,9 +338,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, FSL_SAI_CR2_SYNC); - regmap_read(sai->regmap, FSL_SAI_TCSR, &tcsr); - regmap_read(sai->regmap, FSL_SAI_RCSR, &rcsr); - /* * It is recommended that the transmitter is the last enabled * and the first disabled. @@ -349,17 +346,16 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) { - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, - FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, - FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); - } + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), + FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); + + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, + FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, + FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS); - regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), - FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: @@ -370,11 +366,24 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, FSL_SAI_CSR_xIE_MASK, 0); /* Check if the opposite FRDE is also disabled */ - if (!(tx ? rcsr & FSL_SAI_CSR_FRDE : tcsr & FSL_SAI_CSR_FRDE)) { + regmap_read(sai->regmap, FSL_SAI_xCSR(!tx), &xcsr); + if (!(xcsr & FSL_SAI_CSR_FRDE)) { + /* Disable both directions and reset their FIFOs */ regmap_update_bits(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_TERE, 0); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_TERE, 0); + + /* TERE will remain set till the end of current frame */ + do { + udelay(10); + regmap_read(sai->regmap, FSL_SAI_xCSR(tx), &xcsr); + } while (--count && xcsr & FSL_SAI_CSR_TERE); + + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, + FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, + FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); } break; default: @@ -446,12 +455,14 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) static struct snd_soc_dai_driver fsl_sai_dai = { .probe = fsl_sai_dai_probe, .playback = { + .stream_name = "CPU-Playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = FSL_SAI_FORMATS, }, .capture = { + .stream_name = "CPU-Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index d7a60614dd21..70acfe4a9bd5 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -32,10 +32,13 @@ #define FSL_SPDIF_TXFIFO_WML 0x8 #define FSL_SPDIF_RXFIFO_WML 0x8 -#define INTR_FOR_PLAYBACK (INT_TXFIFO_RESYNC) -#define INTR_FOR_CAPTURE (INT_SYM_ERR | INT_BIT_ERR | INT_URX_FUL | INT_URX_OV|\ - INT_QRX_FUL | INT_QRX_OV | INT_UQ_SYNC | INT_UQ_ERR |\ - INT_RXFIFO_RESYNC | INT_LOSS_LOCK | INT_DPLL_LOCKED) +#define INTR_FOR_PLAYBACK (INT_TXFIFO_RESYNC) +#define INTR_FOR_CAPTURE (INT_SYM_ERR | INT_BIT_ERR | INT_URX_FUL |\ + INT_URX_OV | INT_QRX_FUL | INT_QRX_OV |\ + INT_UQ_SYNC | INT_UQ_ERR | INT_RXFIFO_RESYNC |\ + INT_LOSS_LOCK | INT_DPLL_LOCKED) + +#define SIE_INTR_FOR(tx) (tx ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE) /* Index list for the values that has if (DPLL Locked) condition */ static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb }; @@ -96,7 +99,7 @@ struct fsl_spdif_priv { struct platform_device *pdev; struct regmap *regmap; bool dpll_locked; - u16 txrate[SPDIF_TXRATE_MAX]; + u32 txrate[SPDIF_TXRATE_MAX]; u8 txclk_df[SPDIF_TXRATE_MAX]; u8 sysclk_df[SPDIF_TXRATE_MAX]; u8 txclk_src[SPDIF_TXRATE_MAX]; @@ -137,10 +140,9 @@ static void spdif_irq_sym_error(struct fsl_spdif_priv *spdif_priv) dev_dbg(&pdev->dev, "isr: receiver found illegal symbol\n"); - if (!spdif_priv->dpll_locked) { - /* DPLL unlocked seems no audio stream */ + /* Clear illegal symbol if DPLL unlocked since no audio stream */ + if (!spdif_priv->dpll_locked) regmap_update_bits(regmap, REG_SPDIF_SIE, INT_SYM_ERR, 0); - } } /* U/Q Channel receive register full */ @@ -335,8 +337,8 @@ static void spdif_write_channel_status(struct fsl_spdif_priv *spdif_priv) u32 ch_status; ch_status = (bitrev8(ctrl->ch_status[0]) << 16) | - (bitrev8(ctrl->ch_status[1]) << 8) | - bitrev8(ctrl->ch_status[2]); + (bitrev8(ctrl->ch_status[1]) << 8) | + bitrev8(ctrl->ch_status[2]); regmap_write(regmap, REG_SPDIF_STCSCH, ch_status); dev_dbg(&pdev->dev, "STCSCH: 0x%06x\n", ch_status); @@ -390,6 +392,14 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, rate = SPDIF_TXRATE_48000; csfs = IEC958_AES3_CON_FS_48000; break; + case 96000: + rate = SPDIF_TXRATE_96000; + csfs = IEC958_AES3_CON_FS_96000; + break; + case 192000: + rate = SPDIF_TXRATE_192000; + csfs = IEC958_AES3_CON_FS_192000; + break; default: dev_err(&pdev->dev, "unsupported sample rate %d\n", sample_rate); return -EINVAL; @@ -433,13 +443,12 @@ clk_set_bypass: spdif_set_cstatus(ctrl, IEC958_AES3_CON_FS, csfs); /* select clock source and divisor */ - stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) | STC_TXCLK_DF(txclk_df); - mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK | STC_TXCLK_DF_MASK; + stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) | + STC_TXCLK_DF(txclk_df) | STC_SYSCLK_DF(sysclk_df); + mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK | + STC_TXCLK_DF_MASK | STC_SYSCLK_DF_MASK; regmap_update_bits(regmap, REG_SPDIF_STC, mask, stc); - regmap_update_bits(regmap, REG_SPDIF_STC, - STC_SYSCLK_DF_MASK, STC_SYSCLK_DF(sysclk_df)); - dev_dbg(&pdev->dev, "set sample rate to %dHz for %dHz playback\n", spdif_priv->txrate[rate], sample_rate); @@ -553,7 +562,7 @@ static int fsl_spdif_hw_params(struct snd_pcm_substream *substream, return ret; } spdif_set_cstatus(ctrl, IEC958_AES3_CON_CLOCK, - IEC958_AES3_CON_CLOCK_1000PPM); + IEC958_AES3_CON_CLOCK_1000PPM); spdif_write_channel_status(spdif_priv); } else { /* Setup rx clock source */ @@ -569,9 +578,9 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct regmap *regmap = spdif_priv->regmap; - int is_playack = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - u32 intr = is_playack ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE; - u32 dmaen = is_playack ? SCR_DMA_TX_EN : SCR_DMA_RX_EN;; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 intr = SIE_INTR_FOR(tx); + u32 dmaen = SCR_DMA_xX_EN(tx); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -662,9 +671,8 @@ static int fsl_spdif_capture_get(struct snd_kcontrol *kcontrol, u32 cstatus, val; regmap_read(regmap, REG_SPDIF_SIS, &val); - if (!(val & INT_CNEW)) { + if (!(val & INT_CNEW)) return -EAGAIN; - } regmap_read(regmap, REG_SPDIF_SRCSH, &cstatus); ucontrol->value.iec958.status[0] = (cstatus >> 16) & 0xFF; @@ -693,15 +701,14 @@ static int fsl_spdif_subcode_get(struct snd_kcontrol *kcontrol, struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; unsigned long flags; - int ret = 0; + int ret = -EAGAIN; spin_lock_irqsave(&ctrl->ctl_lock, flags); if (ctrl->ready_buf) { int idx = (ctrl->ready_buf - 1) * SPDIF_UBITS_SIZE; memcpy(&ucontrol->value.iec958.subcode[0], &ctrl->subcode[idx], SPDIF_UBITS_SIZE); - } else { - ret = -EAGAIN; + ret = 0; } spin_unlock_irqrestore(&ctrl->ctl_lock, flags); @@ -726,15 +733,14 @@ static int fsl_spdif_qget(struct snd_kcontrol *kcontrol, struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; unsigned long flags; - int ret = 0; + int ret = -EAGAIN; spin_lock_irqsave(&ctrl->ctl_lock, flags); if (ctrl->ready_buf) { int idx = (ctrl->ready_buf - 1) * SPDIF_QSUB_SIZE; memcpy(&ucontrol->value.bytes.data[0], &ctrl->qsub[idx], SPDIF_QSUB_SIZE); - } else { - ret = -EAGAIN; + ret = 0; } spin_unlock_irqrestore(&ctrl->ctl_lock, flags); @@ -799,10 +805,10 @@ static int spdif_get_rxclk_rate(struct fsl_spdif_priv *spdif_priv, regmap_read(regmap, REG_SPDIF_SRPC, &phaseconf); clksrc = (phaseconf >> SRPC_CLKSRC_SEL_OFFSET) & 0xf; - if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED)) { - /* Get bus clock from system */ + + /* Get bus clock from system */ + if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED)) busclk_freq = clk_get_rate(spdif_priv->sysclk); - } /* FreqMeas_CLK = (BUS_CLK * FreqMeas) / 2 ^ 10 / GAINSEL / 128 */ tmpval64 = (u64) busclk_freq * freqmeas; @@ -826,12 +832,12 @@ static int fsl_spdif_rxrate_get(struct snd_kcontrol *kcontrol, { struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); - int rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL); + int rate = 0; if (spdif_priv->dpll_locked) - ucontrol->value.integer.value[0] = rate; - else - ucontrol->value.integer.value[0] = 0; + rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL); + + ucontrol->value.integer.value[0] = rate; return 0; } @@ -969,12 +975,14 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver fsl_spdif_dai = { .probe = &fsl_spdif_dai_probe, .playback = { + .stream_name = "CPU-Playback", .channels_min = 2, .channels_max = 2, .rates = FSL_SPDIF_RATES_PLAYBACK, .formats = FSL_SPDIF_FORMATS_PLAYBACK, }, .capture = { + .stream_name = "CPU-Capture", .channels_min = 2, .channels_max = 2, .rates = FSL_SPDIF_RATES_CAPTURE, @@ -1046,7 +1054,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, struct clk *clk, u64 savesub, enum spdif_txrate index, bool round) { - const u32 rate[] = { 32000, 44100, 48000 }; + const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; bool is_sysclk = clk == spdif_priv->sysclk; u64 rate_ideal, rate_actual, sub; u32 sysclk_dfmin, sysclk_dfmax; @@ -1105,7 +1113,7 @@ out: static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, enum spdif_txrate index) { - const u32 rate[] = { 32000, 44100, 48000 }; + const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; struct platform_device *pdev = spdif_priv->pdev; struct device *dev = &pdev->dev; u64 savesub = 100000, ret; @@ -1238,12 +1246,12 @@ static int fsl_spdif_probe(struct platform_device *pdev) spin_lock_init(&ctrl->ctl_lock); /* Init tx channel status default value */ - ctrl->ch_status[0] = - IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_5015; + ctrl->ch_status[0] = IEC958_AES0_CON_NOT_COPYRIGHT | + IEC958_AES0_CON_EMPHASIS_5015; ctrl->ch_status[1] = IEC958_AES1_CON_DIGDIGCONV_ID; ctrl->ch_status[2] = 0x00; - ctrl->ch_status[3] = - IEC958_AES3_CON_FS_44100 | IEC958_AES3_CON_CLOCK_1000PPM; + ctrl->ch_status[3] = IEC958_AES3_CON_FS_44100 | + IEC958_AES3_CON_CLOCK_1000PPM; spdif_priv->dpll_locked = false; diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h index 16fde4b927d3..00bd3514c610 100644 --- a/sound/soc/fsl/fsl_spdif.h +++ b/sound/soc/fsl/fsl_spdif.h @@ -93,6 +93,8 @@ #define SCR_USRC_SEL_RECV (0x1 << SCR_USRC_SEL_OFFSET) #define SCR_USRC_SEL_CHIP (0x3 << SCR_USRC_SEL_OFFSET) +#define SCR_DMA_xX_EN(tx) (tx ? SCR_DMA_TX_EN : SCR_DMA_RX_EN) + /* SPDIF CDText control */ #define SRCD_CD_USER_OFFSET 1 #define SRCD_CD_USER (1 << SRCD_CD_USER_OFFSET) @@ -164,8 +166,10 @@ enum spdif_txrate { SPDIF_TXRATE_32000 = 0, SPDIF_TXRATE_44100, SPDIF_TXRATE_48000, + SPDIF_TXRATE_96000, + SPDIF_TXRATE_192000, }; -#define SPDIF_TXRATE_MAX (SPDIF_TXRATE_48000 + 1) +#define SPDIF_TXRATE_MAX (SPDIF_TXRATE_192000 + 1) #define SPDIF_CSTATUS_BYTE 6 @@ -175,7 +179,9 @@ enum spdif_txrate { #define FSL_SPDIF_RATES_PLAYBACK (SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000) + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000) #define FSL_SPDIF_RATES_CAPTURE (SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_32000 | \ diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9bfef55d77d1..87eb5776a39b 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -590,8 +590,8 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, else clkrate = clk_round_rate(ssi_private->baudclk, tmprate); - do_div(clkrate, factor); - afreq = (u32)clkrate / (i + 1); + clkrate /= factor; + afreq = clkrate / (i + 1); if (freq == afreq) sub = 0; @@ -1032,12 +1032,14 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { static struct snd_soc_dai_driver fsl_ssi_dai_template = { .probe = fsl_ssi_dai_probe, .playback = { + .stream_name = "CPU-Playback", .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { + .stream_name = "CPU-Capture", .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 267717aa96c1..46f9beb6b273 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -67,7 +67,7 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, { ssize_t ret; char *buf; - int port = (int)file->private_data; + uintptr_t port = (uintptr_t)file->private_data; u32 pdcr, ptcr; if (audmux_clk) { @@ -147,7 +147,7 @@ static const struct file_operations audmux_debugfs_fops = { static void audmux_debugfs_init(void) { - int i; + uintptr_t i; char buf[20]; audmux_debugfs_root = debugfs_create_dir("audmux", NULL); @@ -157,10 +157,10 @@ static void audmux_debugfs_init(void) } for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) { - snprintf(buf, sizeof(buf), "ssi%d", i); + snprintf(buf, sizeof(buf), "ssi%lu", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) - pr_warning("Failed to create AUDMUX port %d debugfs file\n", + pr_warning("Failed to create AUDMUX port %lu debugfs file\n", i); } } diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 03a7fdcdf114..159e517fa09a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -116,6 +116,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, { struct device_node *node; struct clk *clk; + u32 val; int ret; /* @@ -151,10 +152,8 @@ asoc_simple_card_sub_parse_of(struct device_node *np, } dai->sysclk = clk_get_rate(clk); - } else if (of_property_read_bool(np, "system-clock-frequency")) { - of_property_read_u32(np, - "system-clock-frequency", - &dai->sysclk); + } else if (!of_property_read_u32(np, "system-clock-frequency", &val)) { + dai->sysclk = val; } else { clk = of_clk_get(node, 0); if (!IS_ERR(clk)) @@ -303,6 +302,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, { struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; struct simple_dai_props *dai_props = priv->dai_props; + u32 val; int ret; /* parsing the card name from DT */ @@ -325,8 +325,9 @@ static int asoc_simple_card_parse_of(struct device_node *node, } /* Factor to mclk, used in hw_params() */ - of_property_read_u32(node, "simple-audio-card,mclk-fs", - &priv->mclk_fs); + ret = of_property_read_u32(node, "simple-audio-card,mclk-fs", &val); + if (ret == 0) + priv->mclk_fs = val; dev_dbg(dev, "New simple-card: %s\n", priv->snd_card.name ? priv->snd_card.name : ""); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c30fedb3e149..f5b4a9c79cdf 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -58,3 +58,15 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH help This adds audio driver for Intel Baytrail platform based boards with the MAX98090 audio codec. + +config SND_SOC_INTEL_BROADWELL_MACH + tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC + select SND_SOC_INTEL_HASWELL + select SND_COMPRESS_OFFLOAD + select SND_SOC_RT286 + help + This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell + Ultrabook platforms. + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 4bfca79a42ba..7acbfc43a0c6 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -24,7 +24,9 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o +snd-soc-sst-broadwell-objs := broadwell.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o 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_BROADWELL_MACH) += snd-soc-sst-broadwell.o diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c new file mode 100644 index 000000000000..0e550f14028f --- /dev/null +++ b/sound/soc/intel/broadwell.c @@ -0,0 +1,251 @@ +/* + * Intel Broadwell Wildcatpoint SST Audio + * + * Copyright (C) 2013, 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. + * + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include "sst-dsp.h" +#include "sst-haswell-ipc.h" + +#include "../codecs/rt286.h" + +static const struct snd_soc_dapm_widget broadwell_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC1", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_LINE("Line Jack", NULL), +}; + +static const struct snd_soc_dapm_route broadwell_rt286_map[] = { + + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack deteck */ + {"Headphones", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + {"LINE1", NULL, "Line Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC1"}, + {"DMIC2 Pin", NULL, "DMIC2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static int broadwell_ssp0_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); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + 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); + return 0; +} + +static int broadwell_rt286_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; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops broadwell_rt286_ops = { + .hw_params = broadwell_rt286_hw_params, +}; + +static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); + struct sst_hsw *broadwell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings */ + ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_CLOCK_MASTER, 9); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set device config\n"); + return ret; + } + + /* always connected - check HP for jack detect */ + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Line Jack"); + snd_soc_dapm_enable_pin(dapm, "DMIC1"); + snd_soc_dapm_enable_pin(dapm, "DMIC2"); + + return 0; +} + +/* broadwell digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link broadwell_rt286_dais[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback", + .cpu_dai_name = "System Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = broadwell_rtd_init, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload0", + .stream_name = "Offload0 Playback", + .cpu_dai_name = "Offload0 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload1", + .stream_name = "Offload1 Playback", + .cpu_dai_name = "Offload1 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Loopback PCM", + .stream_name = "Loopback", + .cpu_dai_name = "Loopback Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 0, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + { + .name = "Capture PCM", + .stream_name = "Capture", + .cpu_dai_name = "Capture Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .be_id = 0, + .cpu_dai_name = "snd-soc-dummy-dai", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt286-aif1", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broadwell_ssp0_fixup, + .ops = &broadwell_rt286_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +/* broadwell audio machine driver for WPT + RT286S */ +static struct snd_soc_card broadwell_rt286 = { + .name = "broadwell-rt286", + .owner = THIS_MODULE, + .dai_link = broadwell_rt286_dais, + .num_links = ARRAY_SIZE(broadwell_rt286_dais), + .dapm_widgets = broadwell_widgets, + .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), + .dapm_routes = broadwell_rt286_map, + .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map), + .fully_routed = true, +}; + +static int broadwell_audio_probe(struct platform_device *pdev) +{ + broadwell_rt286.dev = &pdev->dev; + + return snd_soc_register_card(&broadwell_rt286); +} + +static int broadwell_audio_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&broadwell_rt286); + return 0; +} + +static struct platform_driver broadwell_audio = { + .probe = broadwell_audio_probe, + .remove = broadwell_audio_remove, + .driver = { + .name = "broadwell-audio", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(broadwell_audio) + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:broadwell-audio"); diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 5fc98c64a3f4..b8b8af571ef1 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -39,8 +39,7 @@ static const struct snd_soc_dapm_widget byt_max98090_widgets[] = { static const struct snd_soc_dapm_route byt_max98090_audio_map[] = { {"IN34", NULL, "Headset Mic"}, - {"IN34", NULL, "MICBIAS"}, - {"MICBIAS", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "MICBIAS"}, {"DMICL", NULL, "Int Mic"}, {"Headphone", NULL, "HPL"}, {"Headphone", NULL, "HPR"}, @@ -64,14 +63,6 @@ static struct snd_soc_jack_pin hs_jack_pins[] = { .pin = "Headset Mic", .mask = SND_JACK_MICROPHONE, }, - { - .pin = "Ext Spk", - .mask = SND_JACK_LINEOUT, - }, - { - .pin = "Int Mic", - .mask = SND_JACK_LINEIN, - }, }; static struct snd_soc_jack_gpio hs_jack_gpios[] = { @@ -84,7 +75,8 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = { { .name = "mic-gpio", .idx = 1, - .report = SND_JACK_MICROPHONE | SND_JACK_LINEIN, + .invert = 1, + .report = SND_JACK_MICROPHONE, .debounce_time = 200, }, }; @@ -108,7 +100,8 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) } /* Enable jack detection */ - ret = snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, jack); + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_LINEOUT | SND_JACK_HEADSET, jack); if (ret) return ret; @@ -117,13 +110,9 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) if (ret) return ret; - ret = snd_soc_jack_add_gpiods(card->dev->parent, jack, - ARRAY_SIZE(hs_jack_gpios), - hs_jack_gpios); - if (ret) - return ret; - - return max98090_mic_detect(codec, jack); + return snd_soc_jack_add_gpiods(card->dev->parent, jack, + ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); } static struct snd_soc_dai_link byt_max98090_dais[] = { diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 53d160d39972..234a58de3c53 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -34,6 +34,7 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { }; static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { + {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, {"IN2N", NULL, "Headset Mic"}, {"DMIC1", NULL, "Internal Mic"}, diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h new file mode 100644 index 000000000000..14063ab8c7c5 --- /dev/null +++ b/sound/soc/intel/sst-atom-controls.h @@ -0,0 +1,30 @@ +/* + * Copyright (C) 2013-14 Intel Corp + * Author: Ramesh Babu <ramesh.babu.koul@intel.com> + * Omair M Abdullah <omair.m.abdullah@intel.com> + * Samreen Nilofer <samreen.nilofer@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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ + +#ifndef __SST_CONTROLS_V2_H__ +#define __SST_CONTROLS_V2_H__ + +enum { + MERR_DPCM_AUDIO = 0, + MERR_DPCM_COMPR, +}; + + +#endif diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index d207b22ea330..67673a2c0f41 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -122,6 +122,26 @@ struct sst_byt_tstamp { u32 channel_peak[8]; } __packed; +struct sst_byt_fw_version { + u8 build; + u8 minor; + u8 major; + u8 type; +} __packed; + +struct sst_byt_fw_build_info { + u8 date[16]; + u8 time[16]; +} __packed; + +struct sst_byt_fw_init { + struct sst_byt_fw_version fw_version; + struct sst_byt_fw_build_info build_info; + u16 result; + u8 module_id; + u8 debug_info; +} __packed; + /* driver internal IPC message structure */ struct ipc_message { struct list_head list; @@ -868,6 +888,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) { struct sst_byt *byt; struct sst_fw *byt_sst_fw; + struct sst_byt_fw_init init; int err; dev_dbg(dev, "initialising Byt DSP IPC\n"); @@ -929,6 +950,15 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) goto boot_err; } + /* show firmware information */ + sst_dsp_inbox_read(byt->dsp, &init, sizeof(init)); + dev_info(byt->dev, "FW version: %02x.%02x.%02x.%02x\n", + init.fw_version.major, init.fw_version.minor, + init.fw_version.build, init.fw_version.type); + dev_info(byt->dev, "Build type: %x\n", init.fw_version.type); + dev_info(byt->dev, "Build date: %s %s\n", + init.build_info.date, init.build_info.time); + pdata->dsp = byt; byt->fw = byt_sst_fw; diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 8eab97368ea7..599401c0c655 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -32,7 +32,7 @@ static const struct snd_pcm_hardware sst_byt_pcm_hardware = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FORMAT_S24_LE, + SNDRV_PCM_FMTBIT_S24_LE, .period_bytes_min = 384, .period_bytes_max = 48000, .periods_min = 2, diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index 0b715b20a2d7..cd23060a0d86 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -224,19 +224,23 @@ EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64); void sst_dsp_dump(struct sst_dsp *sst) { - sst->ops->dump(sst); + if (sst->ops->dump) + sst->ops->dump(sst); } EXPORT_SYMBOL_GPL(sst_dsp_dump); void sst_dsp_reset(struct sst_dsp *sst) { - sst->ops->reset(sst); + if (sst->ops->reset) + sst->ops->reset(sst); } EXPORT_SYMBOL_GPL(sst_dsp_reset); int sst_dsp_boot(struct sst_dsp *sst) { - sst->ops->boot(sst); + if (sst->ops->boot) + sst->ops->boot(sst); + return 0; } EXPORT_SYMBOL_GPL(sst_dsp_boot); diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index e44423be66c4..3165dfa97408 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -52,7 +52,11 @@ #define SST_CLKCTL 0x78 #define SST_CSR2 0x80 #define SST_LTRC 0xE0 -#define SST_HDMC 0xE8 +#define SST_HMDC 0xE8 + +#define SST_SHIM_BEGIN SST_CSR +#define SST_SHIM_END SST_HDMC + #define SST_DBGO 0xF0 #define SST_SHIM_SIZE 0x100 @@ -73,6 +77,8 @@ #define SST_CSR_S0IOCS (0x1 << 21) #define SST_CSR_S1IOCS (0x1 << 23) #define SST_CSR_LPCS (0x1 << 31) +#define SST_CSR_24MHZ_LPCS (SST_CSR_SBCS0 | SST_CSR_SBCS1 | SST_CSR_LPCS) +#define SST_CSR_24MHZ_NO_LPCS (SST_CSR_SBCS0 | SST_CSR_SBCS1) #define SST_BYT_CSR_RST (0x1 << 0) #define SST_BYT_CSR_VECTOR_SEL (0x1 << 1) #define SST_BYT_CSR_STALL (0x1 << 2) @@ -92,6 +98,14 @@ #define SST_IMRX_DONE (0x1 << 0) #define SST_BYT_IMRX_REQUEST (0x1 << 1) +/* IMRD / IMD */ +#define SST_IMRD_DONE (0x1 << 0) +#define SST_IMRD_BUSY (0x1 << 1) +#define SST_IMRD_SSP0 (0x1 << 16) +#define SST_IMRD_DMAC0 (0x1 << 21) +#define SST_IMRD_DMAC1 (0x1 << 22) +#define SST_IMRD_DMAC (SST_IMRD_DMAC0 | SST_IMRD_DMAC1) + /* IPCX / IPCC */ #define SST_IPCX_DONE (0x1 << 30) #define SST_IPCX_BUSY (0x1 << 31) @@ -118,9 +132,21 @@ /* LTRC */ #define SST_LTRC_VAL(x) (x << 0) -/* HDMC */ -#define SST_HDMC_HDDA0(x) (x << 0) -#define SST_HDMC_HDDA1(x) (x << 7) +/* HMDC */ +#define SST_HMDC_HDDA0(x) (x << 0) +#define SST_HMDC_HDDA1(x) (x << 7) +#define SST_HMDC_HDDA_E0_CH0 1 +#define SST_HMDC_HDDA_E0_CH1 2 +#define SST_HMDC_HDDA_E0_CH2 4 +#define SST_HMDC_HDDA_E0_CH3 8 +#define SST_HMDC_HDDA_E1_CH0 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH0) +#define SST_HMDC_HDDA_E1_CH1 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH1) +#define SST_HMDC_HDDA_E1_CH2 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH2) +#define SST_HMDC_HDDA_E1_CH3 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH3) +#define SST_HMDC_HDDA_E0_ALLCH (SST_HMDC_HDDA_E0_CH0 | SST_HMDC_HDDA_E0_CH1 | \ + SST_HMDC_HDDA_E0_CH2 | SST_HMDC_HDDA_E0_CH3) +#define SST_HMDC_HDDA_E1_ALLCH (SST_HMDC_HDDA_E1_CH0 | SST_HMDC_HDDA_E1_CH1 | \ + SST_HMDC_HDDA_E1_CH2 | SST_HMDC_HDDA_E1_CH3) /* SST Vendor Defined Registers and bits */ @@ -130,11 +156,16 @@ #define SST_VDRTCTL3 0xaC /* VDRTCTL0 */ +#define SST_VDRTCL0_APLLSE_MASK 1 #define SST_VDRTCL0_DSRAMPGE_SHIFT 16 #define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT) #define SST_VDRTCL0_ISRAMPGE_SHIFT 6 #define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT) +/* PMCS */ +#define SST_PMCS 0x84 +#define SST_PMCS_PS_MASK 0x3 + struct sst_dsp; /* diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 535f517629fd..4b6c163c10ff 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -28,9 +28,6 @@ #include <linux/firmware.h> #include <linux/pm_runtime.h> -#include <linux/acpi.h> -#include <acpi/acpi_bus.h> - #include "sst-dsp.h" #include "sst-dsp-priv.h" #include "sst-haswell-ipc.h" @@ -272,9 +269,9 @@ static void hsw_boot(struct sst_dsp *sst) SST_CSR2_SDFD_SSP1); /* enable DMA engine 0,1 all channels to access host memory */ - sst_dsp_shim_update_bits_unlocked(sst, SST_HDMC, - SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff), - SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff)); + sst_dsp_shim_update_bits_unlocked(sst, SST_HMDC, + SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff), + SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff)); /* disable all clock gating */ writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2); @@ -313,9 +310,7 @@ static const struct sst_adsp_memregion lp_region[] = { /* wild cat point ADSP mem regions */ static const struct sst_adsp_memregion wpt_region[] = { - {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */ - {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */ - {0x80000, 0xA0000, 4, SST_MEM_DRAM}, /* D-SRAM2 - 4 * 32kB */ + {0x00000, 0xA0000, 20, SST_MEM_DRAM}, /* D-SRAM0,D-SRAM1,D-SRAM2 - 20 * 32kB */ {0xA0000, 0xF0000, 10, SST_MEM_IRAM}, /* I-SRAM - 10 * 32kB */ }; @@ -339,26 +334,56 @@ static int hsw_acpi_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata) return 0; } +struct sst_sram_shift { + u32 dev_id; /* SST Device IDs */ + u32 iram_shift; + u32 dram_shift; +}; + +static const struct sst_sram_shift sram_shift[] = { + {SST_DEV_ID_LYNX_POINT, 6, 16}, /* lp */ + {SST_DEV_ID_WILDCAT_POINT, 2, 12}, /* wpt */ +}; static u32 hsw_block_get_bit(struct sst_mem_block *block) { - u32 bit = 0, shift = 0; + u32 bit = 0, shift = 0, index; + struct sst_dsp *sst = block->dsp; - switch (block->type) { - case SST_MEM_DRAM: - shift = 16; - break; - case SST_MEM_IRAM: - shift = 6; - break; - default: - return 0; + for (index = 0; index < ARRAY_SIZE(sram_shift); index++) { + if (sram_shift[index].dev_id == sst->id) + break; } + if (index < ARRAY_SIZE(sram_shift)) { + switch (block->type) { + case SST_MEM_DRAM: + shift = sram_shift[index].dram_shift; + break; + case SST_MEM_IRAM: + shift = sram_shift[index].iram_shift; + break; + default: + shift = 0; + } + } else + shift = 0; + bit = 1 << (block->index + shift); return bit; } +/*dummy read a SRAM block.*/ +static void sst_mem_block_dummy_read(struct sst_mem_block *block) +{ + u32 size; + u8 tmp_buf[4]; + struct sst_dsp *sst = block->dsp; + + size = block->size > 4 ? 4 : block->size; + memcpy_fromio(tmp_buf, sst->addr.lpe + block->offset, size); +} + /* enable 32kB memory block - locks held by caller */ static int hsw_block_enable(struct sst_mem_block *block) { @@ -378,6 +403,8 @@ static int hsw_block_enable(struct sst_mem_block *block) /* wait 18 DSP clock ticks */ udelay(10); + /*add a dummy read before the SRAM block is written, otherwise the writing may miss bytes sometimes.*/ + sst_mem_block_dummy_read(block); return 0; } @@ -488,8 +515,9 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) } } - /* set default power gating mask */ - writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL0); + /* set default power gating control, enable power gating control for all blocks. that is, + can't be accessed, please enable each block before accessing. */ + writel(0xffffffff, sst->addr.pci_cfg + SST_VDRTCTL0); return 0; } diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 434236343ddf..b6291516dbbf 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -183,7 +183,7 @@ struct sst_hsw_ipc_fw_ready { u32 inbox_size; u32 outbox_size; u32 fw_info_size; - u8 fw_info[1]; + u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)]; } __attribute__((packed)); struct ipc_message { @@ -457,9 +457,10 @@ static void ipc_tx_msgs(struct kthread_work *work) return; } - /* if the DSP is busy we will TX messages after IRQ */ + /* if the DSP is busy, we will TX messages after IRQ. + * also postpone if we are in the middle of procesing completion irq*/ ipcx = sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX); - if (ipcx & SST_IPCX_BUSY) { + if (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)) { spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); return; } @@ -502,6 +503,7 @@ static int tx_wait_done(struct sst_hsw *hsw, struct ipc_message *msg, ipc_shim_dbg(hsw, "message timeout"); trace_ipc_error("error message timeout for", msg->header); + list_del(&msg->list); ret = -ETIMEDOUT; } else { @@ -569,6 +571,9 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) { struct sst_hsw_ipc_fw_ready fw_ready; u32 offset; + u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)]; + char *tmp[5], *pinfo; + int i = 0; offset = (header & 0x1FFFFFFF) << 3; @@ -589,6 +594,19 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) fw_ready.inbox_offset, fw_ready.inbox_size); dev_dbg(hsw->dev, " mailbox downstream 0x%x - size 0x%x\n", fw_ready.outbox_offset, fw_ready.outbox_size); + if (fw_ready.fw_info_size < sizeof(fw_ready.fw_info)) { + fw_ready.fw_info[fw_ready.fw_info_size] = 0; + dev_dbg(hsw->dev, " Firmware info: %s \n", fw_ready.fw_info); + + /* log the FW version info got from the mailbox here. */ + memcpy(fw_info, fw_ready.fw_info, fw_ready.fw_info_size); + pinfo = &fw_info[0]; + for (i = 0; i < sizeof(tmp) / sizeof(char *); i++) + tmp[i] = strsep(&pinfo, " "); + dev_info(hsw->dev, "FW loaded, mailbox readback FW info: type %s, - " + "version: %s.%s, build %s, source commit id: %s\n", + tmp[0], tmp[1], tmp[2], tmp[3], tmp[4]); + } } static void hsw_notification_work(struct work_struct *work) @@ -671,7 +689,9 @@ static void hsw_stream_update(struct sst_hsw *hsw, struct ipc_message *msg) switch (stream_msg) { case IPC_STR_STAGE_MESSAGE: case IPC_STR_NOTIFICATION: + break; case IPC_STR_RESET: + trace_ipc_notification("stream reset", stream->reply.stream_hw_id); break; case IPC_STR_PAUSE: stream->running = false; @@ -762,7 +782,8 @@ static int hsw_process_reply(struct sst_hsw *hsw, u32 header) } /* update any stream states */ - hsw_stream_update(hsw, msg); + if (msg_get_global_type(header) == IPC_GLB_STREAM_MESSAGE) + hsw_stream_update(hsw, msg); /* wake up and return the error if we have waiters on this message ? */ list_del(&msg->list); @@ -1628,7 +1649,7 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw, enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx) { u32 header, state_; - int ret; + int ret, item; header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE); state_ = state; @@ -1642,6 +1663,13 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw, return ret; } + for (item = 0; item < dx->entries_no; item++) { + dev_dbg(hsw->dev, + "Item[%d] offset[%x] - size[%x] - source[%x]\n", + item, dx->mem_info[item].offset, + dx->mem_info[item].size, + dx->mem_info[item].source); + } dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n", dx->entries_no, state); @@ -1775,8 +1803,6 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) /* get the FW version */ sst_hsw_fw_get_version(hsw, &version); - dev_info(hsw->dev, "FW loaded: type %d - version: %d.%d build %d\n", - version.type, version.major, version.minor, version.build); /* get the globalmixer */ ret = sst_hsw_mixer_get_info(hsw); diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 058efb17c568..61bf6da4bb02 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -80,7 +80,7 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE | + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = PAGE_SIZE, .period_bytes_max = (HSW_PCM_PERIODS_MAX / HSW_PCM_PERIODS_MIN) * PAGE_SIZE, @@ -400,7 +400,15 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, sst_hsw_stream_set_valid(hsw, pcm_data->stream, 16); break; case SNDRV_PCM_FORMAT_S24_LE: - bits = SST_HSW_DEPTH_24BIT; + bits = SST_HSW_DEPTH_32BIT; + sst_hsw_stream_set_valid(hsw, pcm_data->stream, 24); + break; + case SNDRV_PCM_FORMAT_S8: + bits = SST_HSW_DEPTH_8BIT; + sst_hsw_stream_set_valid(hsw, pcm_data->stream, 8); + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = SST_HSW_DEPTH_32BIT; sst_hsw_stream_set_valid(hsw, pcm_data->stream, 32); break; default: @@ -685,8 +693,9 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) } #define HSW_FORMATS \ - (SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S32_LE) + (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S8) static struct snd_soc_dai_driver hsw_dais[] = { { @@ -696,7 +705,7 @@ static struct snd_soc_dai_driver hsw_dais[] = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, { @@ -727,8 +736,8 @@ static struct snd_soc_dai_driver hsw_dais[] = { .stream_name = "Loopback Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = HSW_FORMATS, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, { @@ -737,8 +746,8 @@ static struct snd_soc_dai_driver hsw_dais[] = { .stream_name = "Analog Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = HSW_FORMATS, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, }; diff --git a/sound/soc/intel/sst-mfld-dsp.h b/sound/soc/intel/sst-mfld-dsp.h index 8d482d76475a..4257263157cd 100644 --- a/sound/soc/intel/sst-mfld-dsp.h +++ b/sound/soc/intel/sst-mfld-dsp.h @@ -3,7 +3,7 @@ /* * sst_mfld_dsp.h - Intel SST Driver for audio engine * - * Copyright (C) 2008-12 Intel Corporation + * Copyright (C) 2008-14 Intel Corporation * Authors: Vinod Koul <vinod.koul@linux.intel.com> * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ * @@ -19,6 +19,142 @@ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ +#define SST_MAX_BIN_BYTES 1024 + +#define MAX_DBG_RW_BYTES 80 +#define MAX_NUM_SCATTER_BUFFERS 8 +#define MAX_LOOP_BACK_DWORDS 8 +/* IPC base address and mailbox, timestamp offsets */ +#define SST_MAILBOX_SIZE 0x0400 +#define SST_MAILBOX_SEND 0x0000 +#define SST_TIME_STAMP 0x1800 +#define SST_TIME_STAMP_MRFLD 0x800 +#define SST_RESERVED_OFFSET 0x1A00 +#define SST_SCU_LPE_MAILBOX 0x1000 +#define SST_LPE_SCU_MAILBOX 0x1400 +#define SST_SCU_LPE_LOG_BUF (SST_SCU_LPE_MAILBOX+16) +#define PROCESS_MSG 0x80 + +/* Message ID's for IPC messages */ +/* Bits B7: SST or IA/SC ; B6-B4: Msg Category; B3-B0: Msg Type */ + +/* I2L Firmware/Codec Download msgs */ +#define IPC_IA_PREP_LIB_DNLD 0x01 +#define IPC_IA_LIB_DNLD_CMPLT 0x02 +#define IPC_IA_GET_FW_VERSION 0x04 +#define IPC_IA_GET_FW_BUILD_INF 0x05 +#define IPC_IA_GET_FW_INFO 0x06 +#define IPC_IA_GET_FW_CTXT 0x07 +#define IPC_IA_SET_FW_CTXT 0x08 +#define IPC_IA_PREPARE_SHUTDOWN 0x31 +/* I2L Codec Config/control msgs */ +#define IPC_PREP_D3 0x10 +#define IPC_IA_SET_CODEC_PARAMS 0x10 +#define IPC_IA_GET_CODEC_PARAMS 0x11 +#define IPC_IA_SET_PPP_PARAMS 0x12 +#define IPC_IA_GET_PPP_PARAMS 0x13 +#define IPC_SST_PERIOD_ELAPSED_MRFLD 0xA +#define IPC_IA_ALG_PARAMS 0x1A +#define IPC_IA_TUNING_PARAMS 0x1B +#define IPC_IA_SET_RUNTIME_PARAMS 0x1C +#define IPC_IA_SET_PARAMS 0x1 +#define IPC_IA_GET_PARAMS 0x2 + +#define IPC_EFFECTS_CREATE 0xE +#define IPC_EFFECTS_DESTROY 0xF + +/* I2L Stream config/control msgs */ +#define IPC_IA_ALLOC_STREAM_MRFLD 0x2 +#define IPC_IA_ALLOC_STREAM 0x20 /* Allocate a stream ID */ +#define IPC_IA_FREE_STREAM_MRFLD 0x03 +#define IPC_IA_FREE_STREAM 0x21 /* Free the stream ID */ +#define IPC_IA_SET_STREAM_PARAMS 0x22 +#define IPC_IA_SET_STREAM_PARAMS_MRFLD 0x12 +#define IPC_IA_GET_STREAM_PARAMS 0x23 +#define IPC_IA_PAUSE_STREAM 0x24 +#define IPC_IA_PAUSE_STREAM_MRFLD 0x4 +#define IPC_IA_RESUME_STREAM 0x25 +#define IPC_IA_RESUME_STREAM_MRFLD 0x5 +#define IPC_IA_DROP_STREAM 0x26 +#define IPC_IA_DROP_STREAM_MRFLD 0x07 +#define IPC_IA_DRAIN_STREAM 0x27 /* Short msg with str_id */ +#define IPC_IA_DRAIN_STREAM_MRFLD 0x8 +#define IPC_IA_CONTROL_ROUTING 0x29 +#define IPC_IA_VTSV_UPDATE_MODULES 0x20 +#define IPC_IA_VTSV_DETECTED 0x21 + +#define IPC_IA_START_STREAM_MRFLD 0X06 +#define IPC_IA_START_STREAM 0x30 /* Short msg with str_id */ + +#define IPC_IA_SET_GAIN_MRFLD 0x21 +/* Debug msgs */ +#define IPC_IA_DBG_MEM_READ 0x40 +#define IPC_IA_DBG_MEM_WRITE 0x41 +#define IPC_IA_DBG_LOOP_BACK 0x42 +#define IPC_IA_DBG_LOG_ENABLE 0x45 +#define IPC_IA_DBG_SET_PROBE_PARAMS 0x47 + +/* L2I Firmware/Codec Download msgs */ +#define IPC_IA_FW_INIT_CMPLT 0x81 +#define IPC_IA_FW_INIT_CMPLT_MRFLD 0x01 +#define IPC_IA_FW_ASYNC_ERR_MRFLD 0x11 + +/* L2I Codec Config/control msgs */ +#define IPC_SST_FRAGMENT_ELPASED 0x90 /* Request IA more data */ + +#define IPC_SST_BUF_UNDER_RUN 0x92 /* PB Under run and stopped */ +#define IPC_SST_BUF_OVER_RUN 0x93 /* CAP Under run and stopped */ +#define IPC_SST_DRAIN_END 0x94 /* PB Drain complete and stopped */ +#define IPC_SST_CHNGE_SSP_PARAMS 0x95 /* PB SSP parameters changed */ +#define IPC_SST_STREAM_PROCESS_FATAL_ERR 0x96/* error in processing a stream */ +#define IPC_SST_PERIOD_ELAPSED 0x97 /* period elapsed */ + +#define IPC_SST_ERROR_EVENT 0x99 /* Buffer over run occurred */ +/* L2S messages */ +#define IPC_SC_DDR_LINK_UP 0xC0 +#define IPC_SC_DDR_LINK_DOWN 0xC1 +#define IPC_SC_SET_LPECLK_REQ 0xC2 +#define IPC_SC_SSP_BIT_BANG 0xC3 + +/* L2I Error reporting msgs */ +#define IPC_IA_MEM_ALLOC_FAIL 0xE0 +#define IPC_IA_PROC_ERR 0xE1 /* error in processing a + stream can be used by playback and + capture modules */ + +/* L2I Debug msgs */ +#define IPC_IA_PRINT_STRING 0xF0 + +/* Buffer under-run */ +#define IPC_IA_BUF_UNDER_RUN_MRFLD 0x0B + +/* Mrfld specific defines: + * For asynchronous messages(INIT_CMPLT, PERIOD_ELAPSED, ASYNC_ERROR) + * received from FW, the format is: + * - IPC High: pvt_id is set to zero. Always short message. + * - msg_id is in lower 16-bits of IPC low payload. + * - pipe_id is in higher 16-bits of IPC low payload for period_elapsed. + * - error id is in higher 16-bits of IPC low payload for async errors. + */ +#define SST_ASYNC_DRV_ID 0 + +/* Command Response or Acknowledge message to any IPC message will have + * same message ID and stream ID information which is sent. + * There is no specific Ack message ID. The data field is used as response + * meaning. + */ +enum ackData { + IPC_ACK_SUCCESS = 0, + IPC_ACK_FAILURE, +}; + +enum ipc_ia_msg_id { + IPC_CMD = 1, /*!< Task Control message ID */ + IPC_SET_PARAMS = 2,/*!< Task Set param message ID */ + IPC_GET_PARAMS = 3, /*!< Task Get param message ID */ + IPC_INVALID = 0xFF, /*!<Task Get param message ID */ +}; + enum sst_codec_types { /* AUDIO/MUSIC CODEC Type Definitions */ SST_CODEC_TYPE_UNKNOWN = 0, @@ -35,14 +171,157 @@ enum stream_type { SST_STREAM_TYPE_MUSIC = 1, }; +enum sst_error_codes { + /* Error code,response to msgId: Description */ + /* Common error codes */ + SST_SUCCESS = 0, /* Success */ + SST_ERR_INVALID_STREAM_ID = 1, + SST_ERR_INVALID_MSG_ID = 2, + SST_ERR_INVALID_STREAM_OP = 3, + SST_ERR_INVALID_PARAMS = 4, + SST_ERR_INVALID_CODEC = 5, + SST_ERR_INVALID_MEDIA_TYPE = 6, + SST_ERR_STREAM_ERR = 7, + + SST_ERR_STREAM_IN_USE = 15, +}; + +struct ipc_dsp_hdr { + u16 mod_index_id:8; /*!< DSP Command ID specific to tasks */ + u16 pipe_id:8; /*!< instance of the module in the pipeline */ + u16 mod_id; /*!< Pipe_id */ + u16 cmd_id; /*!< Module ID = lpe_algo_types_t */ + u16 length; /*!< Length of the payload only */ +} __packed; + +union ipc_header_high { + struct { + u32 msg_id:8; /* Message ID - Max 256 Message Types */ + u32 task_id:4; /* Task ID associated with this comand */ + u32 drv_id:4; /* Identifier for the driver to track*/ + u32 rsvd1:8; /* Reserved */ + u32 result:4; /* Reserved */ + u32 res_rqd:1; /* Response rqd */ + u32 large:1; /* Large Message if large = 1 */ + u32 done:1; /* bit 30 - Done bit */ + u32 busy:1; /* bit 31 - busy bit*/ + } part; + u32 full; +} __packed; +/* IPC header */ +union ipc_header_mrfld { + struct { + u32 header_low_payload; + union ipc_header_high header_high; + } p; + u64 full; +} __packed; +/* CAUTION NOTE: All IPC message body must be multiple of 32 bits.*/ + +/* IPC Header */ +union ipc_header { + struct { + u32 msg_id:8; /* Message ID - Max 256 Message Types */ + u32 str_id:5; + u32 large:1; /* Large Message if large = 1 */ + u32 reserved:2; /* Reserved for future use */ + u32 data:14; /* Ack/Info for msg, size of msg in Mailbox */ + u32 done:1; /* bit 30 */ + u32 busy:1; /* bit 31 */ + } part; + u32 full; +} __packed; + +/* Firmware build info */ +struct sst_fw_build_info { + unsigned char date[16]; /* Firmware build date */ + unsigned char time[16]; /* Firmware build time */ +} __packed; + +/* Firmware Version info */ +struct snd_sst_fw_version { + u8 build; /* build number*/ + u8 minor; /* minor number*/ + u8 major; /* major number*/ + u8 type; /* build type */ +}; + +struct ipc_header_fw_init { + struct snd_sst_fw_version fw_version;/* Firmware version details */ + struct sst_fw_build_info build_info; + u16 result; /* Fw init result */ + u8 module_id; /* Module ID in case of error */ + u8 debug_info; /* Debug info from Module ID in case of fail */ +} __packed; + +struct snd_sst_tstamp { + u64 ring_buffer_counter; /* PB/CP: Bytes copied from/to DDR. */ + u64 hardware_counter; /* PB/CP: Bytes DMAed to/from SSP. */ + u64 frames_decoded; + u64 bytes_decoded; + u64 bytes_copied; + u32 sampling_frequency; + u32 channel_peak[8]; +} __packed; + +/* Stream type params struture for Alloc stream */ +struct snd_sst_str_type { + u8 codec_type; /* Codec type */ + u8 str_type; /* 1 = voice 2 = music */ + u8 operation; /* Playback or Capture */ + u8 protected_str; /* 0=Non DRM, 1=DRM */ + u8 time_slots; + u8 reserved; /* Reserved */ + u16 result; /* Result used for acknowledgment */ +} __packed; + +/* Library info structure */ +struct module_info { + u32 lib_version; + u32 lib_type;/*TBD- KLOCKWORK u8 lib_type;*/ + u32 media_type; + u8 lib_name[12]; + u32 lib_caps; + unsigned char b_date[16]; /* Lib build date */ + unsigned char b_time[16]; /* Lib build time */ +} __packed; + +/* Library slot info */ +struct lib_slot_info { + u8 slot_num; /* 1 or 2 */ + u8 reserved1; + u16 reserved2; + u32 iram_size; /* slot size in IRAM */ + u32 dram_size; /* slot size in DRAM */ + u32 iram_offset; /* starting offset of slot in IRAM */ + u32 dram_offset; /* starting offset of slot in DRAM */ +} __packed; + +struct snd_ppp_mixer_params { + __u32 type; /*Type of the parameter */ + __u32 size; + __u32 input_stream_bitmap; /*Input stream Bit Map*/ +} __packed; + +struct snd_sst_lib_download { + struct module_info lib_info; /* library info type, capabilities etc */ + struct lib_slot_info slot_info; /* slot info to be downloaded */ + u32 mod_entry_pt; +}; + +struct snd_sst_lib_download_info { + struct snd_sst_lib_download dload_lib; + u16 result; /* Result used for acknowledgment */ + u8 pvt_id; /* Private ID */ + u8 reserved; /* for alignment */ +}; struct snd_pcm_params { u8 num_chan; /* 1=Mono, 2=Stereo */ u8 pcm_wd_sz; /* 16/24 - bit*/ - u32 reserved; /* Bitrate in bits per second */ - u32 sfreq; /* Sampling rate in Hz */ - u8 use_offload_path; + u8 use_offload_path; /* 0-PCM using period elpased & ALSA interfaces + 1-PCM stream via compressed interface */ u8 reserved2; - u16 reserved3; + u32 sfreq; /* Sampling rate in Hz */ u8 channel_map[8]; } __packed; @@ -76,6 +355,7 @@ struct snd_aac_params { struct snd_wma_params { u8 num_chan; /* 1=Mono, 2=Stereo */ u8 pcm_wd_sz; /* 16/24 - bit*/ + u16 reserved1; u32 brate; /* Use the hard coded value. */ u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */ u32 channel_mask; /* Channel Mask */ @@ -101,26 +381,153 @@ struct sst_address_info { }; struct snd_sst_alloc_params_ext { - struct sst_address_info ring_buf_info[8]; - u8 sg_count; - u8 reserved; - u16 reserved2; - u32 frag_size; /*Number of samples after which period elapsed + __u16 sg_count; + __u16 reserved; + __u32 frag_size; /*Number of samples after which period elapsed message is sent valid only if path = 0*/ -} __packed; + struct sst_address_info ring_buf_info[8]; +}; struct snd_sst_stream_params { union snd_sst_codec_params uc; } __packed; struct snd_sst_params { + u32 result; u32 stream_id; u8 codec; u8 ops; u8 stream_type; u8 device_type; + u8 task; struct snd_sst_stream_params sparams; struct snd_sst_alloc_params_ext aparams; }; +struct snd_sst_alloc_mrfld { + u16 codec_type; + u8 operation; + u8 sg_count; + struct sst_address_info ring_buf_info[8]; + u32 frag_size; + u32 ts; + struct snd_sst_stream_params codec_params; +} __packed; + +/* Alloc stream params structure */ +struct snd_sst_alloc_params { + struct snd_sst_str_type str_type; + struct snd_sst_stream_params stream_params; + struct snd_sst_alloc_params_ext alloc_params; +} __packed; + +/* Alloc stream response message */ +struct snd_sst_alloc_response { + struct snd_sst_str_type str_type; /* Stream type for allocation */ + struct snd_sst_lib_download lib_dnld; /* Valid only for codec dnld */ +}; + +/* Drop response */ +struct snd_sst_drop_response { + u32 result; + u32 bytes; +}; + +struct snd_sst_async_msg { + u32 msg_id; /* Async msg id */ + u32 payload[0]; +}; + +struct snd_sst_async_err_msg { + u32 fw_resp; /* Firmware Result */ + u32 lib_resp; /*Library result */ +} __packed; + +struct snd_sst_vol { + u32 stream_id; + s32 volume; + u32 ramp_duration; + u32 ramp_type; /* Ramp type, default=0 */ +}; + +/* Gain library parameters for mrfld + * based on DSP command spec v0.82 + */ +struct snd_sst_gain_v2 { + u16 gain_cell_num; /* num of gain cells to modify*/ + u8 cell_nbr_idx; /* instance index*/ + u8 cell_path_idx; /* pipe-id */ + u16 module_id; /*module id */ + u16 left_cell_gain; /* left gain value in dB*/ + u16 right_cell_gain; /* right gain value in dB*/ + u16 gain_time_const; /* gain time constant*/ +} __packed; + +struct snd_sst_mute { + u32 stream_id; + u32 mute; +}; + +struct snd_sst_runtime_params { + u8 type; + u8 str_id; + u8 size; + u8 rsvd; + void *addr; +} __packed; + +enum stream_param_type { + SST_SET_TIME_SLOT = 0, + SST_SET_CHANNEL_INFO = 1, + OTHERS = 2, /*reserved for future params*/ +}; + +/* CSV Voice call routing structure */ +struct snd_sst_control_routing { + u8 control; /* 0=start, 1=Stop */ + u8 reserved[3]; /* Reserved- for 32 bit alignment */ +}; + +struct ipc_post { + struct list_head node; + union ipc_header header; /* driver specific */ + bool is_large; + bool is_process_reply; + union ipc_header_mrfld mrfld_header; + char *mailbox_data; +}; + +struct snd_sst_ctxt_params { + u32 address; /* Physical Address in DDR where the context is stored */ + u32 size; /* size of the context */ +}; + +struct snd_sst_lpe_log_params { + u8 dbg_type; + u8 module_id; + u8 log_level; + u8 reserved; +} __packed; + +enum snd_sst_bytes_type { + SND_SST_BYTES_SET = 0x1, + SND_SST_BYTES_GET = 0x2, +}; + +struct snd_sst_bytes_v2 { + u8 type; + u8 ipc_msg; + u8 block; + u8 task_id; + u8 pipe_id; + u8 rsvd; + u16 len; + char bytes[0]; +}; + +#define MAX_VTSV_FILES 2 +struct snd_sst_vtsv_info { + struct sst_address_info vfiles[MAX_VTSV_FILES]; +} __packed; + #endif /* __SST_MFLD_DSP_H__ */ diff --git a/sound/soc/intel/sst-mfld-platform-compress.c b/sound/soc/intel/sst-mfld-platform-compress.c index 02abd19fce1d..29c059ca19e8 100644 --- a/sound/soc/intel/sst-mfld-platform-compress.c +++ b/sound/soc/intel/sst-mfld-platform-compress.c @@ -100,14 +100,19 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, int retval; struct snd_sst_params str_params; struct sst_compress_cb cb; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct sst_data *ctx = snd_soc_platform_get_drvdata(platform); stream = cstream->runtime->private_data; /* construct fw structure for this*/ memset(&str_params, 0, sizeof(str_params)); - str_params.ops = STREAM_OPS_PLAYBACK; - str_params.stream_type = SST_STREAM_TYPE_MUSIC; - str_params.device_type = SND_SST_DEVICE_COMPRESS; + /* fill the device type and stream id to pass to SST driver */ + retval = sst_fill_stream_params(cstream, ctx, &str_params, true); + pr_debug("compr_set_params: fill stream params ret_val = 0x%x\n", retval); + if (retval < 0) + return retval; switch (params->codec.id) { case SND_AUDIOCODEC_MP3: { diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 7c790f51d259..706212a6a68c 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -1,7 +1,7 @@ /* * sst_mfld_platform.c - Intel MID Platform driver * - * Copyright (C) 2010-2013 Intel Corp + * Copyright (C) 2010-2014 Intel Corp * Author: Vinod Koul <vinod.koul@intel.com> * Author: Harsha Priya <priya.harsha@intel.com> * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ @@ -27,7 +27,9 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/compress_driver.h> +#include <asm/platform_sst_audio.h> #include "sst-mfld-platform.h" +#include "sst-atom-controls.h" struct sst_device *sst; static DEFINE_MUTEX(sst_lock); @@ -92,6 +94,13 @@ static struct snd_pcm_hardware sst_platform_pcm_hw = { .fifo_size = SST_FIFO_SIZE, }; +static struct sst_dev_stream_map dpcm_strm_map[] = { + {0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF}, /* Reserved, not in use */ + {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA1_IN, SST_TASK_ID_MEDIA, 0}, + {MERR_DPCM_COMPR, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA0_IN, SST_TASK_ID_MEDIA, 0}, + {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, 0}, +}; + /* MFLD - MSIC */ static struct snd_soc_dai_driver sst_platform_dai[] = { { @@ -143,58 +152,142 @@ static inline int sst_get_stream_status(struct sst_runtime_stream *stream) return state; } +static void sst_fill_alloc_params(struct snd_pcm_substream *substream, + struct snd_sst_alloc_params_ext *alloc_param) +{ + unsigned int channels; + snd_pcm_uframes_t period_size; + ssize_t periodbytes; + ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); + u32 buffer_addr = virt_to_phys(substream->dma_buffer.area); + + channels = substream->runtime->channels; + period_size = substream->runtime->period_size; + periodbytes = samples_to_bytes(substream->runtime, period_size); + alloc_param->ring_buf_info[0].addr = buffer_addr; + alloc_param->ring_buf_info[0].size = buffer_bytes; + alloc_param->sg_count = 1; + alloc_param->reserved = 0; + alloc_param->frag_size = periodbytes * channels; + +} static void sst_fill_pcm_params(struct snd_pcm_substream *substream, - struct sst_pcm_params *param) + struct snd_sst_stream_params *param) { + param->uc.pcm_params.num_chan = (u8) substream->runtime->channels; + param->uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits; + param->uc.pcm_params.sfreq = substream->runtime->rate; + + /* PCM stream via ALSA interface */ + param->uc.pcm_params.use_offload_path = 0; + param->uc.pcm_params.reserved2 = 0; + memset(param->uc.pcm_params.channel_map, 0, sizeof(u8)); - param->num_chan = (u8) substream->runtime->channels; - param->pcm_wd_sz = substream->runtime->sample_bits; - param->reserved = 0; - param->sfreq = substream->runtime->rate; - param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream); - param->period_count = substream->runtime->period_size; - param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area); - pr_debug("period_cnt = %d\n", param->period_count); - pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz); } -static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) +static int sst_get_stream_mapping(int dev, int sdev, int dir, + struct sst_dev_stream_map *map, int size) +{ + int i; + + if (map == NULL) + return -EINVAL; + + + /* index 0 is not used in stream map */ + for (i = 1; i < size; i++) { + if ((map[i].dev_num == dev) && (map[i].direction == dir)) + return i; + } + return 0; +} + +int sst_fill_stream_params(void *substream, + const struct sst_data *ctx, struct snd_sst_params *str_params, bool is_compress) +{ + int map_size; + int index; + struct sst_dev_stream_map *map; + struct snd_pcm_substream *pstream = NULL; + struct snd_compr_stream *cstream = NULL; + + map = ctx->pdata->pdev_strm_map; + map_size = ctx->pdata->strm_map_size; + + if (is_compress == true) + cstream = (struct snd_compr_stream *)substream; + else + pstream = (struct snd_pcm_substream *)substream; + + str_params->stream_type = SST_STREAM_TYPE_MUSIC; + + /* For pcm streams */ + if (pstream) { + index = sst_get_stream_mapping(pstream->pcm->device, + pstream->number, pstream->stream, + map, map_size); + if (index <= 0) + return -EINVAL; + + str_params->stream_id = index; + str_params->device_type = map[index].device_id; + str_params->task = map[index].task_id; + + str_params->ops = (u8)pstream->stream; + } + + if (cstream) { + index = sst_get_stream_mapping(cstream->device->device, + 0, cstream->direction, + map, map_size); + if (index <= 0) + return -EINVAL; + str_params->stream_id = index; + str_params->device_type = map[index].device_id; + str_params->task = map[index].task_id; + + str_params->ops = (u8)cstream->direction; + } + return 0; +} + +static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, + struct snd_soc_platform *platform) { struct sst_runtime_stream *stream = substream->runtime->private_data; - struct sst_pcm_params param = {0}; - struct sst_stream_params str_params = {0}; - int ret_val; + struct snd_sst_stream_params param = {{{0,},},}; + struct snd_sst_params str_params = {0}; + struct snd_sst_alloc_params_ext alloc_params = {0}; + int ret_val = 0; + struct sst_data *ctx = snd_soc_platform_get_drvdata(platform); /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, ¶m); + sst_fill_alloc_params(substream, &alloc_params); substream->runtime->dma_area = substream->dma_buffer.area; str_params.sparams = param; - str_params.codec = param.codec; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - str_params.ops = STREAM_OPS_PLAYBACK; - str_params.device_type = substream->pcm->device + 1; - pr_debug("Playbck stream,Device %d\n", - substream->pcm->device); - } else { - str_params.ops = STREAM_OPS_CAPTURE; - str_params.device_type = SND_SST_DEVICE_CAPTURE; - pr_debug("Capture stream,Device %d\n", - substream->pcm->device); - } - ret_val = stream->ops->open(&str_params); - pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); + str_params.aparams = alloc_params; + str_params.codec = SST_CODEC_TYPE_PCM; + + /* fill the device type and stream id to pass to SST driver */ + ret_val = sst_fill_stream_params(substream, ctx, &str_params, false); if (ret_val < 0) return ret_val; - stream->stream_info.str_id = ret_val; - pr_debug("str id : %d\n", stream->stream_info.str_id); + stream->stream_info.str_id = str_params.stream_id; + + ret_val = stream->ops->open(&str_params); + if (ret_val <= 0) + return ret_val; + + return ret_val; } -static void sst_period_elapsed(void *mad_substream) +static void sst_period_elapsed(void *arg) { - struct snd_pcm_substream *substream = mad_substream; + struct snd_pcm_substream *substream = arg; struct sst_runtime_stream *stream; int status; @@ -218,7 +311,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) pr_debug("setting buffer ptr param\n"); sst_set_stream_status(stream, SST_PLATFORM_INIT); stream->stream_info.period_elapsed = sst_period_elapsed; - stream->stream_info.mad_substream = substream; + stream->stream_info.arg = substream; stream->stream_info.buffer_ptr = 0; stream->stream_info.sfreq = substream->runtime->rate; ret_val = stream->ops->device_control( @@ -230,19 +323,12 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) } /* end -- helper functions */ -static int sst_platform_open(struct snd_pcm_substream *substream) +static int sst_media_open(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { + int ret_val = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct sst_runtime_stream *stream; - int ret_val; - - pr_debug("sst_platform_open called\n"); - - snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw); - ret_val = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret_val < 0) - return ret_val; stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) @@ -251,50 +337,69 @@ static int sst_platform_open(struct snd_pcm_substream *substream) /* get the sst ops */ mutex_lock(&sst_lock); - if (!sst) { + if (!sst || + !try_module_get(sst->dev->driver->owner)) { pr_err("no device available to run\n"); - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; - } - if (!try_module_get(sst->dev->driver->owner)) { - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; + ret_val = -ENODEV; + goto out_ops; } stream->ops = sst->ops; mutex_unlock(&sst_lock); stream->stream_info.str_id = 0; - sst_set_stream_status(stream, SST_PLATFORM_INIT); - stream->stream_info.mad_substream = substream; + + stream->stream_info.arg = substream; /* allocate memory for SST API set */ runtime->private_data = stream; - return 0; + /* Make sure, that the period size is always even */ + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIODS, 2); + + return snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); +out_ops: + kfree(stream); + mutex_unlock(&sst_lock); + return ret_val; } -static int sst_platform_close(struct snd_pcm_substream *substream) +static void sst_media_close(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct sst_runtime_stream *stream; int ret_val = 0, str_id; - pr_debug("sst_platform_close called\n"); stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (str_id) ret_val = stream->ops->close(str_id); module_put(sst->dev->driver->owner); kfree(stream); - return ret_val; } -static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) +static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform, + struct snd_pcm_substream *substream) +{ + struct sst_data *sst = snd_soc_platform_get_drvdata(platform); + struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map; + struct sst_runtime_stream *stream = + substream->runtime->private_data; + u32 str_id = stream->stream_info.str_id; + unsigned int pipe_id; + pipe_id = map[str_id].device_id; + + pr_debug("%s: got pipe_id = %#x for str_id = %d\n", + __func__, pipe_id, str_id); + return pipe_id; +} + +static int sst_media_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct sst_runtime_stream *stream; int ret_val = 0, str_id; - pr_debug("sst_platform_pcm_prepare called\n"); stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { @@ -303,8 +408,8 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) return ret_val; } - ret_val = sst_platform_alloc_stream(substream); - if (ret_val < 0) + ret_val = sst_platform_alloc_stream(substream, dai->platform); + if (ret_val <= 0) return ret_val; snprintf(substream->pcm->id, sizeof(substream->pcm->id), "%d", stream->stream_info.str_id); @@ -316,6 +421,41 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) return ret_val; } +static int sst_media_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); + return 0; +} + +static int sst_media_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + return snd_pcm_lib_free_pages(substream); +} + +static struct snd_soc_dai_ops sst_media_dai_ops = { + .startup = sst_media_open, + .shutdown = sst_media_close, + .prepare = sst_media_prepare, + .hw_params = sst_media_hw_params, + .hw_free = sst_media_hw_free, +}; + +static int sst_platform_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + + if (substream->pcm->internal) + return 0; + + runtime = substream->runtime; + runtime->hw = sst_platform_pcm_hw; + return 0; +} + static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -331,7 +471,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, pr_debug("sst: Trigger Start\n"); str_cmd = SST_SND_START; status = SST_PLATFORM_RUNNING; - stream->stream_info.mad_substream = substream; + stream->stream_info.arg = substream; break; case SNDRV_PCM_TRIGGER_STOP: pr_debug("sst: in stop\n"); @@ -377,32 +517,15 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer pr_err("sst: error code = %d\n", ret_val); return ret_val; } - return stream->stream_info.buffer_ptr; -} - -static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); - - return 0; -} - -static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); + substream->runtime->delay = str_info->pcm_delay; + return str_info->buffer_ptr; } static struct snd_pcm_ops sst_platform_ops = { .open = sst_platform_open, - .close = sst_platform_close, .ioctl = snd_pcm_lib_ioctl, - .prepare = sst_platform_pcm_prepare, .trigger = sst_platform_pcm_trigger, .pointer = sst_platform_pcm_pointer, - .hw_params = sst_platform_pcm_hw_params, - .hw_free = sst_platform_pcm_hw_free, }; static void sst_pcm_free(struct snd_pcm *pcm) @@ -413,15 +536,15 @@ static void sst_pcm_free(struct snd_pcm *pcm) static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int retval = 0; - pr_debug("sst_pcm_new called\n"); - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || - pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + if (dai->driver->playback.channels_min || + dai->driver->capture.channels_min) { retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + snd_dma_continuous_data(GFP_DMA), SST_MIN_BUFFER, SST_MAX_BUFFER); if (retval) { pr_err("dma buffer allocationf fail\n"); @@ -445,10 +568,28 @@ static const struct snd_soc_component_driver sst_component = { static int sst_platform_probe(struct platform_device *pdev) { + struct sst_data *drv; int ret; + struct sst_platform_data *pdata; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); + if (drv == NULL) { + pr_err("kzalloc failed\n"); + return -ENOMEM; + } + + pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); + if (pdata == NULL) { + pr_err("kzalloc failed for pdata\n"); + return -ENOMEM; + } + + pdata->pdev_strm_map = dpcm_strm_map; + pdata->strm_map_size = ARRAY_SIZE(dpcm_strm_map); + drv->pdata = pdata; + mutex_init(&drv->lock); + dev_set_drvdata(&pdev->dev, drv); - pr_debug("sst_platform_probe called\n"); - sst = NULL; ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); if (ret) { pr_err("registering soc platform failed\n"); diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 6c5e7dc49e3c..6c6a42c08e24 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -39,9 +39,10 @@ extern struct sst_device *sst; struct pcm_stream_info { int str_id; - void *mad_substream; - void (*period_elapsed) (void *mad_substream); + void *arg; + void (*period_elapsed) (void *arg); unsigned long long buffer_ptr; + unsigned long long pcm_delay; int sfreq; }; @@ -62,7 +63,9 @@ enum sst_controls { SST_SND_BUFFER_POINTER = 0x05, SST_SND_STREAM_INIT = 0x06, SST_SND_START = 0x07, - SST_MAX_CONTROLS = 0x07, + SST_SET_BYTE_STREAM = 0x100A, + SST_GET_BYTE_STREAM = 0x100B, + SST_MAX_CONTROLS = SST_GET_BYTE_STREAM, }; enum sst_stream_ops { @@ -124,8 +127,9 @@ struct compress_sst_ops { }; struct sst_ops { - int (*open) (struct sst_stream_params *str_param); + int (*open) (struct snd_sst_params *str_param); int (*device_control) (int cmd, void *arg); + int (*set_generic_params)(enum sst_controls cmd, void *arg); int (*close) (unsigned int str_id); }; @@ -143,10 +147,27 @@ struct sst_device { char *name; struct device *dev; struct sst_ops *ops; + struct platform_device *pdev; struct compress_sst_ops *compr_ops; }; +struct sst_data; void sst_set_stream_status(struct sst_runtime_stream *stream, int state); +int sst_fill_stream_params(void *substream, const struct sst_data *ctx, + struct snd_sst_params *str_params, bool is_compress); + +struct sst_algo_int_control_v2 { + struct soc_mixer_control mc; + u16 module_id; /* module identifieer */ + u16 pipe_id; /* location info: pipe_id + instance_id */ + u16 instance_id; + unsigned int value; /* Value received is stored here */ +}; +struct sst_data { + struct platform_device *pdev; + struct sst_platform_data *pdata; + struct mutex lock; +}; int sst_register_dsp(struct sst_device *sst); int sst_unregister_dsp(struct sst_device *sst); #endif diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 06f4e8aa93ae..132bb83f8e99 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,6 @@ config SND_KIRKWOOD_SOC tristate "SoC Audio for the Marvell Kirkwood and Dove chips" - depends on ARCH_KIRKWOOD || ARCH_DOVE || ARCH_MVEBU || MACH_KIRKWOOD || COMPILE_TEST + depends on ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the @@ -15,20 +15,3 @@ config SND_KIRKWOOD_SOC_ARMADA370_DB Say Y if you want to add support for SoC audio on the Armada 370 Development Board. -config SND_KIRKWOOD_SOC_OPENRD - tristate "SoC Audio support for Kirkwood Openrd Client" - depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST) - depends on I2C - select SND_SOC_CS42L51 - help - Say Y if you want to add support for SoC audio on - Openrd Client. - -config SND_KIRKWOOD_SOC_T5325 - tristate "SoC Audio support for HP t5325" - depends on SND_KIRKWOOD_SOC && (MACH_T5325 || COMPILE_TEST) && I2C - select SND_SOC_ALC5623 - help - Say Y if you want to add support for SoC audio on - the HP t5325 thin client. - diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 7c1d8fe09e6b..c36b03d8006c 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -2,10 +2,6 @@ snd-soc-kirkwood-objs := kirkwood-dma.o kirkwood-i2s.o obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o -snd-soc-openrd-objs := kirkwood-openrd.o -snd-soc-t5325-objs := kirkwood-t5325.o snd-soc-armada-370-db-objs := armada-370-db.o obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index aac22fccdcdc..4cf2245950d7 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -28,11 +28,12 @@ static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs) } static struct snd_pcm_hardware kirkwood_dma_snd_hw = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_PAUSE), + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, .buffer_bytes_max = KIRKWOOD_SND_MAX_BUFFER_BYTES, .period_bytes_min = KIRKWOOD_SND_MIN_PERIOD_BYTES, .period_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES, diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 9f842222e798..0704cd6d2314 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -212,7 +212,8 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, KIRKWOOD_PLAYCTL_SIZE_MASK); priv->ctl_play |= ctl_play; } else { - priv->ctl_rec &= ~KIRKWOOD_RECCTL_SIZE_MASK; + priv->ctl_rec &= ~(KIRKWOOD_RECCTL_ENABLE_MASK | + KIRKWOOD_RECCTL_SIZE_MASK); priv->ctl_rec |= ctl_rec; } @@ -221,14 +222,24 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } +static unsigned kirkwood_i2s_play_mute(unsigned ctl) +{ + if (!(ctl & KIRKWOOD_PLAYCTL_I2S_EN)) + ctl |= KIRKWOOD_PLAYCTL_I2S_MUTE; + if (!(ctl & KIRKWOOD_PLAYCTL_SPDIF_EN)) + ctl |= KIRKWOOD_PLAYCTL_SPDIF_MUTE; + return ctl; +} + static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { + struct snd_pcm_runtime *runtime = substream->runtime; struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai); uint32_t ctl, value; ctl = readl(priv->io + KIRKWOOD_PLAYCTL); - if (ctl & KIRKWOOD_PLAYCTL_PAUSE) { + if ((ctl & KIRKWOOD_PLAYCTL_ENABLE_MASK) == 0) { unsigned timeout = 5000; /* * The Armada510 spec says that if we enter pause mode, the @@ -256,14 +267,16 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ else ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ - + ctl = kirkwood_i2s_play_mute(ctl); value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_PLAYCTL); /* enable interrupts */ - value = readl(priv->io + KIRKWOOD_INT_MASK); - value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES; - writel(value, priv->io + KIRKWOOD_INT_MASK); + if (!runtime->no_period_wakeup) { + value = readl(priv->io + KIRKWOOD_INT_MASK); + value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES; + writel(value, priv->io + KIRKWOOD_INT_MASK); + } /* enable playback */ writel(ctl, priv->io + KIRKWOOD_PLAYCTL); @@ -295,6 +308,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | KIRKWOOD_PLAYCTL_SPDIF_MUTE); + ctl = kirkwood_i2s_play_mute(ctl); writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; @@ -322,8 +336,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, else ctl &= ~KIRKWOOD_RECCTL_I2S_EN; /* spdif */ - value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN | - KIRKWOOD_RECCTL_SPDIF_EN); + value = ctl & ~KIRKWOOD_RECCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_RECCTL); /* enable interrupts */ @@ -347,7 +360,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, /* disable all records */ value = readl(priv->io + KIRKWOOD_RECCTL); - value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN); + value &= ~KIRKWOOD_RECCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_RECCTL); break; @@ -411,7 +424,7 @@ static int kirkwood_i2s_init(struct kirkwood_dma_data *priv) writel(value, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_RECCTL); - value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN); + value &= ~KIRKWOOD_RECCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_RECCTL); return 0; diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c deleted file mode 100644 index 65f2a5b9ec3b..000000000000 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ /dev/null @@ -1,109 +0,0 @@ -/* - * kirkwood-openrd.c - * - * (c) 2010 Arnaud Patard <apatard@mandriva.com> - * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org> - * - * 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. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <sound/soc.h> -#include <linux/platform_data/asoc-kirkwood.h> -#include "../codecs/cs42l51.h" - -static int openrd_client_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; - unsigned int freq; - - switch (params_rate(params)) { - default: - case 44100: - freq = 11289600; - break; - case 48000: - freq = 12288000; - break; - case 96000: - freq = 24576000; - break; - } - - return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); - -} - -static struct snd_soc_ops openrd_client_ops = { - .hw_params = openrd_client_hw_params, -}; - - -static struct snd_soc_dai_link openrd_client_dai[] = { -{ - .name = "CS42L51", - .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "i2s", - .platform_name = "mvebu-audio", - .codec_dai_name = "cs42l51-hifi", - .codec_name = "cs42l51-codec.0-004a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, - .ops = &openrd_client_ops, -}, -}; - - -static struct snd_soc_card openrd_client = { - .name = "OpenRD Client", - .owner = THIS_MODULE, - .dai_link = openrd_client_dai, - .num_links = ARRAY_SIZE(openrd_client_dai), -}; - -static int openrd_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &openrd_client; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static int openrd_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - -static struct platform_driver openrd_driver = { - .driver = { - .name = "openrd-client-audio", - .owner = THIS_MODULE, - }, - .probe = openrd_probe, - .remove = openrd_remove, -}; - -module_platform_driver(openrd_driver); - -/* Module information */ -MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>"); -MODULE_DESCRIPTION("ALSA SoC OpenRD Client"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:openrd-client-audio"); diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c deleted file mode 100644 index 844b8415a011..000000000000 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ /dev/null @@ -1,116 +0,0 @@ -/* - * kirkwood-t5325.c - * - * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org> - * - * 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. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <sound/soc.h> -#include <linux/platform_data/asoc-kirkwood.h> -#include "../codecs/alc5623.h" - -static int t5325_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; - unsigned int freq; - - freq = params_rate(params) * 256; - - return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); - -} - -static struct snd_soc_ops t5325_ops = { - .hw_params = t5325_hw_params, -}; - -static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route t5325_route[] = { - { "Headphone Jack", NULL, "HPL" }, - { "Headphone Jack", NULL, "HPR" }, - - {"Speaker", NULL, "SPKOUT"}, - {"Speaker", NULL, "SPKOUTN"}, - - { "MIC1", NULL, "Mic Jack" }, - { "MIC2", NULL, "Mic Jack" }, -}; - -static struct snd_soc_dai_link t5325_dai[] = { -{ - .name = "ALC5621", - .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "i2s", - .platform_name = "mvebu-audio", - .codec_dai_name = "alc5621-hifi", - .codec_name = "alc562x-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, - .ops = &t5325_ops, -}, -}; - -static struct snd_soc_card t5325 = { - .name = "t5325", - .owner = THIS_MODULE, - .dai_link = t5325_dai, - .num_links = ARRAY_SIZE(t5325_dai), - - .dapm_widgets = t5325_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(t5325_dapm_widgets), - .dapm_routes = t5325_route, - .num_dapm_routes = ARRAY_SIZE(t5325_route), -}; - -static int t5325_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &t5325; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static int t5325_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - -static struct platform_driver t5325_driver = { - .driver = { - .name = "t5325-audio", - .owner = THIS_MODULE, - }, - .probe = t5325_probe, - .remove = t5325_remove, -}; - -module_platform_driver(t5325_driver); - -MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>"); -MODULE_DESCRIPTION("ALSA SoC t5325 audio client"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:t5325-audio"); diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index bf23afbba1d7..90e32a781424 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -38,6 +38,9 @@ #define KIRKWOOD_RECCTL_SIZE_24 (1<<0) #define KIRKWOOD_RECCTL_SIZE_32 (0<<0) +#define KIRKWOOD_RECCTL_ENABLE_MASK (KIRKWOOD_RECCTL_SPDIF_EN | \ + KIRKWOOD_RECCTL_I2S_EN) + #define KIRKWOOD_REC_BUF_ADDR 0x1004 #define KIRKWOOD_REC_BUF_SIZE 0x1008 #define KIRKWOOD_REC_BYTE_COUNT 0x100C @@ -121,9 +124,9 @@ /* Theses values come from the marvell alsa driver */ /* need to find where they come from */ -#define KIRKWOOD_SND_MIN_PERIODS 8 +#define KIRKWOOD_SND_MIN_PERIODS 2 #define KIRKWOOD_SND_MAX_PERIODS 16 -#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x800 +#define KIRKWOOD_SND_MIN_PERIOD_BYTES 256 #define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x8000 #define KIRKWOOD_SND_MAX_BUFFER_BYTES (KIRKWOOD_SND_MAX_PERIOD_BYTES \ * KIRKWOOD_SND_MAX_PERIODS) diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 0cc41f94de4e..8c9cc64a9dfb 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -301,7 +301,7 @@ static int cx81801_open(struct tty_struct *tty) static void cx81801_close(struct tty_struct *tty) { struct snd_soc_codec *codec = tty->disc_data; - struct snd_soc_dapm_context *dapm = &codec->card->dapm; + struct snd_soc_dapm_context *dapm = &codec->component.card->dapm; del_timer_sync(&cx81801_timer); diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 6925d7141215..0f34e28a3d55 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -466,7 +466,7 @@ static int asoc_dmic_probe(struct platform_device *pdev) mutex_init(&dmic->mutex); - dmic->fclk = clk_get(dmic->dev, "fck"); + dmic->fclk = devm_clk_get(dmic->dev, "fck"); if (IS_ERR(dmic->fclk)) { dev_err(dmic->dev, "cant get fck\n"); return -ENODEV; @@ -475,8 +475,7 @@ static int asoc_dmic_probe(struct platform_device *pdev) res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma"); if (!res) { dev_err(dmic->dev, "invalid dma memory resource\n"); - ret = -ENODEV; - goto err_put_clk; + return -ENODEV; } dmic->dma_data.addr = res->start + OMAP_DMIC_DATA_REG; @@ -484,34 +483,19 @@ static int asoc_dmic_probe(struct platform_device *pdev) res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); dmic->io_base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(dmic->io_base)) { - ret = PTR_ERR(dmic->io_base); - goto err_put_clk; - } + if (IS_ERR(dmic->io_base)) + return PTR_ERR(dmic->io_base); - ret = snd_soc_register_component(&pdev->dev, &omap_dmic_component, - &omap_dmic_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &omap_dmic_component, + &omap_dmic_dai, 1); if (ret) - goto err_put_clk; + return ret; ret = omap_pcm_platform_register(&pdev->dev); if (ret) - goto err_put_clk; - - return 0; - -err_put_clk: - clk_put(dmic->fclk); - return ret; -} - -static int asoc_dmic_remove(struct platform_device *pdev) -{ - struct omap_dmic *dmic = platform_get_drvdata(pdev); - - snd_soc_unregister_component(&pdev->dev); - clk_put(dmic->fclk); + return ret; return 0; } @@ -529,7 +513,6 @@ static struct platform_driver asoc_dmic_driver = { .of_match_table = omap_dmic_of_match, }, .probe = asoc_dmic_probe, - .remove = asoc_dmic_remove, }; module_platform_driver(asoc_dmic_driver); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index efe2cd699b77..bd3ef2a88be0 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -805,8 +805,9 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) if (ret) return ret; - ret = snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, - &omap_mcbsp_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &omap_mcbsp_component, + &omap_mcbsp_dai, 1); if (ret) return ret; @@ -817,8 +818,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) { struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - snd_soc_unregister_component(&pdev->dev); - if (mcbsp->pdata->ops && mcbsp->pdata->ops->free) mcbsp->pdata->ops->free(mcbsp->id); diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 8d809f8509c8..f4b05bc23e4b 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -31,6 +31,7 @@ #include <sound/pcm_params.h> #include <sound/dmaengine_pcm.h> #include <sound/soc.h> +#include <sound/omap-pcm.h> #ifdef CONFIG_ARCH_OMAP1 #define pcm_omap1510() cpu_is_omap1510() diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 199a8b377553..0109f6c2334e 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -723,7 +723,8 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) ssp_handle = of_parse_phandle(dev->of_node, "port", 0); if (!ssp_handle) { dev_err(dev, "unable to get 'port' phandle\n"); - return -ENODEV; + ret = -ENODEV; + goto err_priv; } priv->ssp = pxa_ssp_request_of(ssp_handle, "SoC audio"); diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig new file mode 100644 index 000000000000..c196a466eef6 --- /dev/null +++ b/sound/soc/rockchip/Kconfig @@ -0,0 +1,12 @@ +config SND_SOC_ROCKCHIP + tristate "ASoC support for Rockchip" + depends on COMPILE_TEST || ARCH_ROCKCHIP + select SND_SOC_GENERIC_DMAENGINE_PCM + select SND_ROCKCHIP_I2S + help + Say Y or M if you want to add support for codecs attached to + the Rockchip SoCs' Audio interfaces. You will also need to + select the audio interfaces to support below. + +config SND_ROCKCHIP_I2S + tristate diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile new file mode 100644 index 000000000000..1006418e1394 --- /dev/null +++ b/sound/soc/rockchip/Makefile @@ -0,0 +1,4 @@ +# ROCKCHIP Platform Support +snd-soc-i2s-objs := rockchip_i2s.o + +obj-$(CONFIG_SND_ROCKCHIP_I2S) += snd-soc-i2s.o diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c new file mode 100644 index 000000000000..8d8e4b59049f --- /dev/null +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -0,0 +1,529 @@ +/* sound/soc/rockchip/rockchip_i2s.c + * + * ALSA SoC Audio Layer - Rockchip I2S Controller driver + * + * Copyright (c) 2014 Rockchip Electronics Co. Ltd. + * Author: Jianqun <jay.xu@rock-chips.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/delay.h> +#include <linux/of_gpio.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <sound/pcm_params.h> +#include <sound/dmaengine_pcm.h> + +#include "rockchip_i2s.h" + +#define DRV_NAME "rockchip-i2s" + +struct rk_i2s_dev { + struct device *dev; + + struct clk *hclk; + struct clk *mclk; + + struct snd_dmaengine_dai_dma_data capture_dma_data; + struct snd_dmaengine_dai_dma_data playback_dma_data; + + struct regmap *regmap; + +/* + * Used to indicate the tx/rx status. + * I2S controller hopes to start the tx and rx together, + * also to stop them when they are both try to stop. +*/ + bool tx_start; + bool rx_start; +}; + +static int i2s_runtime_suspend(struct device *dev) +{ + struct rk_i2s_dev *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->mclk); + + return 0; +} + +static int i2s_runtime_resume(struct device *dev) +{ + struct rk_i2s_dev *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(i2s->mclk); + if (ret) { + dev_err(i2s->dev, "clock enable failed %d\n", ret); + return ret; + } + + return 0; +} + +static inline struct rk_i2s_dev *to_info(struct snd_soc_dai *dai) +{ + return snd_soc_dai_get_drvdata(dai); +} + +static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) +{ + unsigned int val = 0; + int retry = 10; + + if (on) { + regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); + + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); + + i2s->tx_start = true; + } else { + i2s->tx_start = false; + + regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); + + if (!i2s->rx_start) { + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | + I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | + I2S_XFER_RXS_STOP); + + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); + + regmap_read(i2s->regmap, I2S_CLR, &val); + + /* Should wait for clear operation to finish */ + while (val) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) + dev_warn(i2s->dev, "fail to clear\n"); + } + } + } +} + +static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) +{ + unsigned int val = 0; + int retry = 10; + + if (on) { + regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); + + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); + + i2s->rx_start = true; + } else { + i2s->rx_start = false; + + regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); + + if (!i2s->tx_start) { + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | + I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | + I2S_XFER_RXS_STOP); + + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); + + regmap_read(i2s->regmap, I2S_CLR, &val); + + /* Should wait for clear operation to finish */ + while (val) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) + dev_warn(i2s->dev, "fail to clear\n"); + } + } + } +} + +static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct rk_i2s_dev *i2s = to_info(cpu_dai); + unsigned int mask = 0, val = 0; + + mask = I2S_CKR_MSS_SLAVE; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + val = I2S_CKR_MSS_SLAVE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + val = I2S_CKR_MSS_MASTER; + break; + default: + return -EINVAL; + } + + regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); + + mask = I2S_TXCR_IBM_MASK; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + val = I2S_TXCR_IBM_RSJM; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = I2S_TXCR_IBM_LSJM; + break; + case SND_SOC_DAIFMT_I2S: + val = I2S_TXCR_IBM_NORMAL; + break; + default: + return -EINVAL; + } + + regmap_update_bits(i2s->regmap, I2S_TXCR, mask, val); + + mask = I2S_RXCR_IBM_MASK; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + val = I2S_RXCR_IBM_RSJM; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = I2S_RXCR_IBM_LSJM; + break; + case SND_SOC_DAIFMT_I2S: + val = I2S_RXCR_IBM_NORMAL; + break; + default: + return -EINVAL; + } + + regmap_update_bits(i2s->regmap, I2S_RXCR, mask, val); + + return 0; +} + +static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct rk_i2s_dev *i2s = to_info(dai); + unsigned int val = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + val |= I2S_TXCR_VDW(8); + break; + case SNDRV_PCM_FORMAT_S16_LE: + val |= I2S_TXCR_VDW(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val |= I2S_TXCR_VDW(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + val |= I2S_TXCR_VDW(24); + break; + default: + return -EINVAL; + } + + regmap_update_bits(i2s->regmap, I2S_TXCR, I2S_TXCR_VDW_MASK, val); + regmap_update_bits(i2s->regmap, I2S_RXCR, I2S_RXCR_VDW_MASK, val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dai->playback_dma_data = &i2s->playback_dma_data; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK, + I2S_DMACR_TDL(1) | I2S_DMACR_TDE_ENABLE); + } else { + dai->capture_dma_data = &i2s->capture_dma_data; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDL_MASK, + I2S_DMACR_RDL(1) | I2S_DMACR_RDE_ENABLE); + } + + return 0; +} + +static int rockchip_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct rk_i2s_dev *i2s = to_info(dai); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + rockchip_snd_rxctrl(i2s, 1); + else + rockchip_snd_txctrl(i2s, 1); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + rockchip_snd_rxctrl(i2s, 0); + else + rockchip_snd_txctrl(i2s, 0); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int rockchip_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct rk_i2s_dev *i2s = to_info(cpu_dai); + int ret; + + ret = clk_set_rate(i2s->mclk, freq); + if (ret) + dev_err(i2s->dev, "Fail to set mclk %d\n", ret); + + return ret; +} + +static const struct snd_soc_dai_ops rockchip_i2s_dai_ops = { + .hw_params = rockchip_i2s_hw_params, + .set_sysclk = rockchip_i2s_set_sysclk, + .set_fmt = rockchip_i2s_set_fmt, + .trigger = rockchip_i2s_trigger, +}; + +static struct snd_soc_dai_driver rockchip_i2s_dai = { + .playback = { + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = (SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = (SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &rockchip_i2s_dai_ops, +}; + +static const struct snd_soc_component_driver rockchip_i2s_component = { + .name = DRV_NAME, +}; + +static bool rockchip_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_TXCR: + case I2S_RXCR: + case I2S_CKR: + case I2S_DMACR: + case I2S_INTCR: + case I2S_XFER: + case I2S_CLR: + case I2S_TXDR: + return true; + default: + return false; + } +} + +static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_TXCR: + case I2S_RXCR: + case I2S_CKR: + case I2S_DMACR: + case I2S_INTCR: + case I2S_XFER: + case I2S_CLR: + case I2S_RXDR: + return true; + default: + return false; + } +} + +static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_FIFOLR: + case I2S_INTSR: + return true; + default: + return false; + } +} + +static bool rockchip_i2s_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_FIFOLR: + return true; + default: + return false; + } +} + +static const struct regmap_config rockchip_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = I2S_RXDR, + .writeable_reg = rockchip_i2s_wr_reg, + .readable_reg = rockchip_i2s_rd_reg, + .volatile_reg = rockchip_i2s_volatile_reg, + .precious_reg = rockchip_i2s_precious_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int rockchip_i2s_probe(struct platform_device *pdev) +{ + struct rk_i2s_dev *i2s; + struct resource *res; + void __iomem *regs; + int ret; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) { + dev_err(&pdev->dev, "Can't allocate rk_i2s_dev\n"); + return -ENOMEM; + } + + /* try to prepare related clocks */ + i2s->hclk = devm_clk_get(&pdev->dev, "i2s_hclk"); + if (IS_ERR(i2s->hclk)) { + dev_err(&pdev->dev, "Can't retrieve i2s bus clock\n"); + return PTR_ERR(i2s->hclk); + } + + i2s->mclk = devm_clk_get(&pdev->dev, "i2s_clk"); + if (IS_ERR(i2s->mclk)) { + dev_err(&pdev->dev, "Can't retrieve i2s master clock\n"); + return PTR_ERR(i2s->mclk); + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &rockchip_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, + "Failed to initialise managed register map\n"); + return PTR_ERR(i2s->regmap); + } + + i2s->playback_dma_data.addr = res->start + I2S_TXDR; + i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->playback_dma_data.maxburst = 16; + + i2s->capture_dma_data.addr = res->start + I2S_RXDR; + i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->capture_dma_data.maxburst = 16; + + i2s->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, i2s); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = i2s_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + + ret = devm_snd_soc_register_component(&pdev->dev, + &rockchip_i2s_component, + &rockchip_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI\n"); + goto err_suspend; + } + + ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM\n"); + goto err_pcm_register; + } + + return 0; + +err_pcm_register: + snd_dmaengine_pcm_unregister(&pdev->dev); +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + i2s_runtime_suspend(&pdev->dev); +err_pm_disable: + pm_runtime_disable(&pdev->dev); + + return ret; +} + +static int rockchip_i2s_remove(struct platform_device *pdev) +{ + struct rk_i2s_dev *i2s = dev_get_drvdata(&pdev->dev); + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + i2s_runtime_suspend(&pdev->dev); + + clk_disable_unprepare(i2s->mclk); + clk_disable_unprepare(i2s->hclk); + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); + + return 0; +} + +static const struct of_device_id rockchip_i2s_match[] = { + { .compatible = "rockchip,rk3066-i2s", }, + {}, +}; + +static const struct dev_pm_ops rockchip_i2s_pm_ops = { + SET_RUNTIME_PM_OPS(i2s_runtime_suspend, i2s_runtime_resume, + NULL) +}; + +static struct platform_driver rockchip_i2s_driver = { + .probe = rockchip_i2s_probe, + .remove = rockchip_i2s_remove, + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(rockchip_i2s_match), + .pm = &rockchip_i2s_pm_ops, + }, +}; +module_platform_driver(rockchip_i2s_driver); + +MODULE_DESCRIPTION("ROCKCHIP IIS ASoC Interface"); +MODULE_AUTHOR("jianqun <jay.xu@rock-chips.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, rockchip_i2s_match); diff --git a/sound/soc/rockchip/rockchip_i2s.h b/sound/soc/rockchip/rockchip_i2s.h new file mode 100644 index 000000000000..89a5d8bc6ee7 --- /dev/null +++ b/sound/soc/rockchip/rockchip_i2s.h @@ -0,0 +1,223 @@ +/* + * sound/soc/rockchip/rockchip_i2s.h + * + * ALSA SoC Audio Layer - Rockchip I2S Controller driver + * + * Copyright (c) 2014 Rockchip Electronics Co. Ltd. + * Author: Jianqun xu <jay.xu@rock-chips.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 _ROCKCHIP_IIS_H +#define _ROCKCHIP_IIS_H + +/* + * TXCR + * transmit operation control register +*/ +#define I2S_TXCR_RCNT_SHIFT 17 +#define I2S_TXCR_RCNT_MASK (0x3f << I2S_TXCR_RCNT_SHIFT) +#define I2S_TXCR_CSR_SHIFT 15 +#define I2S_TXCR_CSR(x) (x << I2S_TXCR_CSR_SHIFT) +#define I2S_TXCR_CSR_MASK (3 << I2S_TXCR_CSR_SHIFT) +#define I2S_TXCR_HWT BIT(14) +#define I2S_TXCR_SJM_SHIFT 12 +#define I2S_TXCR_SJM_R (0 << I2S_TXCR_SJM_SHIFT) +#define I2S_TXCR_SJM_L (1 << I2S_TXCR_SJM_SHIFT) +#define I2S_TXCR_FBM_SHIFT 11 +#define I2S_TXCR_FBM_MSB (0 << I2S_TXCR_FBM_SHIFT) +#define I2S_TXCR_FBM_LSB (1 << I2S_TXCR_FBM_SHIFT) +#define I2S_TXCR_IBM_SHIFT 9 +#define I2S_TXCR_IBM_NORMAL (0 << I2S_TXCR_IBM_SHIFT) +#define I2S_TXCR_IBM_LSJM (1 << I2S_TXCR_IBM_SHIFT) +#define I2S_TXCR_IBM_RSJM (2 << I2S_TXCR_IBM_SHIFT) +#define I2S_TXCR_IBM_MASK (3 << I2S_TXCR_IBM_SHIFT) +#define I2S_TXCR_PBM_SHIFT 7 +#define I2S_TXCR_PBM_MODE(x) (x << I2S_TXCR_PBM_SHIFT) +#define I2S_TXCR_PBM_MASK (3 << I2S_TXCR_PBM_SHIFT) +#define I2S_TXCR_TFS_SHIFT 5 +#define I2S_TXCR_TFS_I2S (0 << I2S_TXCR_TFS_SHIFT) +#define I2S_TXCR_TFS_PCM (1 << I2S_TXCR_TFS_SHIFT) +#define I2S_TXCR_VDW_SHIFT 0 +#define I2S_TXCR_VDW(x) ((x - 1) << I2S_TXCR_VDW_SHIFT) +#define I2S_TXCR_VDW_MASK (0x1f << I2S_TXCR_VDW_SHIFT) + +/* + * RXCR + * receive operation control register +*/ +#define I2S_RXCR_HWT BIT(14) +#define I2S_RXCR_SJM_SHIFT 12 +#define I2S_RXCR_SJM_R (0 << I2S_RXCR_SJM_SHIFT) +#define I2S_RXCR_SJM_L (1 << I2S_RXCR_SJM_SHIFT) +#define I2S_RXCR_FBM_SHIFT 11 +#define I2S_RXCR_FBM_MSB (0 << I2S_RXCR_FBM_SHIFT) +#define I2S_RXCR_FBM_LSB (1 << I2S_RXCR_FBM_SHIFT) +#define I2S_RXCR_IBM_SHIFT 9 +#define I2S_RXCR_IBM_NORMAL (0 << I2S_RXCR_IBM_SHIFT) +#define I2S_RXCR_IBM_LSJM (1 << I2S_RXCR_IBM_SHIFT) +#define I2S_RXCR_IBM_RSJM (2 << I2S_RXCR_IBM_SHIFT) +#define I2S_RXCR_IBM_MASK (3 << I2S_RXCR_IBM_SHIFT) +#define I2S_RXCR_PBM_SHIFT 7 +#define I2S_RXCR_PBM_MODE(x) (x << I2S_RXCR_PBM_SHIFT) +#define I2S_RXCR_PBM_MASK (3 << I2S_RXCR_PBM_SHIFT) +#define I2S_RXCR_TFS_SHIFT 5 +#define I2S_RXCR_TFS_I2S (0 << I2S_RXCR_TFS_SHIFT) +#define I2S_RXCR_TFS_PCM (1 << I2S_RXCR_TFS_SHIFT) +#define I2S_RXCR_VDW_SHIFT 0 +#define I2S_RXCR_VDW(x) ((x - 1) << I2S_RXCR_VDW_SHIFT) +#define I2S_RXCR_VDW_MASK (0x1f << I2S_RXCR_VDW_SHIFT) + +/* + * CKR + * clock generation register +*/ +#define I2S_CKR_MSS_SHIFT 27 +#define I2S_CKR_MSS_MASTER (0 << I2S_CKR_MSS_SHIFT) +#define I2S_CKR_MSS_SLAVE (1 << I2S_CKR_MSS_SHIFT) +#define I2S_CKR_MSS_MASK (1 << I2S_CKR_MSS_SHIFT) +#define I2S_CKR_CKP_SHIFT 26 +#define I2S_CKR_CKP_NEG (0 << I2S_CKR_CKP_SHIFT) +#define I2S_CKR_CKP_POS (1 << I2S_CKR_CKP_SHIFT) +#define I2S_CKR_RLP_SHIFT 25 +#define I2S_CKR_RLP_NORMAL (0 << I2S_CKR_RLP_SHIFT) +#define I2S_CKR_RLP_OPPSITE (1 << I2S_CKR_RLP_SHIFT) +#define I2S_CKR_TLP_SHIFT 24 +#define I2S_CKR_TLP_NORMAL (0 << I2S_CKR_TLP_SHIFT) +#define I2S_CKR_TLP_OPPSITE (1 << I2S_CKR_TLP_SHIFT) +#define I2S_CKR_MDIV_SHIFT 16 +#define I2S_CKR_MDIV(x) ((x - 1) << I2S_CKR_MDIV_SHIFT) +#define I2S_CKR_MDIV_MASK (0xff << I2S_CKR_MDIV_SHIFT) +#define I2S_CKR_RSD_SHIFT 8 +#define I2S_CKR_RSD(x) ((x - 1) << I2S_CKR_RSD_SHIFT) +#define I2S_CKR_RSD_MASK (0xff << I2S_CKR_RSD_SHIFT) +#define I2S_CKR_TSD_SHIFT 0 +#define I2S_CKR_TSD(x) ((x - 1) << I2S_CKR_TSD_SHIFT) +#define I2S_CKR_TSD_MASK (0xff << I2S_CKR_TSD_SHIFT) + +/* + * FIFOLR + * FIFO level register +*/ +#define I2S_FIFOLR_RFL_SHIFT 24 +#define I2S_FIFOLR_RFL_MASK (0x3f << I2S_FIFOLR_RFL_SHIFT) +#define I2S_FIFOLR_TFL3_SHIFT 18 +#define I2S_FIFOLR_TFL3_MASK (0x3f << I2S_FIFOLR_TFL3_SHIFT) +#define I2S_FIFOLR_TFL2_SHIFT 12 +#define I2S_FIFOLR_TFL2_MASK (0x3f << I2S_FIFOLR_TFL2_SHIFT) +#define I2S_FIFOLR_TFL1_SHIFT 6 +#define I2S_FIFOLR_TFL1_MASK (0x3f << I2S_FIFOLR_TFL1_SHIFT) +#define I2S_FIFOLR_TFL0_SHIFT 0 +#define I2S_FIFOLR_TFL0_MASK (0x3f << I2S_FIFOLR_TFL0_SHIFT) + +/* + * DMACR + * DMA control register +*/ +#define I2S_DMACR_RDE_SHIFT 24 +#define I2S_DMACR_RDE_DISABLE (0 << I2S_DMACR_RDE_SHIFT) +#define I2S_DMACR_RDE_ENABLE (1 << I2S_DMACR_RDE_SHIFT) +#define I2S_DMACR_RDL_SHIFT 16 +#define I2S_DMACR_RDL(x) ((x - 1) << I2S_DMACR_RDL_SHIFT) +#define I2S_DMACR_RDL_MASK (0x1f << I2S_DMACR_RDL_SHIFT) +#define I2S_DMACR_TDE_SHIFT 8 +#define I2S_DMACR_TDE_DISABLE (0 << I2S_DMACR_TDE_SHIFT) +#define I2S_DMACR_TDE_ENABLE (1 << I2S_DMACR_TDE_SHIFT) +#define I2S_DMACR_TDL_SHIFT 0 +#define I2S_DMACR_TDL(x) ((x - 1) << I2S_DMACR_TDL_SHIFT) +#define I2S_DMACR_TDL_MASK (0x1f << I2S_DMACR_TDL_SHIFT) + +/* + * INTCR + * interrupt control register +*/ +#define I2S_INTCR_RFT_SHIFT 20 +#define I2S_INTCR_RFT(x) ((x - 1) << I2S_INTCR_RFT_SHIFT) +#define I2S_INTCR_RXOIC BIT(18) +#define I2S_INTCR_RXOIE_SHIFT 17 +#define I2S_INTCR_RXOIE_DISABLE (0 << I2S_INTCR_RXOIE_SHIFT) +#define I2S_INTCR_RXOIE_ENABLE (1 << I2S_INTCR_RXOIE_SHIFT) +#define I2S_INTCR_RXFIE_SHIFT 16 +#define I2S_INTCR_RXFIE_DISABLE (0 << I2S_INTCR_RXFIE_SHIFT) +#define I2S_INTCR_RXFIE_ENABLE (1 << I2S_INTCR_RXFIE_SHIFT) +#define I2S_INTCR_TFT_SHIFT 4 +#define I2S_INTCR_TFT(x) ((x - 1) << I2S_INTCR_TFT_SHIFT) +#define I2S_INTCR_TFT_MASK (0x1f << I2S_INTCR_TFT_SHIFT) +#define I2S_INTCR_TXUIC BIT(2) +#define I2S_INTCR_TXUIE_SHIFT 1 +#define I2S_INTCR_TXUIE_DISABLE (0 << I2S_INTCR_TXUIE_SHIFT) +#define I2S_INTCR_TXUIE_ENABLE (1 << I2S_INTCR_TXUIE_SHIFT) + +/* + * INTSR + * interrupt status register +*/ +#define I2S_INTSR_TXEIE_SHIFT 0 +#define I2S_INTSR_TXEIE_DISABLE (0 << I2S_INTSR_TXEIE_SHIFT) +#define I2S_INTSR_TXEIE_ENABLE (1 << I2S_INTSR_TXEIE_SHIFT) +#define I2S_INTSR_RXOI_SHIFT 17 +#define I2S_INTSR_RXOI_INA (0 << I2S_INTSR_RXOI_SHIFT) +#define I2S_INTSR_RXOI_ACT (1 << I2S_INTSR_RXOI_SHIFT) +#define I2S_INTSR_RXFI_SHIFT 16 +#define I2S_INTSR_RXFI_INA (0 << I2S_INTSR_RXFI_SHIFT) +#define I2S_INTSR_RXFI_ACT (1 << I2S_INTSR_RXFI_SHIFT) +#define I2S_INTSR_TXUI_SHIFT 1 +#define I2S_INTSR_TXUI_INA (0 << I2S_INTSR_TXUI_SHIFT) +#define I2S_INTSR_TXUI_ACT (1 << I2S_INTSR_TXUI_SHIFT) +#define I2S_INTSR_TXEI_SHIFT 0 +#define I2S_INTSR_TXEI_INA (0 << I2S_INTSR_TXEI_SHIFT) +#define I2S_INTSR_TXEI_ACT (1 << I2S_INTSR_TXEI_SHIFT) + +/* + * XFER + * Transfer start register +*/ +#define I2S_XFER_RXS_SHIFT 1 +#define I2S_XFER_RXS_STOP (0 << I2S_XFER_RXS_SHIFT) +#define I2S_XFER_RXS_START (1 << I2S_XFER_RXS_SHIFT) +#define I2S_XFER_TXS_SHIFT 0 +#define I2S_XFER_TXS_STOP (0 << I2S_XFER_TXS_SHIFT) +#define I2S_XFER_TXS_START (1 << I2S_XFER_TXS_SHIFT) + +/* + * CLR + * clear SCLK domain logic register +*/ +#define I2S_CLR_RXC BIT(1) +#define I2S_CLR_TXC BIT(0) + +/* + * TXDR + * Transimt FIFO data register, write only. +*/ +#define I2S_TXDR_MASK (0xff) + +/* + * RXDR + * Receive FIFO data register, write only. +*/ +#define I2S_RXDR_MASK (0xff) + +/* Clock divider id */ +enum { + ROCKCHIP_DIV_MCLK = 0, + ROCKCHIP_DIV_BCLK, +}; + +/* I2S REGS */ +#define I2S_TXCR (0x0000) +#define I2S_RXCR (0x0004) +#define I2S_CKR (0x0008) +#define I2S_FIFOLR (0x000c) +#define I2S_DMACR (0x0010) +#define I2S_INTCR (0x0014) +#define I2S_INTSR (0x0018) +#define I2S_XFER (0x001c) +#define I2S_CLR (0x0020) +#define I2S_TXDR (0x0024) +#define I2S_RXDR (0x0028) + +#endif /* _ROCKCHIP_IIS_H */ diff --git a/sound/soc/s6000/Kconfig b/sound/soc/s6000/Kconfig index c74eb3d4a47c..f244a2566f20 100644 --- a/sound/soc/s6000/Kconfig +++ b/sound/soc/s6000/Kconfig @@ -1,17 +1,24 @@ config SND_S6000_SOC tristate "SoC Audio for the Stretch s6000 family" - depends on XTENSA_VARIANT_S6000 + depends on XTENSA_VARIANT_S6000 || COMPILE_TEST + depends on HAS_IOMEM + select SND_S6000_SOC_PCM if XTENSA_VARIANT_S6000 help Say Y or M if you want to add support for codecs attached to s6000 family chips. You will also need to select the platform to support below. +config SND_S6000_SOC_PCM + tristate + config SND_S6000_SOC_I2S tristate config SND_S6000_SOC_S6IPCAM - tristate "SoC Audio support for Stretch 6105 IP Camera" - depends on SND_S6000_SOC && XTENSA_PLATFORM_S6105 + bool "SoC Audio support for Stretch 6105 IP Camera" + depends on SND_S6000_SOC=y + depends on I2C=y + depends on XTENSA_PLATFORM_S6105 || COMPILE_TEST select SND_S6000_SOC_I2S select SND_SOC_TLV320AIC3X help diff --git a/sound/soc/s6000/Makefile b/sound/soc/s6000/Makefile index 7a613612e010..0f0ae2a012aa 100644 --- a/sound/soc/s6000/Makefile +++ b/sound/soc/s6000/Makefile @@ -2,7 +2,7 @@ snd-soc-s6000-objs := s6000-pcm.o snd-soc-s6000-i2s-objs := s6000-i2s.o -obj-$(CONFIG_SND_S6000_SOC) += snd-soc-s6000.o +obj-$(CONFIG_SND_S6000_SOC_PCM) += snd-soc-s6000.o obj-$(CONFIG_SND_S6000_SOC_I2S) += snd-soc-s6000-i2s.o # s6105 Machine Support diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index 7eba7979b9af..1c8d01166e5b 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -570,7 +570,7 @@ err_release_none: return ret; } -static void s6000_i2s_remove(struct platform_device *pdev) +static int s6000_i2s_remove(struct platform_device *pdev) { struct s6000_i2s_dev *dev = dev_get_drvdata(&pdev->dev); struct resource *region; @@ -597,6 +597,8 @@ static void s6000_i2s_remove(struct platform_device *pdev) iounmap(mmio); region = platform_get_resource(pdev, IORESOURCE_IO, 0); release_mem_region(region->start, resource_size(region)); + + return 0; } static struct platform_driver s6000_i2s_driver = { diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 0b21d1dc80c1..3510c01f8a6a 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -19,8 +19,6 @@ #include <sound/pcm.h> #include <sound/soc.h> -#include <variant/dmac.h> - #include "s6000-pcm.h" #include "s6000-i2s.h" @@ -135,22 +133,8 @@ static const struct snd_kcontrol_new audio_out_mux = { /* Logic for a aic3x as connected on the s6105 ip camera ref design */ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_card *card = rtd->card; - /* not present */ - snd_soc_dapm_nc_pin(dapm, "MONO_LOUT"); - snd_soc_dapm_nc_pin(dapm, "LINE2L"); - snd_soc_dapm_nc_pin(dapm, "LINE2R"); - - /* not connected */ - snd_soc_dapm_nc_pin(dapm, "MIC3L"); /* LINE2L on this chip */ - snd_soc_dapm_nc_pin(dapm, "MIC3R"); /* LINE2R on this chip */ - snd_soc_dapm_nc_pin(dapm, "LLOUT"); - snd_soc_dapm_nc_pin(dapm, "RLOUT"); - snd_soc_dapm_nc_pin(dapm, "HPRCOM"); - /* must correspond to audio_out_mux.private_value initializer */ snd_soc_dapm_disable_pin(&card->dapm, "Audio Out Differential"); @@ -182,6 +166,7 @@ static struct snd_soc_card snd_soc_card_s6105 = { .num_dapm_widgets = ARRAY_SIZE(aic3x_dapm_widgets), .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), + .fully_routed = true, }; static struct s6000_snd_platform_data s6105_snd_data __initdata = { diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 753b8c93ab51..55a38697443d 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,25 +1,16 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" depends on PLAT_SAMSUNG - select S3C2410_DMA if ARCH_S3C24XX - select S3C64XX_PL080 if ARCH_S3C64XX - select SND_S3C_DMA if !ARCH_S3C24XX - select SND_S3C_DMA_LEGACY if ARCH_S3C24XX - select SND_SOC_GENERIC_DMAENGINE_PCM if !ARCH_S3C24XX + depends on S3C64XX_PL080 || !ARCH_S3C64XX + depends on S3C24XX_DMAC || !ARCH_S3C24XX + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the Samsung SoCs' Audio interfaces. You will also need to select the audio interfaces to support below. -config SND_S3C_DMA - tristate - -config SND_S3C_DMA_LEGACY - tristate - config SND_S3C24XX_I2S tristate - select S3C24XX_DMA config SND_S3C_I2SV2_SOC tristate @@ -27,7 +18,6 @@ config SND_S3C_I2SV2_SOC config SND_S3C2412_SOC_I2S tristate select SND_S3C_I2SV2_SOC - select S3C2410_DMA config SND_SAMSUNG_PCM tristate @@ -55,7 +45,7 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753 config SND_SOC_SAMSUNG_JIVE_WM8750 tristate "SoC I2S Audio support for Jive" - depends on SND_SOC_SAMSUNG && MACH_JIVE + depends on SND_SOC_SAMSUNG && MACH_JIVE && I2C select SND_SOC_WM8750 select SND_S3C2412_SOC_I2S help @@ -63,7 +53,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750 config SND_SOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDK6440 || MACH_SMDK6450 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) depends on REGMAP_I2C select SND_SOC_WM8580 select SND_SAMSUNG_I2S @@ -83,7 +73,6 @@ config SND_SOC_SAMSUNG_SMDK_WM8994 config SND_SOC_SAMSUNG_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_SOC_SAMSUNG && MACH_SMDK2443 - select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_CODEC select SND_SAMSUNG_AC97 @@ -94,7 +83,6 @@ config SND_SOC_SAMSUNG_SMDK2443_WM9710 config SND_SOC_SAMSUNG_LN2440SBC_ALC650 tristate "SoC AC97 Audio support for LN2440SBC - ALC650" depends on SND_SOC_SAMSUNG && ARCH_S3C24XX - select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_CODEC select SND_SAMSUNG_AC97 @@ -154,7 +142,7 @@ config SND_SOC_SAMSUNG_SMDK_WM9713 config SND_SOC_SMARTQ tristate "SoC I2S Audio support for SmartQ board" - depends on SND_SOC_SAMSUNG && MACH_SMARTQ + depends on SND_SOC_SAMSUNG && MACH_SMARTQ && I2C select SND_SAMSUNG_I2S select SND_SOC_WM8750 @@ -178,7 +166,7 @@ config SND_SOC_SAMSUNG_SMDK_SPDIF config SND_SOC_SMDK_WM8580_PCM tristate "SoC PCM Audio support for WM8580 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6450 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && (MACH_SMDKV210 || MACH_SMDKC110) depends on REGMAP_I2C select SND_SOC_WM8580 select SND_SAMSUNG_PCM @@ -206,7 +194,7 @@ config SND_SOC_SPEYSIDE config SND_SOC_TOBERMORY tristate "Audio support for Wolfson Tobermory" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && INPUT + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && INPUT && I2C select SND_SAMSUNG_I2S select SND_SOC_WM8962 @@ -222,7 +210,7 @@ config SND_SOC_BELLS config SND_SOC_LOWLAND tristate "Audio support for Wolfson Lowland" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C select SND_SAMSUNG_I2S select SND_SOC_WM5100 select SND_SOC_WM9081 @@ -236,10 +224,18 @@ config SND_SOC_LITTLEMILL config SND_SOC_SNOW tristate "Audio support for Google Snow boards" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SOC_MAX98090 select SND_SOC_MAX98095 select SND_SAMSUNG_I2S help Say Y if you want to add audio support for various Snow boards based on Exynos5 series of SoCs. + +config SND_SOC_ODROIDX2 + tristate "Audio support for Odroid-X2 and Odroid-U3" + depends on SND_SOC_SAMSUNG + select SND_SOC_MAX98090 + select SND_SAMSUNG_I2S + help + Say Y here to enable audio support for the Odroid-X2/U3. diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 6d0212ba571c..91505ddaaf95 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,6 +1,5 @@ # S3c24XX Platform Support snd-soc-s3c-dma-objs := dmaengine.o -snd-soc-s3c-dma-legacy-objs := dma.o snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o @@ -10,8 +9,7 @@ snd-soc-samsung-spdif-objs := spdif.o snd-soc-pcm-objs := pcm.o snd-soc-i2s-objs := i2s.o -obj-$(CONFIG_SND_S3C_DMA) += snd-soc-s3c-dma.o -obj-$(CONFIG_SND_S3C_DMA_LEGACY) += snd-soc-s3c-dma-legacy.o +obj-$(CONFIG_SND_SOC_SAMSUNG) += snd-soc-s3c-dma.o obj-$(CONFIG_SND_S3C24XX_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_SAMSUNG_AC97) += snd-soc-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o @@ -46,6 +44,7 @@ snd-soc-tobermory-objs := tobermory.o snd-soc-lowland-objs := lowland.o snd-soc-littlemill-objs := littlemill.o snd-soc-bells-objs := bells.o +snd-soc-odroidx2-max98090-objs := odroidx2_max98090.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -71,3 +70,4 @@ obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o +obj-$(CONFIG_SND_SOC_ODROIDX2) += snd-soc-odroidx2-max98090.o diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 68d9303047e8..e1615113fd84 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -19,7 +19,6 @@ #include <sound/soc.h> -#include <mach/dma.h> #include "regs-ac97.h" #include <linux/platform_data/asoc-s3c.h> @@ -39,30 +38,15 @@ struct s3c_ac97_info { }; static struct s3c_ac97_info s3c_ac97; -static struct s3c_dma_client s3c_dma_client_out = { - .name = "AC97 PCMOut" -}; - -static struct s3c_dma_client s3c_dma_client_in = { - .name = "AC97 PCMIn" -}; - -static struct s3c_dma_client s3c_dma_client_micin = { - .name = "AC97 MicIn" -}; - static struct s3c_dma_params s3c_ac97_pcm_out = { - .client = &s3c_dma_client_out, .dma_size = 4, }; static struct s3c_dma_params s3c_ac97_pcm_in = { - .client = &s3c_dma_client_in, .dma_size = 4, }; static struct s3c_dma_params s3c_ac97_mic_in = { - .client = &s3c_dma_client_micin, .dma_size = 4, }; @@ -225,9 +209,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) @@ -253,11 +234,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - if (!dma_data->ops) - dma_data->ops = samsung_dma_get_ops(); - - dma_data->ops->started(dma_data->channel); - return 0; } @@ -265,9 +241,6 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; @@ -287,11 +260,6 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - if (!dma_data->ops) - dma_data->ops = samsung_dma_get_ops(); - - dma_data->ops->started(dma_data->channel); - return 0; } diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c deleted file mode 100644 index d9dc7bcc0336..000000000000 --- a/sound/soc/samsung/dma.c +++ /dev/null @@ -1,454 +0,0 @@ -/* - * dma.c -- ALSA Soc Audio Layer - * - * (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com - * - * Copyright 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben@simtec.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. - */ - -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/module.h> - -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include <asm/dma.h> -#include <mach/hardware.h> -#include <mach/dma.h> - -#include "dma.h" - -#define ST_RUNNING (1<<0) -#define ST_OPENED (1<<1) - -static const struct snd_pcm_hardware dma_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, - .buffer_bytes_max = 128*1024, - .period_bytes_min = PAGE_SIZE, - .period_bytes_max = PAGE_SIZE*2, - .periods_min = 2, - .periods_max = 128, - .fifo_size = 32, -}; - -struct runtime_data { - spinlock_t lock; - int state; - unsigned int dma_loaded; - unsigned int dma_period; - dma_addr_t dma_start; - dma_addr_t dma_pos; - dma_addr_t dma_end; - struct s3c_dma_params *params; -}; - -static void audio_buffdone(void *data); - -/* dma_enqueue - * - * place a dma buffer onto the queue for the dma system - * to handle. - */ -static void dma_enqueue(struct snd_pcm_substream *substream) -{ - struct runtime_data *prtd = substream->runtime->private_data; - dma_addr_t pos = prtd->dma_pos; - unsigned int limit; - struct samsung_dma_prep dma_info; - - pr_debug("Entered %s\n", __func__); - - limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; - - pr_debug("%s: loaded %d, limit %d\n", - __func__, prtd->dma_loaded, limit); - - dma_info.cap = (samsung_dma_has_circular() ? DMA_CYCLIC : DMA_SLAVE); - dma_info.direction = - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); - dma_info.fp = audio_buffdone; - dma_info.fp_param = substream; - dma_info.period = prtd->dma_period; - dma_info.len = prtd->dma_period*limit; - - if (dma_info.cap == DMA_CYCLIC) { - dma_info.buf = pos; - prtd->params->ops->prepare(prtd->params->ch, &dma_info); - prtd->dma_loaded += limit; - return; - } - - while (prtd->dma_loaded < limit) { - pr_debug("dma_loaded: %d\n", prtd->dma_loaded); - - if ((pos + dma_info.period) > prtd->dma_end) { - dma_info.period = prtd->dma_end - pos; - pr_debug("%s: corrected dma len %ld\n", - __func__, dma_info.period); - } - - dma_info.buf = pos; - prtd->params->ops->prepare(prtd->params->ch, &dma_info); - - prtd->dma_loaded++; - pos += prtd->dma_period; - if (pos >= prtd->dma_end) - pos = prtd->dma_start; - } - - prtd->dma_pos = pos; -} - -static void audio_buffdone(void *data) -{ - struct snd_pcm_substream *substream = data; - struct runtime_data *prtd = substream->runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - if (prtd->state & ST_RUNNING) { - prtd->dma_pos += prtd->dma_period; - if (prtd->dma_pos >= prtd->dma_end) - prtd->dma_pos = prtd->dma_start; - - if (substream) - snd_pcm_period_elapsed(substream); - - spin_lock(&prtd->lock); - if (!samsung_dma_has_circular()) { - prtd->dma_loaded--; - dma_enqueue(substream); - } - spin_unlock(&prtd->lock); - } -} - -static int dma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - unsigned long totbytes = params_buffer_bytes(params); - struct s3c_dma_params *dma = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - struct samsung_dma_req req; - struct samsung_dma_config config; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; - - /* this may get called several times by oss emulation - * with different params -HW */ - if (prtd->params == NULL) { - /* prepare DMA */ - prtd->params = dma; - - pr_debug("params %p, client %p, channel %d\n", prtd->params, - prtd->params->client, prtd->params->channel); - - prtd->params->ops = samsung_dma_get_ops(); - - req.cap = (samsung_dma_has_circular() ? - DMA_CYCLIC : DMA_SLAVE); - req.client = prtd->params->client; - config.direction = - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); - config.width = prtd->params->dma_size; - config.fifo = prtd->params->dma_addr; - prtd->params->ch = prtd->params->ops->request( - prtd->params->channel, &req, rtd->cpu_dai->dev, - prtd->params->ch_name); - if (!prtd->params->ch) { - pr_err("Failed to allocate DMA channel\n"); - return -ENXIO; - } - prtd->params->ops->config(prtd->params->ch, &config); - } - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - runtime->dma_bytes = totbytes; - - spin_lock_irq(&prtd->lock); - prtd->dma_loaded = 0; - prtd->dma_period = params_period_bytes(params); - prtd->dma_start = runtime->dma_addr; - prtd->dma_pos = prtd->dma_start; - prtd->dma_end = prtd->dma_start + totbytes; - spin_unlock_irq(&prtd->lock); - - return 0; -} - -static int dma_hw_free(struct snd_pcm_substream *substream) -{ - struct runtime_data *prtd = substream->runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - snd_pcm_set_runtime_buffer(substream, NULL); - - if (prtd->params) { - prtd->params->ops->flush(prtd->params->ch); - prtd->params->ops->release(prtd->params->ch, - prtd->params->client); - prtd->params = NULL; - } - - return 0; -} - -static int dma_prepare(struct snd_pcm_substream *substream) -{ - struct runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!prtd->params) - return 0; - - /* flush the DMA channel */ - prtd->params->ops->flush(prtd->params->ch); - - prtd->dma_loaded = 0; - prtd->dma_pos = prtd->dma_start; - - /* enqueue dma buffers */ - dma_enqueue(substream); - - return ret; -} - -static int dma_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->state |= ST_RUNNING; - prtd->params->ops->trigger(prtd->params->ch); - break; - - case SNDRV_PCM_TRIGGER_STOP: - prtd->state &= ~ST_RUNNING; - prtd->params->ops->stop(prtd->params->ch); - break; - - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static snd_pcm_uframes_t -dma_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd = runtime->private_data; - unsigned long res; - - pr_debug("Entered %s\n", __func__); - - res = prtd->dma_pos - prtd->dma_start; - - pr_debug("Pointer offset: %lu\n", res); - - /* we seem to be getting the odd error from the pcm library due - * to out-of-bounds pointers. this is maybe due to the dma engine - * not having loaded the new values for the channel before being - * called... (todo - fix ) - */ - - if (res >= snd_pcm_lib_buffer_bytes(substream)) { - if (res == snd_pcm_lib_buffer_bytes(substream)) - res = 0; - } - - return bytes_to_frames(substream->runtime, res); -} - -static int dma_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_soc_set_runtime_hwparams(substream, &dma_hardware); - - prtd = kzalloc(sizeof(struct runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - - runtime->private_data = prtd; - return 0; -} - -static int dma_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd = runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - if (!prtd) - pr_debug("dma_close called with prtd == NULL\n"); - - kfree(prtd); - - return 0; -} - -static int dma_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - pr_debug("Entered %s\n", __func__); - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops dma_ops = { - .open = dma_open, - .close = dma_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = dma_hw_params, - .hw_free = dma_hw_free, - .prepare = dma_prepare, - .trigger = dma_trigger, - .pointer = dma_pointer, - .mmap = dma_mmap, -}; - -static int preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = dma_hardware.buffer_bytes_max; - - pr_debug("Entered %s\n", __func__); - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - return 0; -} - -static void dma_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - pr_debug("Entered %s\n", __func__); - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static int dma_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - pr_debug("Entered %s\n", __func__); - - 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 = preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } -out: - return ret; -} - -static struct snd_soc_platform_driver samsung_asoc_platform = { - .ops = &dma_ops, - .pcm_new = dma_new, - .pcm_free = dma_free_dma_buffers, -}; - -void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, - struct s3c_dma_params *playback, - struct s3c_dma_params *capture) -{ - snd_soc_dai_init_dma_data(dai, playback, capture); -} -EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); - -int samsung_asoc_dma_platform_register(struct device *dev) -{ - return devm_snd_soc_register_platform(dev, &samsung_asoc_platform); -} -EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); - -MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>"); -MODULE_DESCRIPTION("Samsung ASoC DMA Driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 070ab0f09609..0e85dcfec023 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -14,17 +14,10 @@ #include <sound/dmaengine_pcm.h> -struct s3c_dma_client { - char *name; -}; - struct s3c_dma_params { - struct s3c_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ dma_addr_t dma_addr; int dma_size; /* Size of the DMA transfer */ - unsigned ch; - struct samsung_dma_ops *ops; char *ch_name; struct snd_dmaengine_dai_dma_data dma_data; }; diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index a0e4e7948909..506f5bf6d082 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -17,6 +17,7 @@ #include <linux/module.h> #include <linux/amba/pl08x.h> +#include <linux/platform_data/dma-s3c24xx.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,8 @@ #ifdef CONFIG_ARCH_S3C64XX #define filter_fn pl08x_filter_id +#elif defined(CONFIG_ARCH_S3C24XX) +#define filter_fn s3c24xx_dma_filter #else #define filter_fn NULL #endif diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 2ac76fa3e742..03eec22f0f46 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -68,6 +68,8 @@ struct i2s_dai { #define DAI_OPENED (1 << 0) /* Dai is opened */ #define DAI_MANAGER (1 << 1) /* Dai is the manager */ unsigned mode; + /* CDCLK pin direction: 0 - input, 1 - output */ + unsigned int cdclk_out:1; /* Driver for this DAI */ struct snd_soc_dai_driver i2s_dai_drv; /* DMA parameters */ @@ -737,6 +739,9 @@ static int i2s_startup(struct snd_pcm_substream *substream, spin_unlock_irqrestore(&lock, flags); + if (!is_opened(other) && i2s->cdclk_out) + i2s_set_sysclk(dai, SAMSUNG_I2S_CDCLK, + 0, SND_SOC_CLOCK_OUT); return 0; } @@ -752,9 +757,13 @@ static void i2s_shutdown(struct snd_pcm_substream *substream, i2s->mode &= ~DAI_OPENED; i2s->mode &= ~DAI_MANAGER; - if (is_opened(other)) + if (is_opened(other)) { other->mode |= DAI_MANAGER; - + } else { + u32 mod = readl(i2s->addr + I2SMOD); + i2s->cdclk_out = !(mod & MOD_CDCLKCON); + other->cdclk_out = i2s->cdclk_out; + } /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; i2s->bfs = 0; @@ -920,11 +929,9 @@ static int i2s_suspend(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); - if (dai->active) { - i2s->suspend_i2smod = readl(i2s->addr + I2SMOD); - i2s->suspend_i2scon = readl(i2s->addr + I2SCON); - i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR); - } + i2s->suspend_i2smod = readl(i2s->addr + I2SMOD); + i2s->suspend_i2scon = readl(i2s->addr + I2SCON); + i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR); return 0; } @@ -933,11 +940,9 @@ static int i2s_resume(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); - if (dai->active) { - writel(i2s->suspend_i2scon, i2s->addr + I2SCON); - writel(i2s->suspend_i2smod, i2s->addr + I2SMOD); - writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR); - } + writel(i2s->suspend_i2scon, i2s->addr + I2SCON); + writel(i2s->suspend_i2smod, i2s->addr + I2SMOD); + writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR); return 0; } @@ -1216,11 +1221,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_playback.dma_addr = regs_base + I2STXD; pri_dai->dma_capture.dma_addr = regs_base + I2SRXD; - pri_dai->dma_playback.client = - (struct s3c_dma_client *)&pri_dai->dma_playback; pri_dai->dma_playback.ch_name = "tx"; - pri_dai->dma_capture.client = - (struct s3c_dma_client *)&pri_dai->dma_capture; pri_dai->dma_capture.ch_name = "rx"; pri_dai->dma_playback.dma_size = 4; pri_dai->dma_capture.dma_size = 4; @@ -1238,8 +1239,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) goto err; } sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; - sec_dai->dma_playback.client = - (struct s3c_dma_client *)&sec_dai->dma_playback; sec_dai->dma_playback.ch_name = "tx-sec"; if (!np) { diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index 8cc5770abb39..db6cefa18017 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -261,10 +261,9 @@ static int idma_mmap(struct snd_pcm_substream *substream, static irqreturn_t iis_irq(int irqno, void *dev_id) { struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id; - u32 iiscon, iisahb, val, addr; + u32 iisahb, val, addr; iisahb = readl(idma.regs + I2SAHB); - iiscon = readl(idma.regs + I2SCON); val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0; diff --git a/sound/soc/samsung/odroidx2_max98090.c b/sound/soc/samsung/odroidx2_max98090.c new file mode 100644 index 000000000000..278edf9e2a87 --- /dev/null +++ b/sound/soc/samsung/odroidx2_max98090.c @@ -0,0 +1,177 @@ +/* + * Copyright (C) 2014 Samsung 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 as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/of.h> +#include <linux/module.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include "i2s.h" + +struct odroidx2_drv_data { + const struct snd_soc_dapm_widget *dapm_widgets; + unsigned int num_dapm_widgets; +}; + +/* The I2S CDCLK output clock frequency for the MAX98090 codec */ +#define MAX98090_MCLK 19200000 + +static int odroidx2_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *cpu_dai = card->rtd[0].cpu_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, MAX98090_MCLK, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set the cpu DAI configuration in order to use CDCLK */ + return snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK, + 0, SND_SOC_CLOCK_OUT); +} + +static const struct snd_soc_dapm_widget odroidx2_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static const struct snd_soc_dapm_widget odroidu3_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), +}; + +static struct snd_soc_dai_link odroidx2_dai[] = { + { + .name = "MAX98090", + .stream_name = "MAX98090 PCM", + .codec_dai_name = "HiFi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + } +}; + +static struct snd_soc_card odroidx2 = { + .owner = THIS_MODULE, + .dai_link = odroidx2_dai, + .num_links = ARRAY_SIZE(odroidx2_dai), + .fully_routed = true, + .late_probe = odroidx2_late_probe, +}; + +struct odroidx2_drv_data odroidx2_drvdata = { + .dapm_widgets = odroidx2_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(odroidx2_dapm_widgets), +}; + +struct odroidx2_drv_data odroidu3_drvdata = { + .dapm_widgets = odroidu3_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(odroidu3_dapm_widgets), +}; + +static const struct of_device_id odroidx2_audio_of_match[] = { + { + .compatible = "samsung,odroidx2-audio", + .data = &odroidx2_drvdata, + }, { + .compatible = "samsung,odroidu3-audio", + .data = &odroidu3_drvdata, + }, + { }, +}; +MODULE_DEVICE_TABLE(of, odroidx2_audio_of_match); + +static int odroidx2_audio_probe(struct platform_device *pdev) +{ + struct device_node *snd_node = pdev->dev.of_node; + struct snd_soc_card *card = &odroidx2; + struct device_node *i2s_node, *codec_node; + struct odroidx2_drv_data *dd; + const struct of_device_id *of_id; + int ret; + + of_id = of_match_node(odroidx2_audio_of_match, snd_node); + dd = (struct odroidx2_drv_data *)of_id->data; + + card->num_dapm_widgets = dd->num_dapm_widgets; + card->dapm_widgets = dd->dapm_widgets; + + card->dev = &pdev->dev; + + ret = snd_soc_of_parse_card_name(card, "samsung,model"); + if (ret < 0) + return ret; + + ret = snd_soc_of_parse_audio_routing(card, "samsung,audio-routing"); + if (ret < 0) + return ret; + + codec_node = of_parse_phandle(snd_node, "samsung,audio-codec", 0); + if (!codec_node) { + dev_err(&pdev->dev, + "Failed parsing samsung,i2s-codec property\n"); + return -EINVAL; + } + + i2s_node = of_parse_phandle(snd_node, "samsung,i2s-controller", 0); + if (!i2s_node) { + dev_err(&pdev->dev, + "Failed parsing samsung,i2s-controller property\n"); + ret = -EINVAL; + goto err_put_codec_n; + } + + odroidx2_dai[0].codec_of_node = codec_node; + odroidx2_dai[0].cpu_of_node = i2s_node; + odroidx2_dai[0].platform_of_node = i2s_node; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + goto err_put_i2s_n; + } + return 0; + +err_put_i2s_n: + of_node_put(i2s_node); +err_put_codec_n: + of_node_put(codec_node); + return ret; +} + +static int odroidx2_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + of_node_put((struct device_node *)odroidx2_dai[0].cpu_of_node); + of_node_put((struct device_node *)odroidx2_dai[0].codec_of_node); + + return 0; +} + +static struct platform_driver odroidx2_audio_driver = { + .driver = { + .name = "odroidx2-audio", + .owner = THIS_MODULE, + .of_match_table = odroidx2_audio_of_match, + .pm = &snd_soc_pm_ops, + }, + .probe = odroidx2_audio_probe, + .remove = odroidx2_audio_remove, +}; +module_platform_driver(odroidx2_audio_driver); + +MODULE_AUTHOR("Chen Zhen <zhen1.chen@samsung.com>"); +MODULE_AUTHOR("Sylwester Nawrocki <s.nawrocki@samsung.com>"); +MODULE_DESCRIPTION("ALSA SoC Odroid X2/U3 Audio Support"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 4c5f97fe45c8..bac034b15a27 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -131,32 +131,20 @@ struct s3c_pcm_info { struct s3c_dma_params *dma_capture; }; -static struct s3c_dma_client s3c_pcm_dma_client_out = { - .name = "PCM Stereo out" -}; - -static struct s3c_dma_client s3c_pcm_dma_client_in = { - .name = "PCM Stereo in" -}; - static struct s3c_dma_params s3c_pcm_stereo_out[] = { [0] = { - .client = &s3c_pcm_dma_client_out, .dma_size = 4, }, [1] = { - .client = &s3c_pcm_dma_client_out, .dma_size = 4, }, }; static struct s3c_dma_params s3c_pcm_stereo_in[] = { [0] = { - .client = &s3c_pcm_dma_client_in, .dma_size = 4, }, [1] = { - .client = &s3c_pcm_dma_client_in, .dma_size = 4, }, }; diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 0ff4bbe23af3..df65c5b494b1 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -22,8 +22,6 @@ #include <sound/soc.h> #include <sound/pcm_params.h> -#include <mach/dma.h> - #include "regs-i2s-v2.h" #include "s3c-i2s-v2.h" #include "dma.h" @@ -392,8 +390,6 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); pr_debug("Entered %s\n", __func__); @@ -424,13 +420,6 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, local_irq_restore(irqs); - /* - * Load the next buffer to DMA to meet the reqirement - * of the auto reload mechanism of S3C24XX. - * This call won't bother S3C64XX. - */ - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - break; case SNDRV_PCM_TRIGGER_STOP: @@ -644,12 +633,6 @@ int s3c_i2sv2_probe(struct snd_soc_dai *dai, /* record our i2s structure for later use in the callbacks */ snd_soc_dai_set_drvdata(dai, i2s); - i2s->regs = ioremap(base, 0x100); - if (i2s->regs == NULL) { - dev_err(dev, "cannot ioremap registers\n"); - return -ENXIO; - } - i2s->iis_pclk = clk_get(dev, "iis"); if (IS_ERR(i2s->iis_pclk)) { dev_err(dev, "failed to get iis_clock\n"); @@ -729,7 +712,7 @@ int s3c_i2sv2_register_component(struct device *dev, int id, struct snd_soc_component_driver *cmp_drv, struct snd_soc_dai_driver *dai_drv) { - struct snd_soc_dai_ops *ops = dai_drv->ops; + struct snd_soc_dai_ops *ops = (struct snd_soc_dai_ops *)dai_drv->ops; ops->trigger = s3c2412_i2s_trigger; if (!ops->hw_params) diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 08c059be9104..27b339c6580e 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -33,25 +33,15 @@ #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" -static struct s3c_dma_client s3c2412_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c_dma_client s3c2412_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { - .client = &s3c2412_dma_client_out, .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, + .ch_name = "tx", .dma_size = 4, }; static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { - .client = &s3c2412_dma_client_in, .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, + .ch_name = "rx", .dma_size = 4, }; @@ -63,6 +53,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) pr_debug("Entered %s\n", __func__); + samsung_asoc_init_dma_data(dai, &s3c2412_i2s_pcm_stereo_out, + &s3c2412_i2s_pcm_stereo_in); + ret = s3c_i2sv2_probe(dai, &s3c2412_i2s, S3C2410_PA_IIS); if (ret) return ret; @@ -70,17 +63,16 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in; s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; - s3c2412_i2s.iis_cclk = clk_get(dai->dev, "i2sclk"); + s3c2412_i2s.iis_cclk = devm_clk_get(dai->dev, "i2sclk"); if (IS_ERR(s3c2412_i2s.iis_cclk)) { pr_err("failed to get i2sclk clock\n"); - iounmap(s3c2412_i2s.regs); return PTR_ERR(s3c2412_i2s.iis_cclk); } /* Set MPLL as the source for IIS CLK */ clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); - clk_enable(s3c2412_i2s.iis_cclk); + clk_prepare_enable(s3c2412_i2s.iis_cclk); s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; @@ -93,9 +85,7 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) static int s3c2412_i2s_remove(struct snd_soc_dai *dai) { - clk_disable(s3c2412_i2s.iis_cclk); - clk_put(s3c2412_i2s.iis_cclk); - iounmap(s3c2412_i2s.regs); + clk_disable_unprepare(s3c2412_i2s.iis_cclk); return 0; } @@ -105,18 +95,10 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct s3c_i2sv2_info *i2s = snd_soc_dai_get_drvdata(cpu_dai); - struct s3c_dma_params *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - iismod = readl(i2s->regs + S3C2412_IISMOD); pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); @@ -169,6 +151,15 @@ static const struct snd_soc_component_driver s3c2412_i2s_component = { static int s3c2412_iis_dev_probe(struct platform_device *pdev) { int ret = 0; + struct resource *res; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + s3c2412_i2s.regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(s3c2412_i2s.regs)) + return PTR_ERR(s3c2412_i2s.regs); + + s3c2412_i2s_pcm_stereo_out.dma_addr = res->start + S3C2412_IISTXD; + s3c2412_i2s_pcm_stereo_in.dma_addr = res->start + S3C2412_IISRXD; ret = s3c_i2sv2_register_component(&pdev->dev, -1, &s3c2412_i2s_component, diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 9aba9fb7df0e..e87d9a2053b8 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -31,25 +31,15 @@ #include "dma.h" #include "s3c24xx-i2s.h" -static struct s3c_dma_client s3c24xx_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c_dma_client s3c24xx_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { - .client = &s3c24xx_dma_client_out, .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .ch_name = "tx", .dma_size = 2, }; static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { - .client = &s3c24xx_dma_client_in, .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .ch_name = "rx", .dma_size = 2, }; @@ -231,18 +221,12 @@ static int s3c24xx_i2s_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; - struct s3c_dma_params *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c24xx_i2s_pcm_stereo_out; - else - dma_data = &s3c24xx_i2s_pcm_stereo_in; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); + dma_data = snd_soc_dai_get_dma_data(dai, substream); /* Working copies of register */ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -251,11 +235,11 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_width(params)) { case 8: iismod &= ~S3C2410_IISMOD_16BIT; - dma_data->dma_size = 1; + dma_data->addr_width = 1; break; case 16: iismod |= S3C2410_IISMOD_16BIT; - dma_data->dma_size = 2; + dma_data->addr_width = 2; break; default: return -EINVAL; @@ -270,8 +254,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { int ret = 0; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(dai, substream); pr_debug("Entered %s\n", __func__); @@ -290,7 +272,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, else s3c24xx_snd_txctrl(1); - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: @@ -380,17 +361,15 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) { pr_debug("Entered %s\n", __func__); - s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); - if (s3c24xx_i2s.regs == NULL) - return -ENXIO; + samsung_asoc_init_dma_data(dai, &s3c24xx_i2s_pcm_stereo_out, + &s3c24xx_i2s_pcm_stereo_in); - s3c24xx_i2s.iis_clk = clk_get(dai->dev, "iis"); + s3c24xx_i2s.iis_clk = devm_clk_get(dai->dev, "iis"); if (IS_ERR(s3c24xx_i2s.iis_clk)) { pr_err("failed to get iis_clock\n"); - iounmap(s3c24xx_i2s.regs); return PTR_ERR(s3c24xx_i2s.iis_clk); } - clk_enable(s3c24xx_i2s.iis_clk); + clk_prepare_enable(s3c24xx_i2s.iis_clk); /* Configure the I2S pins (GPE0...GPE4) in correct mode */ s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), @@ -414,7 +393,7 @@ static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai) s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); s3c24xx_i2s.iispsr = readl(s3c24xx_i2s.regs + S3C2410_IISPSR); - clk_disable(s3c24xx_i2s.iis_clk); + clk_disable_unprepare(s3c24xx_i2s.iis_clk); return 0; } @@ -422,7 +401,7 @@ static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai) static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) { pr_debug("Entered %s\n", __func__); - clk_enable(s3c24xx_i2s.iis_clk); + clk_prepare_enable(s3c24xx_i2s.iis_clk); writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); writel(s3c24xx_i2s.iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -474,6 +453,19 @@ static const struct snd_soc_component_driver s3c24xx_i2s_component = { static int s3c24xx_iis_dev_probe(struct platform_device *pdev) { int ret = 0; + struct resource *res; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "Can't get IO resource.\n"); + return -ENOENT; + } + s3c24xx_i2s.regs = devm_ioremap_resource(&pdev->dev, res); + if (s3c24xx_i2s.regs == NULL) + return -ENXIO; + + s3c24xx_i2s_pcm_stereo_out.dma_addr = res->start + S3C2410_IISFIFO; + s3c24xx_i2s_pcm_stereo_in.dma_addr = res->start + S3C2410_IISFIFO; ret = devm_snd_soc_register_component(&pdev->dev, &s3c24xx_i2s_component, &s3c24xx_i2s_dai, 1); diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index e119aaa91c28..63d079303561 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -25,7 +25,7 @@ * o '0' means 'OFF' * o 'X' means 'Don't care' * - * SMDK6410, SMDK6440, SMDK6450 Base B/D: CFG1-0000, CFG2-1111 + * SMDK6410 Base B/D: CFG1-0000, CFG2-1111 * SMDKC110, SMDKV210: CFGB11-100100, CFGB12-0000 */ diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 014c177840ba..0acf5d0eed53 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -92,6 +92,9 @@ static int snow_probe(struct platform_device *pdev) card->dev = &pdev->dev; + /* Update card-name if provided through DT, else use default name */ + snd_soc_of_parse_card_name(card, "samsung,model"); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); @@ -103,6 +106,7 @@ static int snow_probe(struct platform_device *pdev) static const struct of_device_id snow_of_match[] = { { .compatible = "google,snow-audio-max98090", }, + { .compatible = "google,snow-audio-max98091", }, { .compatible = "google,snow-audio-max98095", }, {}, }; diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index d9ffc48fce5e..d7d2e208f486 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -93,10 +93,6 @@ struct samsung_spdif_info { struct s3c_dma_params *dma_playback; }; -static struct s3c_dma_client spdif_dma_client_out = { - .name = "S/PDIF Stereo out", -}; - static struct s3c_dma_params spdif_stereo_out; static struct samsung_spdif_info spdif_info; @@ -435,7 +431,6 @@ static int spdif_probe(struct platform_device *pdev) } spdif_stereo_out.dma_size = 2; - spdif_stereo_out.client = &spdif_dma_client_out; spdif_stereo_out.dma_addr = mem_res->start + DATA_OUTBUF; spdif_stereo_out.channel = dma_res->start; diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index b43fdf0d08af..80245b6eebd6 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -37,7 +37,7 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" select SND_SIMPLE_CARD - select REGMAP + select REGMAP_MMIO help This option enables R-Car SUR/SCU/SSIU/SSI sound support diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 710a079a7377..c76344350e44 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -232,11 +232,7 @@ struct fsi_stream { * these are for DMAEngine */ struct dma_chan *chan; - struct work_struct work; - dma_addr_t dma; int dma_id; - int loop_cnt; - int additional_pos; }; struct fsi_clk { @@ -264,12 +260,12 @@ struct fsi_priv { u32 fmt; int chan_num:16; - int clk_master:1; - int clk_cpg:1; - int spdif:1; - int enable_stream:1; - int bit_clk_inv:1; - int lr_clk_inv:1; + unsigned int clk_master:1; + unsigned int clk_cpg:1; + unsigned int spdif:1; + unsigned int enable_stream:1; + unsigned int bit_clk_inv:1; + unsigned int lr_clk_inv:1; }; struct fsi_stream_handler { @@ -1042,6 +1038,26 @@ static int fsi_clk_set_rate_cpg(struct device *dev, return ret; } +static void fsi_pointer_update(struct fsi_stream *io, int size) +{ + io->buff_sample_pos += size; + + if (io->buff_sample_pos >= + io->period_samples * (io->period_pos + 1)) { + struct snd_pcm_substream *substream = io->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + + io->period_pos++; + + if (io->period_pos >= runtime->periods) { + io->buff_sample_pos = 0; + io->period_pos = 0; + } + + snd_pcm_period_elapsed(substream); + } +} + /* * pio data transfer handler */ @@ -1108,31 +1124,11 @@ static int fsi_pio_transfer(struct fsi_priv *fsi, struct fsi_stream *io, void (*run32)(struct fsi_priv *fsi, u8 *buf, int samples), int samples) { - struct snd_pcm_runtime *runtime; - struct snd_pcm_substream *substream; u8 *buf; - int over_period; if (!fsi_stream_is_working(fsi, io)) return -EINVAL; - over_period = 0; - substream = io->substream; - runtime = substream->runtime; - - /* FSI FIFO has limit. - * So, this driver can not send periods data at a time - */ - if (io->buff_sample_pos >= - io->period_samples * (io->period_pos + 1)) { - - over_period = 1; - io->period_pos = (io->period_pos + 1) % runtime->periods; - - if (0 == io->period_pos) - io->buff_sample_pos = 0; - } - buf = fsi_pio_get_area(fsi, io); switch (io->sample_width) { @@ -1146,11 +1142,7 @@ static int fsi_pio_transfer(struct fsi_priv *fsi, struct fsi_stream *io, return -EINVAL; } - /* update buff_sample_pos */ - io->buff_sample_pos += samples; - - if (over_period) - snd_pcm_period_elapsed(substream); + fsi_pointer_update(io, samples); return 0; } @@ -1279,11 +1271,6 @@ static irqreturn_t fsi_interrupt(int irq, void *data) */ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io) { - struct snd_pcm_runtime *runtime = io->substream->runtime; - struct snd_soc_dai *dai = fsi_get_dai(io->substream); - enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? - DMA_TO_DEVICE : DMA_FROM_DEVICE; - /* * 24bit data : 24bit bus / package in back * 16bit data : 16bit bus / stream mode @@ -1291,107 +1278,48 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io) io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) | BUSOP_SET(16, PACKAGE_16BITBUS_STREAM); - io->loop_cnt = 2; /* push 1st, 2nd period first, then 3rd, 4th... */ - io->additional_pos = 0; - io->dma = dma_map_single(dai->dev, runtime->dma_area, - snd_pcm_lib_buffer_bytes(io->substream), dir); return 0; } -static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io) -{ - struct snd_soc_dai *dai = fsi_get_dai(io->substream); - enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? - DMA_TO_DEVICE : DMA_FROM_DEVICE; - - dma_unmap_single(dai->dev, io->dma, - snd_pcm_lib_buffer_bytes(io->substream), dir); - return 0; -} - -static dma_addr_t fsi_dma_get_area(struct fsi_stream *io, int additional) -{ - struct snd_pcm_runtime *runtime = io->substream->runtime; - int period = io->period_pos + additional; - - if (period >= runtime->periods) - period = 0; - - return io->dma + samples_to_bytes(runtime, period * io->period_samples); -} - static void fsi_dma_complete(void *data) { struct fsi_stream *io = (struct fsi_stream *)data; struct fsi_priv *fsi = fsi_stream_to_priv(io); - struct snd_pcm_runtime *runtime = io->substream->runtime; - struct snd_soc_dai *dai = fsi_get_dai(io->substream); - enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? - DMA_TO_DEVICE : DMA_FROM_DEVICE; - dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io, 0), - samples_to_bytes(runtime, io->period_samples), dir); - - io->buff_sample_pos += io->period_samples; - io->period_pos++; - - if (io->period_pos >= runtime->periods) { - io->period_pos = 0; - io->buff_sample_pos = 0; - } + fsi_pointer_update(io, io->period_samples); fsi_count_fifo_err(fsi); - fsi_stream_transfer(io); - - snd_pcm_period_elapsed(io->substream); } -static void fsi_dma_do_work(struct work_struct *work) +static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) { - struct fsi_stream *io = container_of(work, struct fsi_stream, work); - struct fsi_priv *fsi = fsi_stream_to_priv(io); - struct snd_soc_dai *dai; + struct snd_soc_dai *dai = fsi_get_dai(io->substream); + struct snd_pcm_substream *substream = io->substream; struct dma_async_tx_descriptor *desc; - struct snd_pcm_runtime *runtime; - enum dma_data_direction dir; int is_play = fsi_stream_is_play(fsi, io); - int len, i; - dma_addr_t buf; - - if (!fsi_stream_is_working(fsi, io)) - return; - - dai = fsi_get_dai(io->substream); - runtime = io->substream->runtime; - dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; - len = samples_to_bytes(runtime, io->period_samples); - - for (i = 0; i < io->loop_cnt; i++) { - buf = fsi_dma_get_area(io, io->additional_pos); - - dma_sync_single_for_device(dai->dev, buf, len, dir); - - desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - if (!desc) { - dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); - return; - } - - desc->callback = fsi_dma_complete; - desc->callback_param = io; - - if (dmaengine_submit(desc) < 0) { - dev_err(dai->dev, "tx_submit() fail\n"); - return; - } + enum dma_data_direction dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; + int ret = -EIO; + + desc = dmaengine_prep_dma_cyclic(io->chan, + substream->runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), + dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(dai->dev, "dmaengine_prep_dma_cyclic() fail\n"); + goto fsi_dma_transfer_err; + } - dma_async_issue_pending(io->chan); + desc->callback = fsi_dma_complete; + desc->callback_param = io; - io->additional_pos = 1; + if (dmaengine_submit(desc) < 0) { + dev_err(dai->dev, "tx_submit() fail\n"); + goto fsi_dma_transfer_err; } - io->loop_cnt = 1; + dma_async_issue_pending(io->chan); /* * FIXME @@ -1408,13 +1336,11 @@ static void fsi_dma_do_work(struct work_struct *work) fsi_reg_write(fsi, DIFF_ST, 0); } } -} -static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) -{ - schedule_work(&io->work); + ret = 0; - return 0; +fsi_dma_transfer_err: + return ret; } static int fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, @@ -1475,15 +1401,11 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct dev return fsi_stream_probe(fsi, dev); } - INIT_WORK(&io->work, fsi_dma_do_work); - return 0; } static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io) { - cancel_work_sync(&io->work); - fsi_stream_stop(fsi, io); if (io->chan) @@ -1495,7 +1417,6 @@ static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io) static struct fsi_stream_handler fsi_dma_push_handler = { .init = fsi_dma_init, - .quit = fsi_dma_quit, .probe = fsi_dma_probe, .transfer = fsi_dma_transfer, .remove = fsi_dma_remove, @@ -1657,9 +1578,9 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, if (!ret) ret = fsi_hw_startup(fsi, io, dai->dev); if (!ret) - ret = fsi_stream_transfer(io); + ret = fsi_stream_start(fsi, io); if (!ret) - fsi_stream_start(fsi, io); + ret = fsi_stream_transfer(io); break; case SNDRV_PCM_TRIGGER_STOP: if (!ret) @@ -1850,16 +1771,10 @@ static void fsi_pcm_free(struct snd_pcm *pcm) static int fsi_pcm_new(struct snd_soc_pcm_runtime *rtd) { - struct snd_pcm *pcm = rtd->pcm; - - /* - * dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel - * in MMAP mode (i.e. aplay -M) - */ return snd_pcm_lib_preallocate_pages_for_all( - pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + rtd->pcm, + SNDRV_DMA_TYPE_DEV, + rtd->card->snd_card->dev, PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); } diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4e86265f625c..19f78963e8b9 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -138,6 +138,17 @@ char *rsnd_mod_name(struct rsnd_mod *mod) return mod->ops->name; } +char *rsnd_mod_dma_name(struct rsnd_mod *mod) +{ + if (!mod || !mod->ops) + return "unknown"; + + if (!mod->ops->dma_name) + return mod->ops->name; + + return mod->ops->dma_name(mod); +} + void rsnd_mod_init(struct rsnd_priv *priv, struct rsnd_mod *mod, struct rsnd_mod_ops *ops, @@ -153,26 +164,8 @@ void rsnd_mod_init(struct rsnd_priv *priv, /* * rsnd_dma functions */ -static void __rsnd_dma_start(struct rsnd_dma *dma); -static void rsnd_dma_continue(struct rsnd_dma *dma) -{ - /* push next A or B plane */ - dma->submit_loop = 1; - schedule_work(&dma->work); -} - -void rsnd_dma_start(struct rsnd_dma *dma) -{ - /* push both A and B plane*/ - dma->offset = 0; - dma->submit_loop = 2; - __rsnd_dma_start(dma); -} - void rsnd_dma_stop(struct rsnd_dma *dma) { - dma->submit_loop = 0; - cancel_work_sync(&dma->work); dmaengine_terminate_all(dma->chan); } @@ -180,11 +173,7 @@ static void rsnd_dma_complete(void *data) { struct rsnd_dma *dma = (struct rsnd_dma *)data; struct rsnd_mod *mod = rsnd_dma_to_mod(dma); - struct rsnd_priv *priv = rsnd_mod_to_priv(rsnd_dma_to_mod(dma)); struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - unsigned long flags; - - rsnd_lock(priv, flags); /* * Renesas sound Gen1 needs 1 DMAC, @@ -197,57 +186,41 @@ static void rsnd_dma_complete(void *data) * rsnd_dai_pointer_update() will be called twice, * ant it will breaks io->byte_pos */ - if (dma->submit_loop) - rsnd_dma_continue(dma); - - rsnd_unlock(priv, flags); rsnd_dai_pointer_update(io, io->byte_per_period); } -static void __rsnd_dma_start(struct rsnd_dma *dma) +void rsnd_dma_start(struct rsnd_dma *dma) { struct rsnd_mod *mod = rsnd_dma_to_mod(dma); struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct snd_pcm_substream *substream = io->substream; struct device *dev = rsnd_priv_to_dev(priv); struct dma_async_tx_descriptor *desc; - dma_addr_t buf; - size_t len = io->byte_per_period; - int i; - for (i = 0; i < dma->submit_loop; i++) { - - buf = runtime->dma_addr + - rsnd_dai_pointer_offset(io, dma->offset + len); - dma->offset = len; - - desc = dmaengine_prep_slave_single( - dma->chan, buf, len, dma->dir, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - if (!desc) { - dev_err(dev, "dmaengine_prep_slave_sg() fail\n"); - return; - } + desc = dmaengine_prep_dma_cyclic(dma->chan, + (dma->addr) ? dma->addr : + substream->runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), + dma->dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - desc->callback = rsnd_dma_complete; - desc->callback_param = dma; + if (!desc) { + dev_err(dev, "dmaengine_prep_slave_sg() fail\n"); + return; + } - if (dmaengine_submit(desc) < 0) { - dev_err(dev, "dmaengine_submit() fail\n"); - return; - } + desc->callback = rsnd_dma_complete; + desc->callback_param = dma; - dma_async_issue_pending(dma->chan); + if (dmaengine_submit(desc) < 0) { + dev_err(dev, "dmaengine_submit() fail\n"); + return; } -} - -static void rsnd_dma_do_work(struct work_struct *work) -{ - struct rsnd_dma *dma = container_of(work, struct rsnd_dma, work); - __rsnd_dma_start(dma); + dma_async_issue_pending(dma->chan); } int rsnd_dma_available(struct rsnd_dma *dma) @@ -261,14 +234,27 @@ static int _rsnd_dma_of_name(char *dma_name, struct rsnd_mod *mod) { if (mod) return snprintf(dma_name, DMA_NAME_SIZE / 2, "%s%d", - rsnd_mod_name(mod), rsnd_mod_id(mod)); + rsnd_mod_dma_name(mod), rsnd_mod_id(mod)); else return snprintf(dma_name, DMA_NAME_SIZE / 2, "mem"); } -static void rsnd_dma_of_name(struct rsnd_dma *dma, - int is_play, char *dma_name) +static void rsnd_dma_of_name(struct rsnd_mod *mod_from, + struct rsnd_mod *mod_to, + char *dma_name) +{ + int index = 0; + + index = _rsnd_dma_of_name(dma_name + index, mod_from); + *(dma_name + index++) = '_'; + index = _rsnd_dma_of_name(dma_name + index, mod_to); +} + +static void rsnd_dma_of_path(struct rsnd_dma *dma, + int is_play, + struct rsnd_mod **mod_from, + struct rsnd_mod **mod_to) { struct rsnd_mod *this = rsnd_dma_to_mod(dma); struct rsnd_dai_stream *io = rsnd_mod_to_io(this); @@ -276,7 +262,6 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma, struct rsnd_mod *src = rsnd_io_to_mod_src(io); struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io); struct rsnd_mod *mod[MOD_MAX]; - struct rsnd_mod *src_mod, *dst_mod; int i, index; @@ -297,31 +282,34 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma, for (i = 1; i < MOD_MAX; i++) { if (!src) { mod[i] = ssi; - break; } else if (!dvc) { mod[i] = src; src = NULL; } else { - mod[i] = dvc; + if ((!is_play) && (this == src)) + this = dvc; + + mod[i] = (is_play) ? src : dvc; + i++; + mod[i] = (is_play) ? dvc : src; + src = NULL; dvc = NULL; } if (mod[i] == this) index = i; + + if (mod[i] == ssi) + break; } if (is_play) { - src_mod = mod[index - 1]; - dst_mod = mod[index]; + *mod_from = mod[index - 1]; + *mod_to = mod[index]; } else { - src_mod = mod[index]; - dst_mod = mod[index - 1]; + *mod_from = mod[index]; + *mod_to = mod[index - 1]; } - - index = 0; - index = _rsnd_dma_of_name(dma_name + index, src_mod); - *(dma_name + index++) = '_'; - index = _rsnd_dma_of_name(dma_name + index, dst_mod); } int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, @@ -329,6 +317,8 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, { struct device *dev = rsnd_priv_to_dev(priv); struct dma_slave_config cfg; + struct rsnd_mod *mod_from; + struct rsnd_mod *mod_to; char dma_name[DMA_NAME_SIZE]; dma_cap_mask_t mask; int ret; @@ -341,13 +331,18 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); - if (dev->of_node) - rsnd_dma_of_name(dma, is_play, dma_name); - else - snprintf(dma_name, DMA_NAME_SIZE, - is_play ? "tx" : "rx"); + rsnd_dma_of_path(dma, is_play, &mod_from, &mod_to); + rsnd_dma_of_name(mod_from, mod_to, dma_name); - dev_dbg(dev, "dma name : %s\n", dma_name); + cfg.slave_id = id; + cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + cfg.src_addr = rsnd_gen_dma_addr(priv, mod_from, is_play, 1); + cfg.dst_addr = rsnd_gen_dma_addr(priv, mod_to, is_play, 0); + cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + + dev_dbg(dev, "dma : %s %pad -> %pad\n", + dma_name, &cfg.src_addr, &cfg.dst_addr); dma->chan = dma_request_slave_channel_compat(mask, shdma_chan_filter, (void *)id, dev, @@ -357,14 +352,12 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, return -EIO; } - rsnd_gen_dma_addr(priv, dma, &cfg, is_play, id); - ret = dmaengine_slave_config(dma->chan, &cfg); if (ret < 0) goto rsnd_dma_init_err; - dma->dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; - INIT_WORK(&dma->work, rsnd_dma_do_work); + dma->addr = is_play ? cfg.src_addr : cfg.dst_addr; + dma->dir = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; return 0; @@ -631,40 +624,41 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - /* set clock inversion */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_IF: - rdai->bit_clk_inv = 0; - rdai->frm_clk_inv = 1; - break; - case SND_SOC_DAIFMT_IB_NF: - rdai->bit_clk_inv = 1; - rdai->frm_clk_inv = 0; - break; - case SND_SOC_DAIFMT_IB_IF: - rdai->bit_clk_inv = 1; - rdai->frm_clk_inv = 1; - break; - case SND_SOC_DAIFMT_NB_NF: - default: - rdai->bit_clk_inv = 0; - rdai->frm_clk_inv = 0; - break; - } - /* set format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: rdai->sys_delay = 0; rdai->data_alignment = 0; + rdai->frm_clk_inv = 0; break; case SND_SOC_DAIFMT_LEFT_J: rdai->sys_delay = 1; rdai->data_alignment = 0; + rdai->frm_clk_inv = 1; break; case SND_SOC_DAIFMT_RIGHT_J: rdai->sys_delay = 1; rdai->data_alignment = 1; + rdai->frm_clk_inv = 1; + break; + } + + /* set clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + rdai->bit_clk_inv = rdai->bit_clk_inv; + rdai->frm_clk_inv = !rdai->frm_clk_inv; + break; + case SND_SOC_DAIFMT_IB_NF: + rdai->bit_clk_inv = !rdai->bit_clk_inv; + rdai->frm_clk_inv = rdai->frm_clk_inv; + break; + case SND_SOC_DAIFMT_IB_IF: + rdai->bit_clk_inv = !rdai->bit_clk_inv; + rdai->frm_clk_inv = !rdai->frm_clk_inv; + break; + case SND_SOC_DAIFMT_NB_NF: + default: break; } @@ -734,12 +728,13 @@ static void rsnd_of_parse_dai(struct platform_device *pdev, struct device_node *dai_node, *dai_np; struct device_node *ssi_node, *ssi_np; struct device_node *src_node, *src_np; + struct device_node *dvc_node, *dvc_np; struct device_node *playback, *capture; struct rsnd_dai_platform_info *dai_info; struct rcar_snd_info *info = rsnd_priv_to_info(priv); struct device *dev = &pdev->dev; int nr, i; - int dai_i, ssi_i, src_i; + int dai_i, ssi_i, src_i, dvc_i; if (!of_data) return; @@ -765,6 +760,7 @@ static void rsnd_of_parse_dai(struct platform_device *pdev, ssi_node = of_get_child_by_name(dev->of_node, "rcar_sound,ssi"); src_node = of_get_child_by_name(dev->of_node, "rcar_sound,src"); + dvc_node = of_get_child_by_name(dev->of_node, "rcar_sound,dvc"); #define mod_parse(name) \ if (name##_node) { \ @@ -800,6 +796,7 @@ if (name##_node) { \ mod_parse(ssi); mod_parse(src); + mod_parse(dvc); if (playback) of_node_put(playback); @@ -948,19 +945,17 @@ static struct snd_pcm_ops rsnd_pcm_ops = { static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) { - struct rsnd_priv *priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct rsnd_dai *rdai; - int i, ret; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + int ret; - for_each_rsnd_dai(rdai, priv, i) { - ret = rsnd_dai_call(pcm_new, &rdai->playback, rdai, rtd); - if (ret) - return ret; + ret = rsnd_dai_call(pcm_new, &rdai->playback, rdai, rtd); + if (ret) + return ret; - ret = rsnd_dai_call(pcm_new, &rdai->capture, rdai, rtd); - if (ret) - return ret; - } + ret = rsnd_dai_call(pcm_new, &rdai->capture, rdai, rtd); + if (ret) + return ret; return snd_pcm_lib_preallocate_pages_for_all( rtd->pcm, @@ -1047,11 +1042,11 @@ static int rsnd_probe(struct platform_device *pdev) for_each_rsnd_dai(rdai, priv, i) { ret = rsnd_dai_call(probe, &rdai->playback, rdai); if (ret) - return ret; + goto exit_snd_probe; ret = rsnd_dai_call(probe, &rdai->capture, rdai); if (ret) - return ret; + goto exit_snd_probe; } /* @@ -1079,6 +1074,11 @@ static int rsnd_probe(struct platform_device *pdev) exit_snd_soc: snd_soc_unregister_platform(dev); +exit_snd_probe: + for_each_rsnd_dai(rdai, priv, i) { + rsnd_dai_call(remove, &rdai->playback, rdai); + rsnd_dai_call(remove, &rdai->capture, rdai); + } return ret; } @@ -1087,21 +1087,16 @@ static int rsnd_remove(struct platform_device *pdev) { struct rsnd_priv *priv = dev_get_drvdata(&pdev->dev); struct rsnd_dai *rdai; - int ret, i; + int ret = 0, i; pm_runtime_disable(&pdev->dev); for_each_rsnd_dai(rdai, priv, i) { - ret = rsnd_dai_call(remove, &rdai->playback, rdai); - if (ret) - return ret; - - ret = rsnd_dai_call(remove, &rdai->capture, rdai); - if (ret) - return ret; + ret |= rsnd_dai_call(remove, &rdai->playback, rdai); + ret |= rsnd_dai_call(remove, &rdai->capture, rdai); } - return 0; + return ret; } static struct platform_driver rsnd_driver = { diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index ed0007006899..3f443930c2b1 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -20,7 +20,8 @@ struct rsnd_dvc { struct rsnd_dvc_platform_info *info; /* rcar_snd.h */ struct rsnd_mod mod; struct clk *clk; - long volume[RSND_DVC_VOLUME_NUM]; + u8 volume[RSND_DVC_VOLUME_NUM]; + u8 mute[RSND_DVC_VOLUME_NUM]; }; #define rsnd_mod_to_dvc(_mod) \ @@ -37,13 +38,18 @@ static void rsnd_dvc_volume_update(struct rsnd_mod *mod) struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); u32 max = (0x00800000 - 1); u32 vol[RSND_DVC_VOLUME_NUM]; + u32 mute = 0; int i; - for (i = 0; i < RSND_DVC_VOLUME_NUM; i++) + for (i = 0; i < RSND_DVC_VOLUME_NUM; i++) { vol[i] = max / RSND_DVC_VOLUME_MAX * dvc->volume[i]; + mute |= (!!dvc->mute[i]) << i; + } rsnd_mod_write(mod, DVC_VOL0R, vol[0]); rsnd_mod_write(mod, DVC_VOL1R, vol[1]); + + rsnd_mod_write(mod, DVC_ZCMCR, mute); } static int rsnd_dvc_probe_gen2(struct rsnd_mod *mod, @@ -96,8 +102,8 @@ static int rsnd_dvc_init(struct rsnd_mod *dvc_mod, rsnd_mod_write(dvc_mod, DVC_ADINR, rsnd_get_adinr(dvc_mod)); - /* enable Volume */ - rsnd_mod_write(dvc_mod, DVC_DVUCR, 0x100); + /* enable Volume / Mute */ + rsnd_mod_write(dvc_mod, DVC_DVUCR, 0x101); /* ch0/ch1 Volume */ rsnd_dvc_volume_update(dvc_mod); @@ -140,10 +146,20 @@ static int rsnd_dvc_stop(struct rsnd_mod *mod, static int rsnd_dvc_volume_info(struct snd_kcontrol *kctrl, struct snd_ctl_elem_info *uinfo) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + struct rsnd_mod *mod = snd_kcontrol_chip(kctrl); + struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); + u8 *val = (u8 *)kctrl->private_value; + uinfo->count = RSND_DVC_VOLUME_NUM; uinfo->value.integer.min = 0; - uinfo->value.integer.max = RSND_DVC_VOLUME_MAX; + + if (val == dvc->volume) { + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.max = RSND_DVC_VOLUME_MAX; + } else { + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->value.integer.max = 1; + } return 0; } @@ -151,12 +167,11 @@ static int rsnd_dvc_volume_info(struct snd_kcontrol *kctrl, static int rsnd_dvc_volume_get(struct snd_kcontrol *kctrl, struct snd_ctl_elem_value *ucontrol) { - struct rsnd_mod *mod = snd_kcontrol_chip(kctrl); - struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); + u8 *val = (u8 *)kctrl->private_value; int i; for (i = 0; i < RSND_DVC_VOLUME_NUM; i++) - ucontrol->value.integer.value[i] = dvc->volume[i]; + ucontrol->value.integer.value[i] = val[i]; return 0; } @@ -165,51 +180,38 @@ static int rsnd_dvc_volume_put(struct snd_kcontrol *kctrl, struct snd_ctl_elem_value *ucontrol) { struct rsnd_mod *mod = snd_kcontrol_chip(kctrl); - struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); + u8 *val = (u8 *)kctrl->private_value; int i, change = 0; for (i = 0; i < RSND_DVC_VOLUME_NUM; i++) { - if (ucontrol->value.integer.value[i] < 0 || - ucontrol->value.integer.value[i] > RSND_DVC_VOLUME_MAX) - return -EINVAL; - - change |= (ucontrol->value.integer.value[i] != dvc->volume[i]); + change |= (ucontrol->value.integer.value[i] != val[i]); + val[i] = ucontrol->value.integer.value[i]; } - if (change) { - for (i = 0; i < RSND_DVC_VOLUME_NUM; i++) - dvc->volume[i] = ucontrol->value.integer.value[i]; - + if (change) rsnd_dvc_volume_update(mod); - } return change; } -static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, - struct rsnd_dai *rdai, - struct snd_soc_pcm_runtime *rtd) +static int __rsnd_dvc_pcm_new(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct snd_soc_pcm_runtime *rtd, + const unsigned char *name, + u8 *private) { - struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct device *dev = rsnd_priv_to_dev(priv); struct snd_card *card = rtd->card->snd_card; struct snd_kcontrol *kctrl; - static struct snd_kcontrol_new knew = { + struct snd_kcontrol_new knew = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Playback Volume", + .name = name, .info = rsnd_dvc_volume_info, .get = rsnd_dvc_volume_get, .put = rsnd_dvc_volume_put, + .private_value = (unsigned long)private, }; int ret; - if (!rsnd_dai_is_play(rdai, io)) { - dev_err(dev, "DVC%d is connected to Capture DAI\n", - rsnd_mod_id(mod)); - return -EINVAL; - } - kctrl = snd_ctl_new1(&knew, mod); if (!kctrl) return -ENOMEM; @@ -221,6 +223,33 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, return 0; } +static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct snd_soc_pcm_runtime *rtd) +{ + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); + int ret; + + /* Volume */ + ret = __rsnd_dvc_pcm_new(mod, rdai, rtd, + rsnd_dai_is_play(rdai, io) ? + "DVC Out Playback Volume" : "DVC In Capture Volume", + dvc->volume); + if (ret < 0) + return ret; + + /* Mute */ + ret = __rsnd_dvc_pcm_new(mod, rdai, rtd, + rsnd_dai_is_play(rdai, io) ? + "DVC Out Mute Switch" : "DVC In Mute Switch", + dvc->mute); + if (ret < 0) + return ret; + + return 0; +} + static struct rsnd_mod_ops rsnd_dvc_ops = { .name = DVC_NAME, .probe = rsnd_dvc_probe_gen2, @@ -239,6 +268,42 @@ struct rsnd_mod *rsnd_dvc_mod_get(struct rsnd_priv *priv, int id) return &((struct rsnd_dvc *)(priv->dvc) + id)->mod; } +static void rsnd_of_parse_dvc(struct platform_device *pdev, + const struct rsnd_of_data *of_data, + struct rsnd_priv *priv) +{ + struct device_node *node; + struct rsnd_dvc_platform_info *dvc_info; + struct rcar_snd_info *info = rsnd_priv_to_info(priv); + struct device *dev = &pdev->dev; + int nr; + + if (!of_data) + return; + + node = of_get_child_by_name(dev->of_node, "rcar_sound,dvc"); + if (!node) + return; + + nr = of_get_child_count(node); + if (!nr) + goto rsnd_of_parse_dvc_end; + + dvc_info = devm_kzalloc(dev, + sizeof(struct rsnd_dvc_platform_info) * nr, + GFP_KERNEL); + if (!dvc_info) { + dev_err(dev, "dvc info allocation error\n"); + goto rsnd_of_parse_dvc_end; + } + + info->dvc_info = dvc_info; + info->dvc_info_nr = nr; + +rsnd_of_parse_dvc_end: + of_node_put(node); +} + int rsnd_dvc_probe(struct platform_device *pdev, const struct rsnd_of_data *of_data, struct rsnd_priv *priv) @@ -250,6 +315,8 @@ int rsnd_dvc_probe(struct platform_device *pdev, char name[RSND_DVC_NAME_SIZE]; int i, nr; + rsnd_of_parse_dvc(pdev, of_data, priv); + nr = info->dvc_info_nr; if (!nr) return 0; diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 1dd2b7d38c2c..3fdf3be7b99a 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -15,63 +15,35 @@ struct rsnd_gen { struct rsnd_gen_ops *ops; - struct regmap *regmap; + struct regmap *regmap[RSND_BASE_MAX]; struct regmap_field *regs[RSND_REG_MAX]; }; #define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen) -#define RSND_REG_SET(gen, id, reg_id, offset, _id_offset, _id_size) \ - [id] = { \ - .reg = (unsigned int)gen->base[reg_id] + offset, \ - .lsb = 0, \ - .msb = 31, \ - .id_size = _id_size, \ - .id_offset = _id_offset, \ - } - -/* - * basic function - */ -static int rsnd_regmap_write32(void *context, const void *_data, size_t count) -{ - struct rsnd_priv *priv = context; - struct device *dev = rsnd_priv_to_dev(priv); - u32 *data = (u32 *)_data; - u32 val = data[1]; - void __iomem *reg = (void *)data[0]; - - iowrite32(val, reg); - - dev_dbg(dev, "w %p : %08x\n", reg, val); - - return 0; -} - -static int rsnd_regmap_read32(void *context, - const void *_data, size_t reg_size, - void *_val, size_t val_size) -{ - struct rsnd_priv *priv = context; - struct device *dev = rsnd_priv_to_dev(priv); - u32 *data = (u32 *)_data; - u32 *val = (u32 *)_val; - void __iomem *reg = (void *)data[0]; - - *val = ioread32(reg); - - dev_dbg(dev, "r %p : %08x\n", reg, *val); +struct rsnd_regmap_field_conf { + int idx; + unsigned int reg_offset; + unsigned int id_offset; +}; - return 0; +#define RSND_REG_SET(id, offset, _id_offset) \ +{ \ + .idx = id, \ + .reg_offset = offset, \ + .id_offset = _id_offset, \ } +/* single address mapping */ +#define RSND_GEN_S_REG(id, offset) \ + RSND_REG_SET(RSND_REG_##id, offset, 0) -static struct regmap_bus rsnd_regmap_bus = { - .write = rsnd_regmap_write32, - .read = rsnd_regmap_read32, - .reg_format_endian_default = REGMAP_ENDIAN_NATIVE, - .val_format_endian_default = REGMAP_ENDIAN_NATIVE, -}; +/* multi address mapping */ +#define RSND_GEN_M_REG(id, offset, _id_offset) \ + RSND_REG_SET(RSND_REG_##id, offset, _id_offset) +/* + * basic function + */ static int rsnd_is_accessible_reg(struct rsnd_priv *priv, struct rsnd_gen *gen, enum rsnd_reg reg) { @@ -88,6 +60,7 @@ static int rsnd_is_accessible_reg(struct rsnd_priv *priv, u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg) { + struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); u32 val; @@ -96,6 +69,8 @@ u32 rsnd_read(struct rsnd_priv *priv, regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); + dev_dbg(dev, "r %s - 0x%04d : %08x\n", rsnd_mod_name(mod), reg, val); + return val; } @@ -103,17 +78,21 @@ void rsnd_write(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 data) { + struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); if (!rsnd_is_accessible_reg(priv, gen, reg)) return; regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data); + + dev_dbg(dev, "w %s - 0x%04d : %08x\n", rsnd_mod_name(mod), reg, data); } void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 mask, u32 data) { + struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); if (!rsnd_is_accessible_reg(priv, gen, reg)) @@ -121,35 +100,63 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, regmap_fields_update_bits(gen->regs[reg], rsnd_mod_id(mod), mask, data); + + dev_dbg(dev, "b %s - 0x%04d : %08x/%08x\n", + rsnd_mod_name(mod), reg, data, mask); } -static int rsnd_gen_regmap_init(struct rsnd_priv *priv, - struct rsnd_gen *gen, - struct reg_field *regf) +#define rsnd_gen_regmap_init(priv, id_size, reg_id, conf) \ + _rsnd_gen_regmap_init(priv, id_size, reg_id, conf, ARRAY_SIZE(conf)) +static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, + int id_size, + int reg_id, + struct rsnd_regmap_field_conf *conf, + int conf_size) { - int i; + struct platform_device *pdev = rsnd_priv_to_pdev(priv); + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); struct device *dev = rsnd_priv_to_dev(priv); + struct resource *res; struct regmap_config regc; + struct regmap_field *regs; + struct regmap *regmap; + struct reg_field regf; + void __iomem *base; + int i; memset(®c, 0, sizeof(regc)); regc.reg_bits = 32; regc.val_bits = 32; + regc.reg_stride = 4; - gen->regmap = devm_regmap_init(dev, &rsnd_regmap_bus, priv, ®c); - if (IS_ERR(gen->regmap)) { - dev_err(dev, "regmap error %ld\n", PTR_ERR(gen->regmap)); - return PTR_ERR(gen->regmap); - } + res = platform_get_resource(pdev, IORESOURCE_MEM, reg_id); + if (!res) + return -ENODEV; - for (i = 0; i < RSND_REG_MAX; i++) { - gen->regs[i] = NULL; - if (!regf[i].reg) - continue; + base = devm_ioremap_resource(dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); - gen->regs[i] = devm_regmap_field_alloc(dev, gen->regmap, regf[i]); - if (IS_ERR(gen->regs[i])) - return PTR_ERR(gen->regs[i]); + regmap = devm_regmap_init_mmio(dev, base, ®c); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + gen->base[reg_id] = base; + gen->regmap[reg_id] = regmap; + + for (i = 0; i < conf_size; i++) { + + regf.reg = conf[i].reg_offset; + regf.id_offset = conf[i].id_offset; + regf.lsb = 0; + regf.msb = 31; + regf.id_size = id_size; + + regs = devm_regmap_field_alloc(dev, regmap, regf); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + gen->regs[conf[i].idx] = regs; } return 0; @@ -165,15 +172,19 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv, * * ex) R-Car H2 case * mod / DMAC in / DMAC out / DMAC PP in / DMAC pp out - * SSI : 0xec541000 / 0xec241008 / 0xec24100c / 0xec400000 / 0xec400000 + * SSI : 0xec541000 / 0xec241008 / 0xec24100c + * SSIU: 0xec541000 / 0xec100000 / 0xec100000 / 0xec400000 / 0xec400000 * SCU : 0xec500000 / 0xec000000 / 0xec004000 / 0xec300000 / 0xec304000 - * CMD : 0xec500000 / 0xec008000 0xec308000 + * CMD : 0xec500000 / / 0xec008000 0xec308000 */ #define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8) #define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc) -#define RDMA_SSI_I_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) -#define RDMA_SSI_O_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) +#define RDMA_SSIU_I_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i)) +#define RDMA_SSIU_O_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i)) + +#define RDMA_SSIU_I_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) +#define RDMA_SSIU_O_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) #define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i)) #define RDMA_SRC_O_N(addr, i) (addr ##_reg - 0x004fc000 + (0x400 * i)) @@ -184,14 +195,13 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv, #define RDMA_CMD_O_N(addr, i) (addr ##_reg - 0x004f8000 + (0x400 * i)) #define RDMA_CMD_O_P(addr, i) (addr ##_reg - 0x001f8000 + (0x400 * i)) -void rsnd_gen_dma_addr(struct rsnd_priv *priv, - struct rsnd_dma *dma, - struct dma_slave_config *cfg, - int is_play, int slave_id) +static dma_addr_t +rsnd_gen2_dma_addr(struct rsnd_priv *priv, + struct rsnd_mod *mod, + int is_play, int is_from) { struct platform_device *pdev = rsnd_priv_to_pdev(priv); struct device *dev = rsnd_priv_to_dev(priv); - struct rsnd_mod *mod = rsnd_dma_to_mod(dma); struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); dma_addr_t ssi_reg = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SSI)->start; @@ -202,170 +212,152 @@ void rsnd_gen_dma_addr(struct rsnd_priv *priv, int use_dvc = !!rsnd_io_to_mod_dvc(io); int id = rsnd_mod_id(mod); struct dma_addr { - dma_addr_t src_addr; - dma_addr_t dst_addr; - } dma_addrs[2][2][3] = { - { /* SRC */ - /* Capture */ - {{ 0, 0 }, - { RDMA_SRC_O_N(src, id), 0 }, - { RDMA_CMD_O_N(src, id), 0 }}, - /* Playback */ - {{ 0, 0, }, - { 0, RDMA_SRC_I_N(src, id) }, - { 0, RDMA_SRC_I_N(src, id) }} - }, { /* SSI */ - /* Capture */ - {{ RDMA_SSI_O_N(ssi, id), 0 }, - { RDMA_SSI_O_P(ssi, id), RDMA_SRC_I_P(src, id) }, - { RDMA_SSI_O_P(ssi, id), RDMA_SRC_I_P(src, id) }}, - /* Playback */ - {{ 0, RDMA_SSI_I_N(ssi, id) }, - { RDMA_SRC_O_P(src, id), RDMA_SSI_I_P(ssi, id) }, - { RDMA_CMD_O_P(src, id), RDMA_SSI_I_P(ssi, id) }} - } + dma_addr_t out_addr; + dma_addr_t in_addr; + } dma_addrs[3][2][3] = { + /* SRC */ + {{{ 0, 0 }, + /* Capture */ + { RDMA_SRC_O_N(src, id), RDMA_SRC_I_P(src, id) }, + { RDMA_CMD_O_N(src, id), RDMA_SRC_I_P(src, id) } }, + /* Playback */ + {{ 0, 0, }, + { RDMA_SRC_O_P(src, id), RDMA_SRC_I_N(src, id) }, + { RDMA_CMD_O_P(src, id), RDMA_SRC_I_N(src, id) } } + }, + /* SSI */ + /* Capture */ + {{{ RDMA_SSI_O_N(ssi, id), 0 }, + { RDMA_SSIU_O_P(ssi, id), 0 }, + { RDMA_SSIU_O_P(ssi, id), 0 } }, + /* Playback */ + {{ 0, RDMA_SSI_I_N(ssi, id) }, + { 0, RDMA_SSIU_I_P(ssi, id) }, + { 0, RDMA_SSIU_I_P(ssi, id) } } + }, + /* SSIU */ + /* Capture */ + {{{ RDMA_SSIU_O_N(ssi, id), 0 }, + { RDMA_SSIU_O_P(ssi, id), 0 }, + { RDMA_SSIU_O_P(ssi, id), 0 } }, + /* Playback */ + {{ 0, RDMA_SSIU_I_N(ssi, id) }, + { 0, RDMA_SSIU_I_P(ssi, id) }, + { 0, RDMA_SSIU_I_P(ssi, id) } } }, }; - cfg->slave_id = slave_id; - cfg->src_addr = 0; - cfg->dst_addr = 0; - cfg->direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + /* it shouldn't happen */ + if (use_dvc & !use_src) + dev_err(dev, "DVC is selected without SRC\n"); + + /* use SSIU or SSI ? */ + if (is_ssi && (0 == strcmp(rsnd_mod_dma_name(mod), "ssiu"))) + is_ssi++; + + return (is_from) ? + dma_addrs[is_ssi][is_play][use_src + use_dvc].out_addr : + dma_addrs[is_ssi][is_play][use_src + use_dvc].in_addr; +} +dma_addr_t rsnd_gen_dma_addr(struct rsnd_priv *priv, + struct rsnd_mod *mod, + int is_play, int is_from) +{ /* * gen1 uses default DMA addr */ if (rsnd_is_gen1(priv)) - return; - - /* it shouldn't happen */ - if (use_dvc & !use_src) { - dev_err(dev, "DVC is selected without SRC\n"); - return; - } + return 0; - cfg->src_addr = dma_addrs[is_ssi][is_play][use_src + use_dvc].src_addr; - cfg->dst_addr = dma_addrs[is_ssi][is_play][use_src + use_dvc].dst_addr; + if (!mod) + return 0; - dev_dbg(dev, "dma%d addr - src : %x / dst : %x\n", - id, cfg->src_addr, cfg->dst_addr); + return rsnd_gen2_dma_addr(priv, mod, is_play, is_from); } /* * Gen2 */ - -/* single address mapping */ -#define RSND_GEN2_S_REG(gen, reg, id, offset) \ - RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, 0, 10) - -/* multi address mapping */ -#define RSND_GEN2_M_REG(gen, reg, id, offset, _id_offset) \ - RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, _id_offset, 10) - -static int rsnd_gen2_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) -{ - struct reg_field regf[RSND_REG_MAX] = { - RSND_GEN2_S_REG(gen, SSIU, SSI_MODE0, 0x800), - RSND_GEN2_S_REG(gen, SSIU, SSI_MODE1, 0x804), - /* FIXME: it needs SSI_MODE2/3 in the future */ - RSND_GEN2_M_REG(gen, SSIU, SSI_BUSIF_MODE, 0x0, 0x80), - RSND_GEN2_M_REG(gen, SSIU, SSI_BUSIF_ADINR,0x4, 0x80), - RSND_GEN2_M_REG(gen, SSIU, SSI_CTRL, 0x10, 0x80), - RSND_GEN2_M_REG(gen, SSIU, INT_ENABLE, 0x18, 0x80), - - RSND_GEN2_M_REG(gen, SCU, SRC_BUSIF_MODE, 0x0, 0x20), - RSND_GEN2_M_REG(gen, SCU, SRC_ROUTE_MODE0,0xc, 0x20), - RSND_GEN2_M_REG(gen, SCU, SRC_CTRL, 0x10, 0x20), - RSND_GEN2_M_REG(gen, SCU, CMD_ROUTE_SLCT, 0x18c, 0x20), - RSND_GEN2_M_REG(gen, SCU, CMD_CTRL, 0x190, 0x20), - RSND_GEN2_M_REG(gen, SCU, SRC_SWRSR, 0x200, 0x40), - RSND_GEN2_M_REG(gen, SCU, SRC_SRCIR, 0x204, 0x40), - RSND_GEN2_M_REG(gen, SCU, SRC_ADINR, 0x214, 0x40), - RSND_GEN2_M_REG(gen, SCU, SRC_IFSCR, 0x21c, 0x40), - RSND_GEN2_M_REG(gen, SCU, SRC_IFSVR, 0x220, 0x40), - RSND_GEN2_M_REG(gen, SCU, SRC_SRCCR, 0x224, 0x40), - RSND_GEN2_M_REG(gen, SCU, SRC_BSDSR, 0x22c, 0x40), - RSND_GEN2_M_REG(gen, SCU, SRC_BSISR, 0x238, 0x40), - RSND_GEN2_M_REG(gen, SCU, DVC_SWRSR, 0xe00, 0x100), - RSND_GEN2_M_REG(gen, SCU, DVC_DVUIR, 0xe04, 0x100), - RSND_GEN2_M_REG(gen, SCU, DVC_ADINR, 0xe08, 0x100), - RSND_GEN2_M_REG(gen, SCU, DVC_DVUCR, 0xe10, 0x100), - RSND_GEN2_M_REG(gen, SCU, DVC_ZCMCR, 0xe14, 0x100), - RSND_GEN2_M_REG(gen, SCU, DVC_VOL0R, 0xe28, 0x100), - RSND_GEN2_M_REG(gen, SCU, DVC_VOL1R, 0xe2c, 0x100), - RSND_GEN2_M_REG(gen, SCU, DVC_DVUER, 0xe48, 0x100), - - RSND_GEN2_S_REG(gen, ADG, BRRA, 0x00), - RSND_GEN2_S_REG(gen, ADG, BRRB, 0x04), - RSND_GEN2_S_REG(gen, ADG, SSICKR, 0x08), - RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c), - RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10), - RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL2, 0x14), - RSND_GEN2_S_REG(gen, ADG, DIV_EN, 0x30), - RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL0, 0x34), - RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL1, 0x38), - RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL2, 0x3c), - RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL3, 0x40), - RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL4, 0x44), - RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL0, 0x48), - RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL1, 0x4c), - RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL2, 0x50), - RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL3, 0x54), - RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL4, 0x58), - RSND_GEN2_S_REG(gen, ADG, CMDOUT_TIMSEL, 0x5c), - - RSND_GEN2_M_REG(gen, SSI, SSICR, 0x00, 0x40), - RSND_GEN2_M_REG(gen, SSI, SSISR, 0x04, 0x40), - RSND_GEN2_M_REG(gen, SSI, SSITDR, 0x08, 0x40), - RSND_GEN2_M_REG(gen, SSI, SSIRDR, 0x0c, 0x40), - RSND_GEN2_M_REG(gen, SSI, SSIWSR, 0x20, 0x40), - }; - - return rsnd_gen_regmap_init(priv, gen, regf); -} - static int rsnd_gen2_probe(struct platform_device *pdev, struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); - struct rsnd_gen *gen = rsnd_priv_to_gen(priv); - struct resource *scu_res; - struct resource *adg_res; - struct resource *ssiu_res; - struct resource *ssi_res; - int ret; - - /* - * map address - */ - scu_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SCU); - adg_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_ADG); - ssiu_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SSIU); - ssi_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SSI); - - gen->base[RSND_GEN2_SCU] = devm_ioremap_resource(dev, scu_res); - gen->base[RSND_GEN2_ADG] = devm_ioremap_resource(dev, adg_res); - gen->base[RSND_GEN2_SSIU] = devm_ioremap_resource(dev, ssiu_res); - gen->base[RSND_GEN2_SSI] = devm_ioremap_resource(dev, ssi_res); - if (IS_ERR(gen->base[RSND_GEN2_SCU]) || - IS_ERR(gen->base[RSND_GEN2_ADG]) || - IS_ERR(gen->base[RSND_GEN2_SSIU]) || - IS_ERR(gen->base[RSND_GEN2_SSI])) - return -ENODEV; - - ret = rsnd_gen2_regmap_init(priv, gen); - if (ret < 0) - return ret; - - dev_dbg(dev, "Gen2 device probed\n"); - dev_dbg(dev, "SCU : %pap => %p\n", &scu_res->start, - gen->base[RSND_GEN2_SCU]); - dev_dbg(dev, "ADG : %pap => %p\n", &adg_res->start, - gen->base[RSND_GEN2_ADG]); - dev_dbg(dev, "SSIU : %pap => %p\n", &ssiu_res->start, - gen->base[RSND_GEN2_SSIU]); - dev_dbg(dev, "SSI : %pap => %p\n", &ssi_res->start, - gen->base[RSND_GEN2_SSI]); + struct rsnd_regmap_field_conf conf_ssiu[] = { + RSND_GEN_S_REG(SSI_MODE0, 0x800), + RSND_GEN_S_REG(SSI_MODE1, 0x804), + /* FIXME: it needs SSI_MODE2/3 in the future */ + RSND_GEN_M_REG(SSI_BUSIF_MODE, 0x0, 0x80), + RSND_GEN_M_REG(SSI_BUSIF_ADINR, 0x4, 0x80), + RSND_GEN_M_REG(BUSIF_DALIGN, 0x8, 0x80), + RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80), + RSND_GEN_M_REG(INT_ENABLE, 0x18, 0x80), + }; + struct rsnd_regmap_field_conf conf_scu[] = { + RSND_GEN_M_REG(SRC_BUSIF_MODE, 0x0, 0x20), + RSND_GEN_M_REG(SRC_ROUTE_MODE0, 0xc, 0x20), + RSND_GEN_M_REG(SRC_CTRL, 0x10, 0x20), + RSND_GEN_M_REG(CMD_ROUTE_SLCT, 0x18c, 0x20), + RSND_GEN_M_REG(CMD_CTRL, 0x190, 0x20), + RSND_GEN_M_REG(SRC_SWRSR, 0x200, 0x40), + RSND_GEN_M_REG(SRC_SRCIR, 0x204, 0x40), + RSND_GEN_M_REG(SRC_ADINR, 0x214, 0x40), + RSND_GEN_M_REG(SRC_IFSCR, 0x21c, 0x40), + RSND_GEN_M_REG(SRC_IFSVR, 0x220, 0x40), + RSND_GEN_M_REG(SRC_SRCCR, 0x224, 0x40), + RSND_GEN_M_REG(SRC_BSDSR, 0x22c, 0x40), + RSND_GEN_M_REG(SRC_BSISR, 0x238, 0x40), + RSND_GEN_M_REG(DVC_SWRSR, 0xe00, 0x100), + RSND_GEN_M_REG(DVC_DVUIR, 0xe04, 0x100), + RSND_GEN_M_REG(DVC_ADINR, 0xe08, 0x100), + RSND_GEN_M_REG(DVC_DVUCR, 0xe10, 0x100), + RSND_GEN_M_REG(DVC_ZCMCR, 0xe14, 0x100), + RSND_GEN_M_REG(DVC_VOL0R, 0xe28, 0x100), + RSND_GEN_M_REG(DVC_VOL1R, 0xe2c, 0x100), + RSND_GEN_M_REG(DVC_DVUER, 0xe48, 0x100), + }; + struct rsnd_regmap_field_conf conf_adg[] = { + RSND_GEN_S_REG(BRRA, 0x00), + RSND_GEN_S_REG(BRRB, 0x04), + RSND_GEN_S_REG(SSICKR, 0x08), + RSND_GEN_S_REG(AUDIO_CLK_SEL0, 0x0c), + RSND_GEN_S_REG(AUDIO_CLK_SEL1, 0x10), + RSND_GEN_S_REG(AUDIO_CLK_SEL2, 0x14), + RSND_GEN_S_REG(DIV_EN, 0x30), + RSND_GEN_S_REG(SRCIN_TIMSEL0, 0x34), + RSND_GEN_S_REG(SRCIN_TIMSEL1, 0x38), + RSND_GEN_S_REG(SRCIN_TIMSEL2, 0x3c), + RSND_GEN_S_REG(SRCIN_TIMSEL3, 0x40), + RSND_GEN_S_REG(SRCIN_TIMSEL4, 0x44), + RSND_GEN_S_REG(SRCOUT_TIMSEL0, 0x48), + RSND_GEN_S_REG(SRCOUT_TIMSEL1, 0x4c), + RSND_GEN_S_REG(SRCOUT_TIMSEL2, 0x50), + RSND_GEN_S_REG(SRCOUT_TIMSEL3, 0x54), + RSND_GEN_S_REG(SRCOUT_TIMSEL4, 0x58), + RSND_GEN_S_REG(CMDOUT_TIMSEL, 0x5c), + }; + struct rsnd_regmap_field_conf conf_ssi[] = { + RSND_GEN_M_REG(SSICR, 0x00, 0x40), + RSND_GEN_M_REG(SSISR, 0x04, 0x40), + RSND_GEN_M_REG(SSITDR, 0x08, 0x40), + RSND_GEN_M_REG(SSIRDR, 0x0c, 0x40), + RSND_GEN_M_REG(SSIWSR, 0x20, 0x40), + }; + int ret_ssiu; + int ret_scu; + int ret_adg; + int ret_ssi; + + ret_ssiu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSIU, conf_ssiu); + ret_scu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SCU, conf_scu); + ret_adg = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_ADG, conf_adg); + ret_ssi = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSI, conf_ssi); + if (ret_ssiu < 0 || + ret_scu < 0 || + ret_adg < 0 || + ret_ssi < 0) + return ret_ssiu | ret_scu | ret_adg | ret_ssi; + + dev_dbg(dev, "Gen2 is probed\n"); return 0; } @@ -374,92 +366,60 @@ static int rsnd_gen2_probe(struct platform_device *pdev, * Gen1 */ -/* single address mapping */ -#define RSND_GEN1_S_REG(gen, reg, id, offset) \ - RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN1_##reg, offset, 0, 9) - -/* multi address mapping */ -#define RSND_GEN1_M_REG(gen, reg, id, offset, _id_offset) \ - RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN1_##reg, offset, _id_offset, 9) - -static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) -{ - struct reg_field regf[RSND_REG_MAX] = { - RSND_GEN1_S_REG(gen, SRU, SRC_ROUTE_SEL, 0x00), - RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL0, 0x08), - RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL1, 0x0c), - RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL2, 0x10), - RSND_GEN1_S_REG(gen, SRU, SRC_ROUTE_CTRL, 0xc0), - RSND_GEN1_S_REG(gen, SRU, SSI_MODE0, 0xD0), - RSND_GEN1_S_REG(gen, SRU, SSI_MODE1, 0xD4), - RSND_GEN1_M_REG(gen, SRU, SRC_BUSIF_MODE, 0x20, 0x4), - RSND_GEN1_M_REG(gen, SRU, SRC_ROUTE_MODE0,0x50, 0x8), - RSND_GEN1_M_REG(gen, SRU, SRC_SWRSR, 0x200, 0x40), - RSND_GEN1_M_REG(gen, SRU, SRC_SRCIR, 0x204, 0x40), - RSND_GEN1_M_REG(gen, SRU, SRC_ADINR, 0x214, 0x40), - RSND_GEN1_M_REG(gen, SRU, SRC_IFSCR, 0x21c, 0x40), - RSND_GEN1_M_REG(gen, SRU, SRC_IFSVR, 0x220, 0x40), - RSND_GEN1_M_REG(gen, SRU, SRC_SRCCR, 0x224, 0x40), - RSND_GEN1_M_REG(gen, SRU, SRC_MNFSR, 0x228, 0x40), - - RSND_GEN1_S_REG(gen, ADG, BRRA, 0x00), - RSND_GEN1_S_REG(gen, ADG, BRRB, 0x04), - RSND_GEN1_S_REG(gen, ADG, SSICKR, 0x08), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL3, 0x18), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL4, 0x1c), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL5, 0x20), - - RSND_GEN1_M_REG(gen, SSI, SSICR, 0x00, 0x40), - RSND_GEN1_M_REG(gen, SSI, SSISR, 0x04, 0x40), - RSND_GEN1_M_REG(gen, SSI, SSITDR, 0x08, 0x40), - RSND_GEN1_M_REG(gen, SSI, SSIRDR, 0x0c, 0x40), - RSND_GEN1_M_REG(gen, SSI, SSIWSR, 0x20, 0x40), - }; - - return rsnd_gen_regmap_init(priv, gen, regf); -} - static int rsnd_gen1_probe(struct platform_device *pdev, struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); - struct rsnd_gen *gen = rsnd_priv_to_gen(priv); - struct resource *sru_res; - struct resource *adg_res; - struct resource *ssi_res; - int ret; - - /* - * map address - */ - sru_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SRU); - adg_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_ADG); - ssi_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SSI); - - gen->base[RSND_GEN1_SRU] = devm_ioremap_resource(dev, sru_res); - gen->base[RSND_GEN1_ADG] = devm_ioremap_resource(dev, adg_res); - gen->base[RSND_GEN1_SSI] = devm_ioremap_resource(dev, ssi_res); - if (IS_ERR(gen->base[RSND_GEN1_SRU]) || - IS_ERR(gen->base[RSND_GEN1_ADG]) || - IS_ERR(gen->base[RSND_GEN1_SSI])) - return -ENODEV; + struct rsnd_regmap_field_conf conf_sru[] = { + RSND_GEN_S_REG(SRC_ROUTE_SEL, 0x00), + RSND_GEN_S_REG(SRC_TMG_SEL0, 0x08), + RSND_GEN_S_REG(SRC_TMG_SEL1, 0x0c), + RSND_GEN_S_REG(SRC_TMG_SEL2, 0x10), + RSND_GEN_S_REG(SRC_ROUTE_CTRL, 0xc0), + RSND_GEN_S_REG(SSI_MODE0, 0xD0), + RSND_GEN_S_REG(SSI_MODE1, 0xD4), + RSND_GEN_M_REG(SRC_BUSIF_MODE, 0x20, 0x4), + RSND_GEN_M_REG(SRC_ROUTE_MODE0, 0x50, 0x8), + RSND_GEN_M_REG(SRC_SWRSR, 0x200, 0x40), + RSND_GEN_M_REG(SRC_SRCIR, 0x204, 0x40), + RSND_GEN_M_REG(SRC_ADINR, 0x214, 0x40), + RSND_GEN_M_REG(SRC_IFSCR, 0x21c, 0x40), + RSND_GEN_M_REG(SRC_IFSVR, 0x220, 0x40), + RSND_GEN_M_REG(SRC_SRCCR, 0x224, 0x40), + RSND_GEN_M_REG(SRC_MNFSR, 0x228, 0x40), + }; + struct rsnd_regmap_field_conf conf_adg[] = { + RSND_GEN_S_REG(BRRA, 0x00), + RSND_GEN_S_REG(BRRB, 0x04), + RSND_GEN_S_REG(SSICKR, 0x08), + RSND_GEN_S_REG(AUDIO_CLK_SEL0, 0x0c), + RSND_GEN_S_REG(AUDIO_CLK_SEL1, 0x10), + RSND_GEN_S_REG(AUDIO_CLK_SEL3, 0x18), + RSND_GEN_S_REG(AUDIO_CLK_SEL4, 0x1c), + RSND_GEN_S_REG(AUDIO_CLK_SEL5, 0x20), + }; + struct rsnd_regmap_field_conf conf_ssi[] = { + RSND_GEN_M_REG(SSICR, 0x00, 0x40), + RSND_GEN_M_REG(SSISR, 0x04, 0x40), + RSND_GEN_M_REG(SSITDR, 0x08, 0x40), + RSND_GEN_M_REG(SSIRDR, 0x0c, 0x40), + RSND_GEN_M_REG(SSIWSR, 0x20, 0x40), + }; + int ret_sru; + int ret_adg; + int ret_ssi; - ret = rsnd_gen1_regmap_init(priv, gen); - if (ret < 0) - return ret; + ret_sru = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SRU, conf_sru); + ret_adg = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_ADG, conf_adg); + ret_ssi = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SSI, conf_ssi); + if (ret_sru < 0 || + ret_adg < 0 || + ret_ssi < 0) + return ret_sru | ret_adg | ret_ssi; - dev_dbg(dev, "Gen1 device probed\n"); - dev_dbg(dev, "SRU : %pap => %p\n", &sru_res->start, - gen->base[RSND_GEN1_SRU]); - dev_dbg(dev, "ADG : %pap => %p\n", &adg_res->start, - gen->base[RSND_GEN1_ADG]); - dev_dbg(dev, "SSI : %pap => %p\n", &ssi_res->start, - gen->base[RSND_GEN1_SSI]); + dev_dbg(dev, "Gen1 is probed\n"); return 0; - } /* diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 39d98af5ee05..d119adf97c9c 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -90,6 +90,7 @@ enum rsnd_reg { RSND_REG_SHARE19, RSND_REG_SHARE20, RSND_REG_SHARE21, + RSND_REG_SHARE22, RSND_REG_MAX, }; @@ -127,6 +128,7 @@ enum rsnd_reg { #define RSND_REG_AUDIO_CLK_SEL2 RSND_REG_SHARE19 #define RSND_REG_CMD_CTRL RSND_REG_SHARE20 #define RSND_REG_CMDOUT_TIMSEL RSND_REG_SHARE21 +#define RSND_REG_BUSIF_DALIGN RSND_REG_SHARE22 struct rsnd_of_data; struct rsnd_priv; @@ -156,12 +158,9 @@ u32 rsnd_get_adinr(struct rsnd_mod *mod); */ struct rsnd_dma { struct sh_dmae_slave slave; - struct work_struct work; struct dma_chan *chan; - enum dma_data_direction dir; - - int submit_loop; - int offset; /* it cares A/B plane */ + enum dma_transfer_direction dir; + dma_addr_t addr; }; void rsnd_dma_start(struct rsnd_dma *dma); @@ -185,6 +184,7 @@ enum rsnd_mod_type { struct rsnd_mod_ops { char *name; + char* (*dma_name)(struct rsnd_mod *mod); int (*probe)(struct rsnd_mod *mod, struct rsnd_dai *rdai); int (*remove)(struct rsnd_mod *mod, @@ -224,6 +224,7 @@ void rsnd_mod_init(struct rsnd_priv *priv, enum rsnd_mod_type type, int id); char *rsnd_mod_name(struct rsnd_mod *mod); +char *rsnd_mod_dma_name(struct rsnd_mod *mod); /* * R-Car sound DAI @@ -281,10 +282,9 @@ int rsnd_gen_probe(struct platform_device *pdev, void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); -void rsnd_gen_dma_addr(struct rsnd_priv *priv, - struct rsnd_dma *dma, - struct dma_slave_config *cfg, - int is_play, int slave_id); +dma_addr_t rsnd_gen_dma_addr(struct rsnd_priv *priv, + struct rsnd_mod *mod, + int is_play, int is_from); #define rsnd_is_gen1(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN1) #define rsnd_is_gen2(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN2) @@ -391,8 +391,12 @@ struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id); unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv, struct rsnd_dai_stream *io, struct snd_pcm_runtime *runtime); -int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod, - struct rsnd_dai *rdai); +int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod, + struct rsnd_dai *rdai, + int use_busif); +int rsnd_src_ssiu_stop(struct rsnd_mod *ssi_mod, + struct rsnd_dai *rdai, + int use_busif); int rsnd_src_enable_ssi_irq(struct rsnd_mod *ssi_mod, struct rsnd_dai *rdai); diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 200eda019bc7..9183e0145503 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -106,18 +106,19 @@ struct rsnd_src { /* * Gen1/Gen2 common functions */ -int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod, - struct rsnd_dai *rdai) +int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod, + struct rsnd_dai *rdai, + int use_busif) { struct rsnd_dai_stream *io = rsnd_mod_to_io(ssi_mod); - struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); int ssi_id = rsnd_mod_id(ssi_mod); /* * SSI_MODE0 */ rsnd_mod_bset(ssi_mod, SSI_MODE0, (1 << ssi_id), - src_mod ? 0 : (1 << ssi_id)); + !use_busif << ssi_id); /* * SSI_MODE1 @@ -143,6 +144,46 @@ int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod, 0x2 << shift : 0x1 << shift); } + /* + * DMA settings for SSIU + */ + if (use_busif) { + u32 val = 0x76543210; + u32 mask = ~0; + + rsnd_mod_write(ssi_mod, SSI_BUSIF_ADINR, + rsnd_get_adinr(ssi_mod)); + rsnd_mod_write(ssi_mod, SSI_BUSIF_MODE, 1); + rsnd_mod_write(ssi_mod, SSI_CTRL, 0x1); + + mask <<= runtime->channels * 4; + val = val & mask; + + switch (runtime->sample_bits) { + case 16: + val |= 0x67452301 & ~mask; + break; + case 32: + val |= 0x76543210 & ~mask; + break; + } + rsnd_mod_write(ssi_mod, BUSIF_DALIGN, val); + + } + + return 0; +} + +int rsnd_src_ssiu_stop(struct rsnd_mod *ssi_mod, + struct rsnd_dai *rdai, + int use_busif) +{ + /* + * DMA settings for SSIU + */ + if (use_busif) + rsnd_mod_write(ssi_mod, SSI_CTRL, 0); + return 0; } @@ -461,18 +502,45 @@ static struct rsnd_mod_ops rsnd_src_gen1_ops = { static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod, struct rsnd_dai *rdai) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct rsnd_src *src = rsnd_mod_to_src(mod); + uint ratio; int ret; + /* 6 - 1/6 are very enough ratio for SRC_BSDSR */ + if (!rsnd_src_convert_rate(src)) + ratio = 0; + else if (rsnd_src_convert_rate(src) > runtime->rate) + ratio = 100 * rsnd_src_convert_rate(src) / runtime->rate; + else + ratio = 100 * runtime->rate / rsnd_src_convert_rate(src); + + if (ratio > 600) { + dev_err(dev, "FSO/FSI ratio error\n"); + return -EINVAL; + } + ret = rsnd_src_set_convert_rate(mod, rdai); if (ret < 0) return ret; - rsnd_mod_write(mod, SSI_BUSIF_ADINR, rsnd_get_adinr(mod)); - rsnd_mod_write(mod, SSI_BUSIF_MODE, 1); - rsnd_mod_write(mod, SRC_SRCCR, 0x00011110); - rsnd_mod_write(mod, SRC_BSDSR, 0x01800000); + switch (rsnd_mod_id(mod)) { + case 5: + case 6: + case 7: + case 8: + rsnd_mod_write(mod, SRC_BSDSR, 0x02400000); + break; + default: + rsnd_mod_write(mod, SRC_BSDSR, 0x01800000); + break; + } + rsnd_mod_write(mod, SRC_BSISR, 0x00100060); return 0; @@ -554,7 +622,6 @@ static int rsnd_src_start_gen2(struct rsnd_mod *mod, rsnd_dma_start(rsnd_mod_to_dma(&src->mod)); - rsnd_mod_write(mod, SSI_CTRL, 0x1); rsnd_mod_write(mod, SRC_CTRL, val); return rsnd_src_start(mod, rdai); @@ -565,7 +632,6 @@ static int rsnd_src_stop_gen2(struct rsnd_mod *mod, { struct rsnd_src *src = rsnd_mod_to_src(mod); - rsnd_mod_write(mod, SSI_CTRL, 0); rsnd_mod_write(mod, SRC_CTRL, 0); rsnd_dma_stop(rsnd_mod_to_dma(&src->mod)); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 2df723df5d19..34e84009162b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -90,6 +90,20 @@ struct rsnd_ssi { #define rsnd_ssi_mode_flags(p) ((p)->info->flags) #define rsnd_ssi_dai_id(ssi) ((ssi)->info->dai_id) +static int rsnd_ssi_use_busif(struct rsnd_mod *mod) +{ + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + int use_busif = 0; + + if (!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_NO_BUSIF)) + use_busif = 1; + if (rsnd_io_to_mod_src(io)) + use_busif = 1; + + return use_busif; +} + static void rsnd_ssi_status_check(struct rsnd_mod *mod, u32 bit) { @@ -289,8 +303,6 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, ssi->cr_own = cr; ssi->err = -1; /* ignore 1st error */ - rsnd_src_ssi_mode_init(mod, rdai); - return 0; } @@ -389,6 +401,8 @@ static int rsnd_ssi_pio_start(struct rsnd_mod *mod, /* enable PIO IRQ */ ssi->cr_etc = UIEN | OIEN | DIEN; + rsnd_src_ssiu_start(mod, rdai, 0); + rsnd_src_enable_ssi_irq(mod, rdai); rsnd_ssi_hw_start(ssi, rdai, io); @@ -405,6 +419,8 @@ static int rsnd_ssi_pio_stop(struct rsnd_mod *mod, rsnd_ssi_hw_stop(ssi, rdai); + rsnd_src_ssiu_stop(mod, rdai, 0); + return 0; } @@ -457,6 +473,8 @@ static int rsnd_ssi_dma_start(struct rsnd_mod *mod, /* enable DMA transfer */ ssi->cr_etc = DMEN; + rsnd_src_ssiu_start(mod, rdai, rsnd_ssi_use_busif(mod)); + rsnd_dma_start(dma); rsnd_ssi_hw_start(ssi, ssi->rdai, io); @@ -482,11 +500,19 @@ static int rsnd_ssi_dma_stop(struct rsnd_mod *mod, rsnd_dma_stop(dma); + rsnd_src_ssiu_stop(mod, rdai, 1); + return 0; } +static char *rsnd_ssi_dma_name(struct rsnd_mod *mod) +{ + return rsnd_ssi_use_busif(mod) ? "ssiu" : SSI_NAME; +} + static struct rsnd_mod_ops rsnd_ssi_dma_ops = { .name = SSI_NAME, + .dma_name = rsnd_ssi_dma_name, .probe = rsnd_ssi_dma_probe, .remove = rsnd_ssi_dma_remove, .init = rsnd_ssi_init, @@ -595,6 +621,9 @@ static void rsnd_of_parse_ssi(struct platform_device *pdev, */ ssi_info->dma_id = of_get_property(np, "pio-transfer", NULL) ? 0 : 1; + + if (of_get_property(np, "no-busif", NULL)) + ssi_info->flags |= RSND_SSI_NO_BUSIF; } rsnd_of_parse_ssi_end: diff --git a/sound/soc/sirf/Kconfig b/sound/soc/sirf/Kconfig index 89e89429b04a..840058dcad09 100644 --- a/sound/soc/sirf/Kconfig +++ b/sound/soc/sirf/Kconfig @@ -12,3 +12,9 @@ config SND_SOC_SIRF_AUDIO config SND_SOC_SIRF_AUDIO_PORT select REGMAP_MMIO tristate + +config SND_SOC_SIRF_USP + tristate "SoC Audio (I2S protocol) for SiRF SoC USP interface" + depends on SND_SOC_SIRF + select REGMAP_MMIO + tristate diff --git a/sound/soc/sirf/Makefile b/sound/soc/sirf/Makefile index 913b93231d4e..dd917f20f12f 100644 --- a/sound/soc/sirf/Makefile +++ b/sound/soc/sirf/Makefile @@ -1,5 +1,7 @@ snd-soc-sirf-audio-objs := sirf-audio.o snd-soc-sirf-audio-port-objs := sirf-audio-port.o +snd-soc-sirf-usp-objs := sirf-usp.o obj-$(CONFIG_SND_SOC_SIRF_AUDIO) += snd-soc-sirf-audio.o obj-$(CONFIG_SND_SOC_SIRF_AUDIO_PORT) += snd-soc-sirf-audio-port.o +obj-$(CONFIG_SND_SOC_SIRF_USP) += snd-soc-sirf-usp.o diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c new file mode 100644 index 000000000000..3a730374e259 --- /dev/null +++ b/sound/soc/sirf/sirf-usp.c @@ -0,0 +1,415 @@ +/* + * SiRF USP in I2S/DSP mode + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ +#include <linux/module.h> +#include <linux/io.h> +#include <linux/of.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/dmaengine_pcm.h> + +#include "sirf-usp.h" + +struct sirf_usp { + struct regmap *regmap; + struct clk *clk; + u32 mode1_reg; + u32 mode2_reg; + int daifmt_format; + struct snd_dmaengine_dai_dma_data playback_dma_data; + struct snd_dmaengine_dai_dma_data capture_dma_data; +}; + +static void sirf_usp_tx_enable(struct sirf_usp *usp) +{ + regmap_update_bits(usp->regmap, USP_TX_FIFO_OP, + USP_TX_FIFO_RESET, USP_TX_FIFO_RESET); + regmap_write(usp->regmap, USP_TX_FIFO_OP, 0); + + regmap_update_bits(usp->regmap, USP_TX_FIFO_OP, + USP_TX_FIFO_START, USP_TX_FIFO_START); + + regmap_update_bits(usp->regmap, USP_TX_RX_ENABLE, + USP_TX_ENA, USP_TX_ENA); +} + +static void sirf_usp_tx_disable(struct sirf_usp *usp) +{ + regmap_update_bits(usp->regmap, USP_TX_RX_ENABLE, + USP_TX_ENA, ~USP_TX_ENA); + /* FIFO stop */ + regmap_write(usp->regmap, USP_TX_FIFO_OP, 0); +} + +static void sirf_usp_rx_enable(struct sirf_usp *usp) +{ + regmap_update_bits(usp->regmap, USP_RX_FIFO_OP, + USP_RX_FIFO_RESET, USP_RX_FIFO_RESET); + regmap_write(usp->regmap, USP_RX_FIFO_OP, 0); + + regmap_update_bits(usp->regmap, USP_RX_FIFO_OP, + USP_RX_FIFO_START, USP_RX_FIFO_START); + + regmap_update_bits(usp->regmap, USP_TX_RX_ENABLE, + USP_RX_ENA, USP_RX_ENA); +} + +static void sirf_usp_rx_disable(struct sirf_usp *usp) +{ + regmap_update_bits(usp->regmap, USP_TX_RX_ENABLE, + USP_RX_ENA, ~USP_RX_ENA); + /* FIFO stop */ + regmap_write(usp->regmap, USP_RX_FIFO_OP, 0); +} + +static int sirf_usp_pcm_dai_probe(struct snd_soc_dai *dai) +{ + struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai); + snd_soc_dai_init_dma_data(dai, &usp->playback_dma_data, + &usp->capture_dma_data); + return 0; +} + +static int sirf_usp_pcm_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + dev_err(dai->dev, "Only CBM and CFM supported\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + usp->daifmt_format = (fmt & SND_SOC_DAIFMT_FORMAT_MASK); + break; + default: + dev_err(dai->dev, "Only I2S and DSP_A format supported\n"); + return -EINVAL; + } + + return 0; +} + +static void sirf_usp_i2s_init(struct sirf_usp *usp) +{ + /* Configure RISC mode */ + regmap_update_bits(usp->regmap, USP_RISC_DSP_MODE, + USP_RISC_DSP_SEL, ~USP_RISC_DSP_SEL); + + /* + * Configure DMA IO Length register + * Set no limit, USP can receive data continuously until it is diabled + */ + regmap_write(usp->regmap, USP_TX_DMA_IO_LEN, 0); + regmap_write(usp->regmap, USP_RX_DMA_IO_LEN, 0); + + /* Configure Mode2 register */ + regmap_write(usp->regmap, USP_MODE2, (1 << USP_RXD_DELAY_LEN_OFFSET) | + (0 << USP_TXD_DELAY_LEN_OFFSET) | + USP_TFS_CLK_SLAVE_MODE | USP_RFS_CLK_SLAVE_MODE); + + /* Configure Mode1 register */ + regmap_write(usp->regmap, USP_MODE1, + USP_SYNC_MODE | USP_EN | USP_TXD_ACT_EDGE_FALLING | + USP_RFS_ACT_LEVEL_LOGIC1 | USP_TFS_ACT_LEVEL_LOGIC1 | + USP_TX_UFLOW_REPEAT_ZERO | USP_CLOCK_MODE_SLAVE); + + /* Configure RX DMA IO Control register */ + regmap_write(usp->regmap, USP_RX_DMA_IO_CTRL, 0); + + /* Congiure RX FIFO Control register */ + regmap_write(usp->regmap, USP_RX_FIFO_CTRL, + (USP_RX_FIFO_THRESHOLD << USP_RX_FIFO_THD_OFFSET) | + (USP_TX_RX_FIFO_WIDTH_DWORD << USP_RX_FIFO_WIDTH_OFFSET)); + + /* Congiure RX FIFO Level Check register */ + regmap_write(usp->regmap, USP_RX_FIFO_LEVEL_CHK, + RX_FIFO_SC(0x04) | RX_FIFO_LC(0x0E) | RX_FIFO_HC(0x1B)); + + /* Configure TX DMA IO Control register*/ + regmap_write(usp->regmap, USP_TX_DMA_IO_CTRL, 0); + + /* Configure TX FIFO Control register */ + regmap_write(usp->regmap, USP_TX_FIFO_CTRL, + (USP_TX_FIFO_THRESHOLD << USP_TX_FIFO_THD_OFFSET) | + (USP_TX_RX_FIFO_WIDTH_DWORD << USP_TX_FIFO_WIDTH_OFFSET)); + /* Congiure TX FIFO Level Check register */ + regmap_write(usp->regmap, USP_TX_FIFO_LEVEL_CHK, + TX_FIFO_SC(0x1B) | TX_FIFO_LC(0x0E) | TX_FIFO_HC(0x04)); +} + +static int sirf_usp_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai); + u32 data_len, frame_len, shifter_len; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + data_len = 16; + frame_len = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + data_len = 24; + frame_len = 32; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + data_len = 24; + frame_len = 24; + break; + default: + dev_err(dai->dev, "Format unsupported\n"); + return -EINVAL; + } + + shifter_len = data_len; + + switch (usp->daifmt_format) { + case SND_SOC_DAIFMT_I2S: + regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL, + USP_I2S_SYNC_CHG, USP_I2S_SYNC_CHG); + break; + case SND_SOC_DAIFMT_DSP_A: + regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL, + USP_I2S_SYNC_CHG, 0); + frame_len = data_len * params_channels(params); + data_len = frame_len; + break; + default: + dev_err(dai->dev, "Only support I2S and DSP_A mode\n"); + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_update_bits(usp->regmap, USP_TX_FRAME_CTRL, + USP_TXC_DATA_LEN_MASK | USP_TXC_FRAME_LEN_MASK + | USP_TXC_SHIFTER_LEN_MASK | USP_TXC_SLAVE_CLK_SAMPLE, + ((data_len - 1) << USP_TXC_DATA_LEN_OFFSET) + | ((frame_len - 1) << USP_TXC_FRAME_LEN_OFFSET) + | ((shifter_len - 1) << USP_TXC_SHIFTER_LEN_OFFSET) + | USP_TXC_SLAVE_CLK_SAMPLE); + else + regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL, + USP_RXC_DATA_LEN_MASK | USP_RXC_FRAME_LEN_MASK + | USP_RXC_SHIFTER_LEN_MASK | USP_SINGLE_SYNC_MODE, + ((data_len - 1) << USP_RXC_DATA_LEN_OFFSET) + | ((frame_len - 1) << USP_RXC_FRAME_LEN_OFFSET) + | ((shifter_len - 1) << USP_RXC_SHIFTER_LEN_OFFSET) + | USP_SINGLE_SYNC_MODE); + + return 0; +} + +static int sirf_usp_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sirf_usp_tx_enable(usp); + else + sirf_usp_rx_enable(usp); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sirf_usp_tx_disable(usp); + else + sirf_usp_rx_disable(usp); + break; + } + + return 0; +} + +static const struct snd_soc_dai_ops sirf_usp_pcm_dai_ops = { + .trigger = sirf_usp_pcm_trigger, + .set_fmt = sirf_usp_pcm_set_dai_fmt, + .hw_params = sirf_usp_pcm_hw_params, +}; + +static struct snd_soc_dai_driver sirf_usp_pcm_dai = { + .probe = sirf_usp_pcm_dai_probe, + .name = "sirf-usp-pcm", + .id = 0, + .playback = { + .stream_name = "SiRF USP PCM Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + }, + .capture = { + .stream_name = "SiRF USP PCM Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + }, + .ops = &sirf_usp_pcm_dai_ops, +}; + +static int sirf_usp_pcm_runtime_suspend(struct device *dev) +{ + struct sirf_usp *usp = dev_get_drvdata(dev); + clk_disable_unprepare(usp->clk); + return 0; +} + +static int sirf_usp_pcm_runtime_resume(struct device *dev) +{ + struct sirf_usp *usp = dev_get_drvdata(dev); + int ret; + ret = clk_prepare_enable(usp->clk); + if (ret) { + dev_err(dev, "clk_enable failed: %d\n", ret); + return ret; + } + sirf_usp_i2s_init(usp); + return 0; +} + +#ifdef CONFIG_PM_SLEEP +static int sirf_usp_pcm_suspend(struct device *dev) +{ + struct sirf_usp *usp = dev_get_drvdata(dev); + + if (!pm_runtime_status_suspended(dev)) { + regmap_read(usp->regmap, USP_MODE1, &usp->mode1_reg); + regmap_read(usp->regmap, USP_MODE2, &usp->mode2_reg); + sirf_usp_pcm_runtime_suspend(dev); + } + return 0; +} + +static int sirf_usp_pcm_resume(struct device *dev) +{ + struct sirf_usp *usp = dev_get_drvdata(dev); + int ret; + + if (!pm_runtime_status_suspended(dev)) { + ret = sirf_usp_pcm_runtime_resume(dev); + if (ret) + return ret; + regmap_write(usp->regmap, USP_MODE1, usp->mode1_reg); + regmap_write(usp->regmap, USP_MODE2, usp->mode2_reg); + } + return 0; +} +#endif + +static const struct snd_soc_component_driver sirf_usp_component = { + .name = "sirf-usp", +}; + +static const struct regmap_config sirf_usp_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = USP_RX_FIFO_DATA, + .cache_type = REGCACHE_NONE, +}; + +static int sirf_usp_pcm_probe(struct platform_device *pdev) +{ + int ret; + struct sirf_usp *usp; + void __iomem *base; + struct resource *mem_res; + + usp = devm_kzalloc(&pdev->dev, sizeof(struct sirf_usp), + GFP_KERNEL); + if (!usp) + return -ENOMEM; + + 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; + usp->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sirf_usp_regmap_config); + if (IS_ERR(usp->regmap)) + return PTR_ERR(usp->regmap); + + usp->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(usp->clk)) { + dev_err(&pdev->dev, "Get clock failed.\n"); + return PTR_ERR(usp->clk); + } + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sirf_usp_pcm_runtime_resume(&pdev->dev); + if (ret) + return ret; + } + + ret = devm_snd_soc_register_component(&pdev->dev, &sirf_usp_component, + &sirf_usp_pcm_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Register Audio SoC dai failed.\n"); + return ret; + } + return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); +} + +static int sirf_usp_pcm_remove(struct platform_device *pdev) +{ + if (!pm_runtime_enabled(&pdev->dev)) + sirf_usp_pcm_runtime_suspend(&pdev->dev); + else + pm_runtime_disable(&pdev->dev); + return 0; +} + +static const struct of_device_id sirf_usp_pcm_of_match[] = { + { .compatible = "sirf,prima2-usp-pcm", }, + {} +}; +MODULE_DEVICE_TABLE(of, sirf_usp_pcm_of_match); + +static const struct dev_pm_ops sirf_usp_pcm_pm_ops = { + SET_RUNTIME_PM_OPS(sirf_usp_pcm_runtime_suspend, + sirf_usp_pcm_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(sirf_usp_pcm_suspend, sirf_usp_pcm_resume) +}; + +static struct platform_driver sirf_usp_pcm_driver = { + .driver = { + .name = "sirf-usp-pcm", + .owner = THIS_MODULE, + .of_match_table = sirf_usp_pcm_of_match, + .pm = &sirf_usp_pcm_pm_ops, + }, + .probe = sirf_usp_pcm_probe, + .remove = sirf_usp_pcm_remove, +}; + +module_platform_driver(sirf_usp_pcm_driver); + +MODULE_DESCRIPTION("SiRF SoC USP PCM bus driver"); +MODULE_AUTHOR("RongJun Ying <Rongjun.Ying@csr.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/sirf/sirf-usp.h b/sound/soc/sirf/sirf-usp.h new file mode 100644 index 000000000000..bf0201cb15bc --- /dev/null +++ b/sound/soc/sirf/sirf-usp.h @@ -0,0 +1,293 @@ +/* + * arch/arm/mach-prima2/include/mach/sirfsoc_usp.h + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ + +#ifndef _SIRF_USP_H +#define _SIRF_USP_H + +/* USP Registers */ +#define USP_MODE1 0x00 +#define USP_MODE2 0x04 +#define USP_TX_FRAME_CTRL 0x08 +#define USP_RX_FRAME_CTRL 0x0C +#define USP_TX_RX_ENABLE 0x10 +#define USP_INT_ENABLE 0x14 +#define USP_INT_STATUS 0x18 +#define USP_PIN_IO_DATA 0x1C +#define USP_RISC_DSP_MODE 0x20 +#define USP_AYSNC_PARAM_REG 0x24 +#define USP_IRDA_X_MODE_DIV 0x28 +#define USP_SM_CFG 0x2C +#define USP_TX_DMA_IO_CTRL 0x100 +#define USP_TX_DMA_IO_LEN 0x104 +#define USP_TX_FIFO_CTRL 0x108 +#define USP_TX_FIFO_LEVEL_CHK 0x10C +#define USP_TX_FIFO_OP 0x110 +#define USP_TX_FIFO_STATUS 0x114 +#define USP_TX_FIFO_DATA 0x118 +#define USP_RX_DMA_IO_CTRL 0x120 +#define USP_RX_DMA_IO_LEN 0x124 +#define USP_RX_FIFO_CTRL 0x128 +#define USP_RX_FIFO_LEVEL_CHK 0x12C +#define USP_RX_FIFO_OP 0x130 +#define USP_RX_FIFO_STATUS 0x134 +#define USP_RX_FIFO_DATA 0x138 + +/* USP MODE register-1 */ +#define USP_SYNC_MODE 0x00000001 +#define USP_CLOCK_MODE_SLAVE 0x00000002 +#define USP_LOOP_BACK_EN 0x00000004 +#define USP_HPSIR_EN 0x00000008 +#define USP_ENDIAN_CTRL_LSBF 0x00000010 +#define USP_EN 0x00000020 +#define USP_RXD_ACT_EDGE_FALLING 0x00000040 +#define USP_TXD_ACT_EDGE_FALLING 0x00000080 +#define USP_RFS_ACT_LEVEL_LOGIC1 0x00000100 +#define USP_TFS_ACT_LEVEL_LOGIC1 0x00000200 +#define USP_SCLK_IDLE_MODE_TOGGLE 0x00000400 +#define USP_SCLK_IDLE_LEVEL_LOGIC1 0x00000800 +#define USP_SCLK_PIN_MODE_IO 0x00001000 +#define USP_RFS_PIN_MODE_IO 0x00002000 +#define USP_TFS_PIN_MODE_IO 0x00004000 +#define USP_RXD_PIN_MODE_IO 0x00008000 +#define USP_TXD_PIN_MODE_IO 0x00010000 +#define USP_SCLK_IO_MODE_INPUT 0x00020000 +#define USP_RFS_IO_MODE_INPUT 0x00040000 +#define USP_TFS_IO_MODE_INPUT 0x00080000 +#define USP_RXD_IO_MODE_INPUT 0x00100000 +#define USP_TXD_IO_MODE_INPUT 0x00200000 +#define USP_IRDA_WIDTH_DIV_MASK 0x3FC00000 +#define USP_IRDA_WIDTH_DIV_OFFSET 0 +#define USP_IRDA_IDLE_LEVEL_HIGH 0x40000000 +#define USP_TX_UFLOW_REPEAT_ZERO 0x80000000 +#define USP_TX_ENDIAN_MODE 0x00000020 +#define USP_RX_ENDIAN_MODE 0x00000020 + +/* USP Mode Register-2 */ +#define USP_RXD_DELAY_LEN_MASK 0x000000FF +#define USP_RXD_DELAY_LEN_OFFSET 0 + +#define USP_TXD_DELAY_LEN_MASK 0x0000FF00 +#define USP_TXD_DELAY_LEN_OFFSET 8 + +#define USP_ENA_CTRL_MODE 0x00010000 +#define USP_FRAME_CTRL_MODE 0x00020000 +#define USP_TFS_SOURCE_MODE 0x00040000 +#define USP_TFS_MS_MODE 0x00080000 +#define USP_CLK_DIVISOR_MASK 0x7FE00000 +#define USP_CLK_DIVISOR_OFFSET 21 + +#define USP_TFS_CLK_SLAVE_MODE (1<<20) +#define USP_RFS_CLK_SLAVE_MODE (1<<19) + +#define USP_IRDA_DATA_WIDTH 0x80000000 + +/* USP Transmit Frame Control Register */ + +#define USP_TXC_DATA_LEN_MASK 0x000000FF +#define USP_TXC_DATA_LEN_OFFSET 0 + +#define USP_TXC_SYNC_LEN_MASK 0x0000FF00 +#define USP_TXC_SYNC_LEN_OFFSET 8 + +#define USP_TXC_FRAME_LEN_MASK 0x00FF0000 +#define USP_TXC_FRAME_LEN_OFFSET 16 + +#define USP_TXC_SHIFTER_LEN_MASK 0x1F000000 +#define USP_TXC_SHIFTER_LEN_OFFSET 24 + +#define USP_TXC_SLAVE_CLK_SAMPLE 0x20000000 + +#define USP_TXC_CLK_DIVISOR_MASK 0xC0000000 +#define USP_TXC_CLK_DIVISOR_OFFSET 30 + +/* USP Receive Frame Control Register */ + +#define USP_RXC_DATA_LEN_MASK 0x000000FF +#define USP_RXC_DATA_LEN_OFFSET 0 + +#define USP_RXC_FRAME_LEN_MASK 0x0000FF00 +#define USP_RXC_FRAME_LEN_OFFSET 8 + +#define USP_RXC_SHIFTER_LEN_MASK 0x001F0000 +#define USP_RXC_SHIFTER_LEN_OFFSET 16 + +#define USP_START_EDGE_MODE 0x00800000 +#define USP_I2S_SYNC_CHG 0x00200000 + +#define USP_RXC_CLK_DIVISOR_MASK 0x0F000000 +#define USP_RXC_CLK_DIVISOR_OFFSET 24 +#define USP_SINGLE_SYNC_MODE 0x00400000 + +/* Tx - RX Enable Register */ + +#define USP_RX_ENA 0x00000001 +#define USP_TX_ENA 0x00000002 + +/* USP Interrupt Enable and status Register */ +#define USP_RX_DONE_INT 0x00000001 +#define USP_TX_DONE_INT 0x00000002 +#define USP_RX_OFLOW_INT 0x00000004 +#define USP_TX_UFLOW_INT 0x00000008 +#define USP_RX_IO_DMA_INT 0x00000010 +#define USP_TX_IO_DMA_INT 0x00000020 +#define USP_RXFIFO_FULL_INT 0x00000040 +#define USP_TXFIFO_EMPTY_INT 0x00000080 +#define USP_RXFIFO_THD_INT 0x00000100 +#define USP_TXFIFO_THD_INT 0x00000200 +#define USP_UART_FRM_ERR_INT 0x00000400 +#define USP_RX_TIMEOUT_INT 0x00000800 +#define USP_TX_ALLOUT_INT 0x00001000 +#define USP_RXD_BREAK_INT 0x00008000 + +/* All possible TX interruots */ +#define USP_TX_INTERRUPT (USP_TX_DONE_INT|USP_TX_UFLOW_INT|\ + USP_TX_IO_DMA_INT|\ + USP_TXFIFO_EMPTY_INT|\ + USP_TXFIFO_THD_INT) +/* All possible RX interruots */ +#define USP_RX_INTERRUPT (USP_RX_DONE_INT|USP_RX_OFLOW_INT|\ + USP_RX_IO_DMA_INT|\ + USP_RXFIFO_FULL_INT|\ + USP_RXFIFO_THD_INT|\ + USP_RXFIFO_THD_INT|USP_RX_TIMEOUT_INT) + +#define USP_INT_ALL 0x1FFF + +/* USP Pin I/O Data Register */ + +#define USP_RFS_PIN_VALUE_MASK 0x00000001 +#define USP_TFS_PIN_VALUE_MASK 0x00000002 +#define USP_RXD_PIN_VALUE_MASK 0x00000004 +#define USP_TXD_PIN_VALUE_MASK 0x00000008 +#define USP_SCLK_PIN_VALUE_MASK 0x00000010 + +/* USP RISC/DSP Mode Register */ +#define USP_RISC_DSP_SEL 0x00000001 + +/* USP ASYNC PARAMETER Register*/ + +#define USP_ASYNC_TIMEOUT_MASK 0x0000FFFF +#define USP_ASYNC_TIMEOUT_OFFSET 0 +#define USP_ASYNC_TIMEOUT(x) (((x)&USP_ASYNC_TIMEOUT_MASK) \ + <<USP_ASYNC_TIMEOUT_OFFSET) + +#define USP_ASYNC_DIV2_MASK 0x003F0000 +#define USP_ASYNC_DIV2_OFFSET 16 + +/* USP TX DMA I/O MODE Register */ +#define USP_TX_MODE_IO 0x00000001 + +/* USP TX DMA I/O Length Register */ +#define USP_TX_DATA_LEN_MASK 0xFFFFFFFF +#define USP_TX_DATA_LEN_OFFSET 0 + +/* USP TX FIFO Control Register */ +#define USP_TX_FIFO_WIDTH_MASK 0x00000003 +#define USP_TX_FIFO_WIDTH_OFFSET 0 + +#define USP_TX_FIFO_THD_MASK 0x000001FC +#define USP_TX_FIFO_THD_OFFSET 2 + +/* USP TX FIFO Level Check Register */ +#define USP_TX_FIFO_LEVEL_CHECK_MASK 0x1F +#define USP_TX_FIFO_SC_OFFSET 0 +#define USP_TX_FIFO_LC_OFFSET 10 +#define USP_TX_FIFO_HC_OFFSET 20 + +#define TX_FIFO_SC(x) (((x) & USP_TX_FIFO_LEVEL_CHECK_MASK) \ + << USP_TX_FIFO_SC_OFFSET) +#define TX_FIFO_LC(x) (((x) & USP_TX_FIFO_LEVEL_CHECK_MASK) \ + << USP_TX_FIFO_LC_OFFSET) +#define TX_FIFO_HC(x) (((x) & USP_TX_FIFO_LEVEL_CHECK_MASK) \ + << USP_TX_FIFO_HC_OFFSET) + +/* USP TX FIFO Operation Register */ +#define USP_TX_FIFO_RESET 0x00000001 +#define USP_TX_FIFO_START 0x00000002 + +/* USP TX FIFO Status Register */ +#define USP_TX_FIFO_LEVEL_MASK 0x0000007F +#define USP_TX_FIFO_LEVEL_OFFSET 0 + +#define USP_TX_FIFO_FULL 0x00000080 +#define USP_TX_FIFO_EMPTY 0x00000100 + +/* USP TX FIFO Data Register */ +#define USP_TX_FIFO_DATA_MASK 0xFFFFFFFF +#define USP_TX_FIFO_DATA_OFFSET 0 + +/* USP RX DMA I/O MODE Register */ +#define USP_RX_MODE_IO 0x00000001 +#define USP_RX_DMA_FLUSH 0x00000004 + +/* USP RX DMA I/O Length Register */ +#define USP_RX_DATA_LEN_MASK 0xFFFFFFFF +#define USP_RX_DATA_LEN_OFFSET 0 + +/* USP RX FIFO Control Register */ +#define USP_RX_FIFO_WIDTH_MASK 0x00000003 +#define USP_RX_FIFO_WIDTH_OFFSET 0 + +#define USP_RX_FIFO_THD_MASK 0x000001FC +#define USP_RX_FIFO_THD_OFFSET 2 + +/* USP RX FIFO Level Check Register */ + +#define USP_RX_FIFO_LEVEL_CHECK_MASK 0x1F +#define USP_RX_FIFO_SC_OFFSET 0 +#define USP_RX_FIFO_LC_OFFSET 10 +#define USP_RX_FIFO_HC_OFFSET 20 + +#define RX_FIFO_SC(x) (((x) & USP_RX_FIFO_LEVEL_CHECK_MASK) \ + << USP_RX_FIFO_SC_OFFSET) +#define RX_FIFO_LC(x) (((x) & USP_RX_FIFO_LEVEL_CHECK_MASK) \ + << USP_RX_FIFO_LC_OFFSET) +#define RX_FIFO_HC(x) (((x) & USP_RX_FIFO_LEVEL_CHECK_MASK) \ + << USP_RX_FIFO_HC_OFFSET) + +/* USP RX FIFO Operation Register */ +#define USP_RX_FIFO_RESET 0x00000001 +#define USP_RX_FIFO_START 0x00000002 + +/* USP RX FIFO Status Register */ + +#define USP_RX_FIFO_LEVEL_MASK 0x0000007F +#define USP_RX_FIFO_LEVEL_OFFSET 0 + +#define USP_RX_FIFO_FULL 0x00000080 +#define USP_RX_FIFO_EMPTY 0x00000100 + +/* USP RX FIFO Data Register */ + +#define USP_RX_FIFO_DATA_MASK 0xFFFFFFFF +#define USP_RX_FIFO_DATA_OFFSET 0 + +/* + * When rx thd irq occur, sender just disable tx empty irq, + * Remaining data in tx fifo wil also be sent out. + */ +#define USP_FIFO_SIZE 128 +#define USP_TX_FIFO_THRESHOLD (USP_FIFO_SIZE/2) +#define USP_RX_FIFO_THRESHOLD (USP_FIFO_SIZE/2) + +/* FIFO_WIDTH for the USP_TX_FIFO_CTRL and USP_RX_FIFO_CTRL registers */ +#define USP_FIFO_WIDTH_BYTE 0x00 +#define USP_FIFO_WIDTH_WORD 0x01 +#define USP_FIFO_WIDTH_DWORD 0x02 + +#define USP_ASYNC_DIV2 16 + +#define USP_PLUGOUT_RETRY_CNT 2 + +#define USP_TX_RX_FIFO_WIDTH_DWORD 2 + +#define SIRF_USP_DIV_MCLK 0 + +#define SIRF_USP_I2S_TFS_SYNC 0 +#define SIRF_USP_I2S_RFS_SYNC 1 +#endif diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 00e70b6c7da2..a9f82b5aba9d 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -78,7 +78,7 @@ int snd_soc_cache_init(struct snd_soc_codec *codec) mutex_init(&codec->cache_rw_mutex); dev_dbg(codec->dev, "ASoC: Initializing cache for %s codec\n", - codec->name); + codec->component.name); if (codec_drv->reg_cache_default) codec->reg_cache = kmemdup(codec_drv->reg_cache_default, @@ -98,8 +98,7 @@ int snd_soc_cache_init(struct snd_soc_codec *codec) int snd_soc_cache_exit(struct snd_soc_codec *codec) { dev_dbg(codec->dev, "ASoC: Destroying cache for %s codec\n", - codec->name); - + codec->component.name); kfree(codec->reg_cache); codec->reg_cache = NULL; return 0; @@ -192,7 +191,7 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return 0; dev_dbg(codec->dev, "ASoC: Syncing cache for %s codec\n", - codec->name); + codec->component.name); trace_snd_soc_cache_sync(codec, name, "start"); ret = snd_soc_flat_cache_sync(codec); if (!ret) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 10f7f1da2aca..27c06acce205 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -37,7 +37,8 @@ static int soc_compr_open(struct snd_compr_stream *cstream) if (platform->driver->compr_ops && platform->driver->compr_ops->open) { ret = platform->driver->compr_ops->open(cstream); if (ret < 0) { - pr_err("compress asoc: can't open platform %s\n", platform->name); + pr_err("compress asoc: can't open platform %s\n", + platform->component.name); goto out; } } @@ -84,7 +85,8 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) if (platform->driver->compr_ops && platform->driver->compr_ops->open) { ret = platform->driver->compr_ops->open(cstream); if (ret < 0) { - pr_err("compress asoc: can't open platform %s\n", platform->name); + pr_err("compress asoc: can't open platform %s\n", + platform->component.name); goto out; } } @@ -627,6 +629,11 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) char new_name[64]; int ret = 0, direction = 0; + if (rtd->num_codecs > 1) { + dev_err(rtd->card->dev, "Multicodec not supported for compressed stream\n"); + return -EINVAL; + } + /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); @@ -680,7 +687,7 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) ret = snd_compress_new(rtd->card->snd_card, num, direction, compr); if (ret < 0) { pr_err("compress asoc: can't create compress for codec %s\n", - codec->name); + codec->component.name); goto compr_err; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b87d7d882e6d..d4bfd4a9076f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -270,12 +270,33 @@ static const struct file_operations codec_reg_fops = { .llseek = default_llseek, }; +static struct dentry *soc_debugfs_create_dir(struct dentry *parent, + const char *fmt, ...) +{ + struct dentry *de; + va_list ap; + char *s; + + va_start(ap, fmt); + s = kvasprintf(GFP_KERNEL, fmt, ap); + va_end(ap); + + if (!s) + return NULL; + + de = debugfs_create_dir(s, parent); + kfree(s); + + return de; +} + static void soc_init_codec_debugfs(struct snd_soc_codec *codec) { - struct dentry *debugfs_card_root = codec->card->debugfs_card_root; + struct dentry *debugfs_card_root = codec->component.card->debugfs_card_root; - codec->debugfs_codec_root = debugfs_create_dir(codec->name, - debugfs_card_root); + codec->debugfs_codec_root = soc_debugfs_create_dir(debugfs_card_root, + "codec:%s", + codec->component.name); if (!codec->debugfs_codec_root) { dev_warn(codec->dev, "ASoC: Failed to create codec debugfs directory\n"); @@ -304,17 +325,18 @@ static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) static void soc_init_platform_debugfs(struct snd_soc_platform *platform) { - struct dentry *debugfs_card_root = platform->card->debugfs_card_root; + struct dentry *debugfs_card_root = platform->component.card->debugfs_card_root; - platform->debugfs_platform_root = debugfs_create_dir(platform->name, - debugfs_card_root); + platform->debugfs_platform_root = soc_debugfs_create_dir(debugfs_card_root, + "platform:%s", + platform->component.name); if (!platform->debugfs_platform_root) { dev_warn(platform->dev, "ASoC: Failed to create platform debugfs directory\n"); return; } - snd_soc_dapm_debugfs_init(&platform->dapm, + snd_soc_dapm_debugfs_init(&platform->component.dapm, platform->debugfs_platform_root); } @@ -335,7 +357,7 @@ static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, list_for_each_entry(codec, &codec_list, list) { len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", - codec->name); + codec->component.name); if (len >= 0) ret += len; if (ret > PAGE_SIZE) { @@ -406,7 +428,7 @@ static ssize_t platform_list_read_file(struct file *file, list_for_each_entry(platform, &platform_list, list) { len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", - platform->name); + platform->component.name); if (len >= 0) ret += len; if (ret > PAGE_SIZE) { @@ -524,11 +546,12 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) int err; codec->ac97->dev.bus = &ac97_bus_type; - codec->ac97->dev.parent = codec->card->dev; + codec->ac97->dev.parent = codec->component.card->dev; codec->ac97->dev.release = soc_ac97_device_release; dev_set_name(&codec->ac97->dev, "%d-%d:%s", - codec->card->snd_card->number, 0, codec->name); + codec->component.card->snd_card->number, 0, + codec->component.name); err = device_register(&codec->ac97->dev); if (err < 0) { dev_err(codec->dev, "ASoC: Can't register ac97 bus\n"); @@ -554,7 +577,7 @@ int snd_soc_suspend(struct device *dev) { struct snd_soc_card *card = dev_get_drvdata(dev); struct snd_soc_codec *codec; - int i; + int i, j; /* If the initialization of this soc device failed, there is no codec * associated with it. Just bail out in this case. @@ -574,14 +597,17 @@ int snd_soc_suspend(struct device *dev) /* mute any active DACs */ for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *dai = card->rtd[i].codec_dai; - struct snd_soc_dai_driver *drv = dai->driver; if (card->rtd[i].dai_link->ignore_suspend) continue; - if (drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, 1); + for (j = 0; j < card->rtd[i].num_codecs; j++) { + struct snd_soc_dai *dai = card->rtd[i].codec_dais[j]; + struct snd_soc_dai_driver *drv = dai->driver; + + if (drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, 1); + } } /* suspend all pcms */ @@ -612,8 +638,12 @@ int snd_soc_suspend(struct device *dev) /* close any waiting streams and save state */ for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai **codec_dais = card->rtd[i].codec_dais; flush_delayed_work(&card->rtd[i].delayed_work); - card->rtd[i].codec->dapm.suspend_bias_level = card->rtd[i].codec->dapm.bias_level; + for (j = 0; j < card->rtd[i].num_codecs; j++) { + codec_dais[j]->codec->dapm.suspend_bias_level = + codec_dais[j]->codec->dapm.bias_level; + } } for (i = 0; i < card->num_rtd; i++) { @@ -697,7 +727,7 @@ static void soc_resume_deferred(struct work_struct *work) struct snd_soc_card *card = container_of(work, struct snd_soc_card, deferred_resume_work); struct snd_soc_codec *codec; - int i; + int i, j; /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, * so userspace apps are blocked from touching us @@ -758,14 +788,17 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *dai = card->rtd[i].codec_dai; - struct snd_soc_dai_driver *drv = dai->driver; if (card->rtd[i].dai_link->ignore_suspend) continue; - if (drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, 0); + for (j = 0; j < card->rtd[i].num_codecs; j++) { + struct snd_soc_dai *dai = card->rtd[i].codec_dais[j]; + struct snd_soc_dai_driver *drv = dai->driver; + + if (drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, 0); + } } for (i = 0; i < card->num_rtd; i++) { @@ -810,12 +843,19 @@ int snd_soc_resume(struct device *dev) /* activate pins from sleep state */ for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + struct snd_soc_dai **codec_dais = rtd->codec_dais; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int j; + if (cpu_dai->active) pinctrl_pm_select_default_state(cpu_dai->dev); - if (codec_dai->active) - pinctrl_pm_select_default_state(codec_dai->dev); + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = codec_dais[j]; + if (codec_dai->active) + pinctrl_pm_select_default_state(codec_dai->dev); + } } /* AC97 devices might have other drivers hanging off them so @@ -847,8 +887,9 @@ EXPORT_SYMBOL_GPL(snd_soc_resume); static const struct snd_soc_dai_ops null_dai_ops = { }; -static struct snd_soc_codec *soc_find_codec(const struct device_node *codec_of_node, - const char *codec_name) +static struct snd_soc_codec *soc_find_codec( + const struct device_node *codec_of_node, + const char *codec_name) { struct snd_soc_codec *codec; @@ -857,7 +898,7 @@ static struct snd_soc_codec *soc_find_codec(const struct device_node *codec_of_n if (codec->dev->of_node != codec_of_node) continue; } else { - if (strcmp(codec->name, codec_name)) + if (strcmp(codec->component.name, codec_name)) continue; } @@ -886,9 +927,12 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_component *component; + struct snd_soc_dai_link_component *codecs = dai_link->codecs; + struct snd_soc_dai **codec_dais = rtd->codec_dais; struct snd_soc_platform *platform; struct snd_soc_dai *cpu_dai; const char *platform_name; + int i; dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num); @@ -915,24 +959,30 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) return -EPROBE_DEFER; } - /* Find CODEC from registered list */ - rtd->codec = soc_find_codec(dai_link->codec_of_node, - dai_link->codec_name); - if (!rtd->codec) { - dev_err(card->dev, "ASoC: CODEC %s not registered\n", - dai_link->codec_name); - return -EPROBE_DEFER; - } + rtd->num_codecs = dai_link->num_codecs; - /* Find CODEC DAI from registered list */ - rtd->codec_dai = soc_find_codec_dai(rtd->codec, - dai_link->codec_dai_name); - if (!rtd->codec_dai) { - dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", - dai_link->codec_dai_name); - return -EPROBE_DEFER; + /* Find CODEC from registered CODECs */ + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_codec *codec; + codec = soc_find_codec(codecs[i].of_node, codecs[i].name); + if (!codec) { + dev_err(card->dev, "ASoC: CODEC %s not registered\n", + codecs[i].name); + return -EPROBE_DEFER; + } + + codec_dais[i] = soc_find_codec_dai(codec, codecs[i].dai_name); + if (!codec_dais[i]) { + dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", + codecs[i].dai_name); + return -EPROBE_DEFER; + } } + /* Single codec links expect codec and codec_dai in runtime data */ + rtd->codec_dai = codec_dais[0]; + rtd->codec = rtd->codec_dai->codec; + /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; if (!platform_name && !dai_link->platform_of_node) @@ -945,7 +995,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) dai_link->platform_of_node) continue; } else { - if (strcmp(platform->name, platform_name)) + if (strcmp(platform->component.name, platform_name)) continue; } @@ -974,11 +1024,10 @@ static int soc_remove_platform(struct snd_soc_platform *platform) } /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&platform->dapm); + snd_soc_dapm_free(&platform->component.dapm); soc_cleanup_platform_debugfs(platform); platform->probed = 0; - list_del(&platform->card_list); module_put(platform->dev->driver->owner); return 0; @@ -1023,8 +1072,8 @@ static void soc_remove_codec_dai(struct snd_soc_dai *codec_dai, int order) static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; - int err; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int i, err; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -1035,7 +1084,8 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) } /* remove the CODEC DAI */ - soc_remove_codec_dai(codec_dai, order); + for (i = 0; i < rtd->num_codecs; i++) + soc_remove_codec_dai(rtd->codec_dais[i], order); /* remove the cpu_dai */ if (cpu_dai && cpu_dai->probed && @@ -1048,11 +1098,8 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) cpu_dai->name, err); } cpu_dai->probed = 0; - - if (!cpu_dai->codec) { - snd_soc_dapm_free(&cpu_dai->dapm); + if (!cpu_dai->codec) module_put(cpu_dai->dev->driver->owner); - } } } @@ -1061,9 +1108,9 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_codec *codec; + int i; /* remove the platform */ if (platform && platform->probed && @@ -1072,8 +1119,8 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, } /* remove the CODEC-side CODEC */ - if (codec_dai) { - codec = codec_dai->codec; + for (i = 0; i < rtd->num_codecs; i++) { + codec = rtd->codec_dais[i]->codec; if (codec && codec->probed && codec->driver->remove_order == order) soc_remove_codec(codec); @@ -1108,7 +1155,7 @@ static void soc_remove_dai_links(struct snd_soc_card *card) } static void soc_set_name_prefix(struct snd_soc_card *card, - struct snd_soc_codec *codec) + struct snd_soc_component *component) { int i; @@ -1117,11 +1164,11 @@ static void soc_set_name_prefix(struct snd_soc_card *card, for (i = 0; i < card->num_configs; i++) { struct snd_soc_codec_conf *map = &card->codec_conf[i]; - if (map->of_node && codec->dev->of_node != map->of_node) + if (map->of_node && component->dev->of_node != map->of_node) continue; - if (map->dev_name && strcmp(codec->name, map->dev_name)) + if (map->dev_name && strcmp(component->name, map->dev_name)) continue; - codec->name_prefix = map->name_prefix; + component->name_prefix = map->name_prefix; break; } } @@ -1133,9 +1180,9 @@ static int soc_probe_codec(struct snd_soc_card *card, const struct snd_soc_codec_driver *driver = codec->driver; struct snd_soc_dai *dai; - codec->card = card; + codec->component.card = card; codec->dapm.card = card; - soc_set_name_prefix(card, codec); + soc_set_name_prefix(card, &codec->component); if (!try_module_get(codec->dev->driver->owner)) return -ENODEV; @@ -1177,7 +1224,7 @@ static int soc_probe_codec(struct snd_soc_card *card, WARN(codec->dapm.idle_bias_off && codec->dapm.bias_level != SND_SOC_BIAS_OFF, "codec %s can not start from non-off bias with idle_bias_off==1\n", - codec->name); + codec->component.name); } if (driver->controls) @@ -1209,8 +1256,8 @@ static int soc_probe_platform(struct snd_soc_card *card, struct snd_soc_component *component; struct snd_soc_dai *dai; - platform->card = card; - platform->dapm.card = card; + platform->component.card = card; + platform->component.dapm.card = card; if (!try_module_get(platform->dev->driver->owner)) return -ENODEV; @@ -1218,7 +1265,7 @@ static int soc_probe_platform(struct snd_soc_card *card, soc_init_platform_debugfs(platform); if (driver->dapm_widgets) - snd_soc_dapm_new_controls(&platform->dapm, + snd_soc_dapm_new_controls(&platform->component.dapm, driver->dapm_widgets, driver->num_dapm_widgets); /* Create DAPM widgets for each DAI stream */ @@ -1226,10 +1273,11 @@ static int soc_probe_platform(struct snd_soc_card *card, if (component->dev != platform->dev) continue; list_for_each_entry(dai, &component->dai_list, list) - snd_soc_dapm_new_dai_widgets(&platform->dapm, dai); + snd_soc_dapm_new_dai_widgets(&platform->component.dapm, + dai); } - platform->dapm.idle_bias_off = 1; + platform->component.dapm.idle_bias_off = 1; if (driver->probe) { ret = driver->probe(platform); @@ -1244,13 +1292,12 @@ static int soc_probe_platform(struct snd_soc_card *card, snd_soc_add_platform_controls(platform, driver->controls, driver->num_controls); if (driver->dapm_routes) - snd_soc_dapm_add_routes(&platform->dapm, driver->dapm_routes, - driver->num_dapm_routes); + snd_soc_dapm_add_routes(&platform->component.dapm, + driver->dapm_routes, driver->num_dapm_routes); /* mark platform as probed and add to card platform list */ platform->probed = 1; - list_add(&platform->card_list, &card->platform_dev_list); - list_add(&platform->dapm.list, &card->dapm_list); + list_add(&platform->component.dapm.list, &card->dapm_list); return 0; @@ -1266,83 +1313,17 @@ static void rtd_release(struct device *dev) kfree(dev); } -static int soc_aux_dev_init(struct snd_soc_card *card, - struct snd_soc_codec *codec, - int num) -{ - struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - int ret; - - rtd->card = card; - - /* do machine specific initialization */ - if (aux_dev->init) { - ret = aux_dev->init(&codec->dapm); - if (ret < 0) - return ret; - } - - rtd->codec = codec; - - return 0; -} - -static int soc_dai_link_init(struct snd_soc_card *card, - struct snd_soc_codec *codec, - int num) +static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, + const char *name) { - struct snd_soc_dai_link *dai_link = &card->dai_link[num]; - struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - int ret; - - rtd->card = card; - - /* do machine specific initialization */ - if (dai_link->init) { - ret = dai_link->init(rtd); - if (ret < 0) - return ret; - } - - rtd->codec = codec; - - return 0; -} - -static int soc_post_component_init(struct snd_soc_card *card, - struct snd_soc_codec *codec, - int num, int dailess) -{ - struct snd_soc_dai_link *dai_link = NULL; - struct snd_soc_aux_dev *aux_dev = NULL; - struct snd_soc_pcm_runtime *rtd; - const char *name; int ret = 0; - if (!dailess) { - dai_link = &card->dai_link[num]; - rtd = &card->rtd[num]; - name = dai_link->name; - ret = soc_dai_link_init(card, codec, num); - } else { - aux_dev = &card->aux_dev[num]; - rtd = &card->rtd_aux[num]; - name = aux_dev->name; - ret = soc_aux_dev_init(card, codec, num); - } - - if (ret < 0) { - dev_err(card->dev, "ASoC: failed to init %s: %d\n", name, ret); - return ret; - } - /* register the rtd device */ rtd->dev = kzalloc(sizeof(struct device), GFP_KERNEL); if (!rtd->dev) return -ENOMEM; device_initialize(rtd->dev); - rtd->dev->parent = card->dev; + rtd->dev->parent = rtd->card->dev; rtd->dev->release = rtd_release; rtd->dev->init_name = name; dev_set_drvdata(rtd->dev, rtd); @@ -1355,7 +1336,7 @@ static int soc_post_component_init(struct snd_soc_card *card, if (ret < 0) { /* calling put_device() here to free the rtd->dev */ put_device(rtd->dev); - dev_err(card->dev, + dev_err(rtd->card->dev, "ASoC: failed to register runtime device: %d\n", ret); return ret; } @@ -1364,26 +1345,15 @@ static int soc_post_component_init(struct snd_soc_card *card, /* add DAPM sysfs entries for this codec */ ret = snd_soc_dapm_sys_add(rtd->dev); if (ret < 0) - dev_err(codec->dev, + dev_err(rtd->dev, "ASoC: failed to add codec dapm sysfs entries: %d\n", ret); /* add codec sysfs entries */ ret = device_create_file(rtd->dev, &dev_attr_codec_reg); if (ret < 0) - dev_err(codec->dev, + dev_err(rtd->dev, "ASoC: failed to add codec sysfs files: %d\n", ret); -#ifdef CONFIG_DEBUG_FS - /* add DPCM sysfs entries */ - if (!dailess && !dai_link->dynamic) - goto out; - - ret = soc_dpcm_debugfs_add(rtd); - if (ret < 0) - dev_err(rtd->dev, "ASoC: failed to add dpcm sysfs entries: %d\n", ret); - -out: -#endif return 0; } @@ -1392,9 +1362,8 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_platform *platform = rtd->platform; - int ret; + int i, ret; /* probe the CPU-side component, if it is a CODEC */ if (cpu_dai->codec && @@ -1405,12 +1374,14 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, return ret; } - /* probe the CODEC-side component */ - if (!codec_dai->codec->probed && - codec_dai->codec->driver->probe_order == order) { - ret = soc_probe_codec(card, codec_dai->codec); - if (ret < 0) - return ret; + /* probe the CODEC-side components */ + for (i = 0; i < rtd->num_codecs; i++) { + if (!rtd->codec_dais[i]->codec->probed && + rtd->codec_dais[i]->codec->driver->probe_order == order) { + ret = soc_probe_codec(card, rtd->codec_dais[i]->codec); + if (ret < 0) + return ret; + } } /* probe the platform */ @@ -1450,12 +1421,16 @@ static int soc_probe_codec_dai(struct snd_soc_card *card, static int soc_link_dai_widgets(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link, - struct snd_soc_dai *cpu_dai, - struct snd_soc_dai *codec_dai) + struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dapm_widget *play_w, *capture_w; int ret; + if (rtd->num_codecs > 1) + dev_warn(card->dev, "ASoC: Multiple codecs not supported yet\n"); + /* link the DAI widgets */ play_w = codec_dai->playback_widget; capture_w = cpu_dai->capture_widget; @@ -1488,19 +1463,18 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret; + int i, ret; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, num, order); /* config components */ cpu_dai->platform = platform; - codec_dai->card = card; cpu_dai->card = card; + for (i = 0; i < rtd->num_codecs; i++) + rtd->codec_dais[i]->card = card; /* set default power off timeout */ rtd->pmdown_time = pmdown_time; @@ -1509,11 +1483,8 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) if (!cpu_dai->probed && cpu_dai->driver->probe_order == order) { if (!cpu_dai->codec) { - cpu_dai->dapm.card = card; if (!try_module_get(cpu_dai->dev->driver->owner)) return -ENODEV; - - list_add(&cpu_dai->dapm.list, &card->dapm_list); } if (cpu_dai->driver->probe) { @@ -1530,18 +1501,43 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } /* probe the CODEC DAI */ - ret = soc_probe_codec_dai(card, codec_dai, order); - if (ret) - return ret; + for (i = 0; i < rtd->num_codecs; i++) { + ret = soc_probe_codec_dai(card, rtd->codec_dais[i], order); + if (ret) + return ret; + } /* complete DAI probe during last probe */ if (order != SND_SOC_COMP_ORDER_LAST) return 0; - ret = soc_post_component_init(card, codec, num, 0); + /* do machine specific initialization */ + if (dai_link->init) { + ret = dai_link->init(rtd); + if (ret < 0) { + dev_err(card->dev, "ASoC: failed to init %s: %d\n", + dai_link->name, ret); + return ret; + } + } + + ret = soc_post_component_init(rtd, dai_link->name); if (ret) return ret; +#ifdef CONFIG_DEBUG_FS + /* add DPCM sysfs entries */ + if (dai_link->dynamic) { + ret = soc_dpcm_debugfs_add(rtd); + if (ret < 0) { + dev_err(rtd->dev, + "ASoC: failed to add dpcm sysfs entries: %d\n", + ret); + return ret; + } + } +#endif + ret = device_create_file(rtd->dev, &dev_attr_pmdown_time); if (ret < 0) dev_warn(rtd->dev, "ASoC: failed to add pmdown_time sysfs: %d\n", @@ -1570,16 +1566,18 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) codec2codec_close_delayed_work); /* link the DAI widgets */ - ret = soc_link_dai_widgets(card, dai_link, - cpu_dai, codec_dai); + ret = soc_link_dai_widgets(card, dai_link, rtd); if (ret) return ret; } } /* add platform data for AC97 devices */ - if (rtd->codec_dai->driver->ac97_control) - snd_ac97_dev_add_pdata(codec->ac97, rtd->cpu_dai->ac97_pdata); + for (i = 0; i < rtd->num_codecs; i++) { + if (rtd->codec_dais[i]->driver->ac97_control) + snd_ac97_dev_add_pdata(rtd->codec_dais[i]->codec->ac97, + rtd->cpu_dai->ac97_pdata); + } return 0; } @@ -1617,11 +1615,6 @@ static int soc_register_ac97_codec(struct snd_soc_codec *codec, return 0; } -static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) -{ - return soc_register_ac97_codec(rtd->codec, rtd->codec_dai); -} - static void soc_unregister_ac97_codec(struct snd_soc_codec *codec) { if (codec->ac97_registered) { @@ -1630,74 +1623,77 @@ static void soc_unregister_ac97_codec(struct snd_soc_codec *codec) } } -static void soc_unregister_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) +static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) { - soc_unregister_ac97_codec(rtd->codec); -} -#endif + int i, ret; -static struct snd_soc_codec *soc_find_matching_codec(struct snd_soc_card *card, - int num) -{ - struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - struct snd_soc_codec *codec; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; - /* find CODEC from registered CODECs */ - list_for_each_entry(codec, &codec_list, list) { - if (aux_dev->codec_of_node && - (codec->dev->of_node != aux_dev->codec_of_node)) - continue; - if (aux_dev->codec_name && strcmp(codec->name, aux_dev->codec_name)) - continue; - return codec; + ret = soc_register_ac97_codec(codec_dai->codec, codec_dai); + if (ret) { + while (--i >= 0) + soc_unregister_ac97_codec(codec_dai->codec); + return ret; + } } - return NULL; + return 0; } -static int soc_check_aux_dev(struct snd_soc_card *card, int num) +static void soc_unregister_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - const char *codecname = aux_dev->codec_name; - struct snd_soc_codec *codec = soc_find_matching_codec(card, num); - - if (codec) - return 0; - if (aux_dev->codec_of_node) - codecname = of_node_full_name(aux_dev->codec_of_node); + int i; - dev_err(card->dev, "ASoC: %s not registered\n", codecname); - return -EPROBE_DEFER; + for (i = 0; i < rtd->num_codecs; i++) + soc_unregister_ac97_codec(rtd->codec_dais[i]->codec); } +#endif -static int soc_probe_aux_dev(struct snd_soc_card *card, int num) +static int soc_bind_aux_dev(struct snd_soc_card *card, int num) { + struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; const char *codecname = aux_dev->codec_name; - int ret = -ENODEV; - struct snd_soc_codec *codec = soc_find_matching_codec(card, num); - if (!codec) { + rtd->codec = soc_find_codec(aux_dev->codec_of_node, codecname); + if (!rtd->codec) { if (aux_dev->codec_of_node) codecname = of_node_full_name(aux_dev->codec_of_node); - /* codec not found */ - dev_err(card->dev, "ASoC: codec %s not found", codecname); + dev_err(card->dev, "ASoC: %s not registered\n", codecname); return -EPROBE_DEFER; } - if (codec->probed) { - dev_err(codec->dev, "ASoC: codec already probed"); + return 0; +} + +static int soc_probe_aux_dev(struct snd_soc_card *card, int num) +{ + struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; + struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; + int ret; + + if (rtd->codec->probed) { + dev_err(rtd->codec->dev, "ASoC: codec already probed\n"); return -EBUSY; } - ret = soc_probe_codec(card, codec); + ret = soc_probe_codec(card, rtd->codec); if (ret < 0) return ret; - ret = soc_post_component_init(card, codec, num, 1); + /* do machine specific initialization */ + if (aux_dev->init) { + ret = aux_dev->init(&rtd->codec->dapm); + if (ret < 0) { + dev_err(card->dev, "ASoC: failed to init %s: %d\n", + aux_dev->name, ret); + return ret; + } + } - return ret; + return soc_post_component_init(rtd, aux_dev->name); } static void soc_remove_aux_dev(struct snd_soc_card *card, int num) @@ -1749,9 +1745,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) goto base_error; } - /* check aux_devs too */ + /* bind aux_devs too */ for (i = 0; i < card->num_aux_devs; i++) { - ret = soc_check_aux_dev(card, i); + ret = soc_bind_aux_dev(card, i); if (ret != 0) goto base_error; } @@ -1849,16 +1845,23 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) card->num_dapm_routes); for (i = 0; i < card->num_links; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; dai_link = &card->dai_link[i]; dai_fmt = dai_link->dai_fmt; if (dai_fmt) { - ret = snd_soc_dai_set_fmt(card->rtd[i].codec_dai, - dai_fmt); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(card->rtd[i].codec_dai->dev, - "ASoC: Failed to set DAI format: %d\n", - ret); + struct snd_soc_dai **codec_dais = rtd->codec_dais; + int j; + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = codec_dais[j]; + + ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(codec_dai->dev, + "ASoC: Failed to set DAI format: %d\n", + ret); + } } /* If this is a regular CPU link there will be a platform */ @@ -1927,8 +1930,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } if (card->fully_routed) - list_for_each_entry(codec, &card->codec_dev_list, card_list) - snd_soc_dapm_auto_nc_codec_pins(codec); + snd_soc_dapm_auto_nc_pins(card); snd_soc_dapm_new_widgets(card); @@ -2058,10 +2060,15 @@ int snd_soc_poweroff(struct device *dev) /* deactivate pins to sleep state */ for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; - pinctrl_pm_select_sleep_state(codec_dai->dev); + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int j; + pinctrl_pm_select_sleep_state(cpu_dai->dev); + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + pinctrl_pm_select_sleep_state(codec_dai->dev); + } } return 0; @@ -2387,6 +2394,25 @@ struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol); /** + * snd_soc_add_component_controls - Add an array of controls to a component. + * + * @component: Component to add controls to + * @controls: Array of controls to add + * @num_controls: Number of elements in the array + * + * Return: 0 for success, else error. + */ +int snd_soc_add_component_controls(struct snd_soc_component *component, + const struct snd_kcontrol_new *controls, unsigned int num_controls) +{ + struct snd_card *card = component->card->snd_card; + + return snd_soc_add_controls(card, component->dev, controls, + num_controls, component->name_prefix, component); +} +EXPORT_SYMBOL_GPL(snd_soc_add_component_controls); + +/** * snd_soc_add_codec_controls - add an array of controls to a codec. * Convenience function to add a list of controls. Many codecs were * duplicating this code. @@ -2398,12 +2424,10 @@ EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol); * Return 0 for success, else error. */ int snd_soc_add_codec_controls(struct snd_soc_codec *codec, - const struct snd_kcontrol_new *controls, int num_controls) + const struct snd_kcontrol_new *controls, unsigned int num_controls) { - struct snd_card *card = codec->card->snd_card; - - return snd_soc_add_controls(card, codec->dev, controls, num_controls, - codec->name_prefix, &codec->component); + return snd_soc_add_component_controls(&codec->component, controls, + num_controls); } EXPORT_SYMBOL_GPL(snd_soc_add_codec_controls); @@ -2418,12 +2442,10 @@ EXPORT_SYMBOL_GPL(snd_soc_add_codec_controls); * Return 0 for success, else error. */ int snd_soc_add_platform_controls(struct snd_soc_platform *platform, - const struct snd_kcontrol_new *controls, int num_controls) + const struct snd_kcontrol_new *controls, unsigned int num_controls) { - struct snd_card *card = platform->card->snd_card; - - return snd_soc_add_controls(card, platform->dev, controls, num_controls, - NULL, &platform->component); + return snd_soc_add_component_controls(&platform->component, controls, + num_controls); } EXPORT_SYMBOL_GPL(snd_soc_add_platform_controls); @@ -3095,7 +3117,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); int snd_soc_limit_volume(struct snd_soc_codec *codec, const char *name, int max) { - struct snd_card *card = codec->card->snd_card; + struct snd_card *card = codec->component.card->snd_card; struct snd_kcontrol *kctl; struct soc_mixer_control *mc; int found = 0; @@ -3267,6 +3289,27 @@ int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_bytes_info_ext); +int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + unsigned int count = size < params->max ? size : params->max; + int ret = -ENXIO; + + switch (op_flag) { + case SNDRV_CTL_TLV_OP_READ: + if (params->get) + ret = params->get(tlv, count); + break; + case SNDRV_CTL_TLV_OP_WRITE: + if (params->put) + ret = params->put(tlv, count); + break; + } + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_tlv_callback); + /** * snd_soc_info_xr_sx - signed multi register info callback * @kcontrol: mreg control @@ -3641,6 +3684,9 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, else snd_soc_xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask); + dai->tx_mask = tx_mask; + dai->rx_mask = rx_mask; + if (dai->driver && dai->driver->ops->set_tdm_slot) return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask, slots, slot_width); @@ -3713,6 +3759,33 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); +static int snd_soc_init_multicodec(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + /* Legacy codec/codec_dai link is a single entry in multicodec */ + if (dai_link->codec_name || dai_link->codec_of_node || + dai_link->codec_dai_name) { + dai_link->num_codecs = 1; + + dai_link->codecs = devm_kzalloc(card->dev, + sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!dai_link->codecs) + return -ENOMEM; + + dai_link->codecs[0].name = dai_link->codec_name; + dai_link->codecs[0].of_node = dai_link->codec_of_node; + dai_link->codecs[0].dai_name = dai_link->codec_dai_name; + } + + if (!dai_link->codecs) { + dev_err(card->dev, "ASoC: DAI link has no CODECs\n"); + return -EINVAL; + } + + return 0; +} + /** * snd_soc_register_card - Register a card with the ASoC core * @@ -3721,7 +3794,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); */ int snd_soc_register_card(struct snd_soc_card *card) { - int i, ret; + int i, j, ret; if (!card->name || !card->dev) return -EINVAL; @@ -3729,22 +3802,29 @@ int snd_soc_register_card(struct snd_soc_card *card) for (i = 0; i < card->num_links; i++) { struct snd_soc_dai_link *link = &card->dai_link[i]; - /* - * Codec must be specified by 1 of name or OF node, - * not both or neither. - */ - if (!!link->codec_name == !!link->codec_of_node) { - dev_err(card->dev, - "ASoC: Neither/both codec name/of_node are set for %s\n", - link->name); - return -EINVAL; + ret = snd_soc_init_multicodec(card, link); + if (ret) { + dev_err(card->dev, "ASoC: failed to init multicodec\n"); + return ret; } - /* Codec DAI name must be specified */ - if (!link->codec_dai_name) { - dev_err(card->dev, - "ASoC: codec_dai_name not set for %s\n", - link->name); - return -EINVAL; + + for (j = 0; j < link->num_codecs; j++) { + /* + * Codec must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->codecs[j].name == + !!link->codecs[j].of_node) { + dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + /* Codec DAI name must be specified */ + if (!link->codecs[j].dai_name) { + dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n", + link->name); + return -EINVAL; + } } /* @@ -3797,8 +3877,19 @@ int snd_soc_register_card(struct snd_soc_card *card) card->num_rtd = 0; card->rtd_aux = &card->rtd[card->num_links]; - for (i = 0; i < card->num_links; i++) + for (i = 0; i < card->num_links; i++) { + card->rtd[i].card = card; card->rtd[i].dai_link = &card->dai_link[i]; + card->rtd[i].codec_dais = devm_kzalloc(card->dev, + sizeof(struct snd_soc_dai *) * + (card->rtd[i].dai_link->num_codecs), + GFP_KERNEL); + if (card->rtd[i].codec_dais == NULL) + return -ENOMEM; + } + + for (i = 0; i < card->num_aux_devs; i++) + card->rtd_aux[i].card = card; INIT_LIST_HEAD(&card->dapm_dirty); card->instantiated = 0; @@ -3811,10 +3902,16 @@ int snd_soc_register_card(struct snd_soc_card *card) /* deactivate pins to sleep state */ for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int j; + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + } + if (!cpu_dai->active) pinctrl_pm_select_sleep_state(cpu_dai->dev); } @@ -3921,16 +4018,14 @@ static void snd_soc_unregister_dais(struct snd_soc_component *component) * snd_soc_register_dais - Register a DAI with the ASoC core * * @component: The component the DAIs are registered for - * @codec: The CODEC that the DAIs are registered for, NULL if the component is - * not a CODEC. * @dai_drv: DAI driver to use for the DAIs * @count: Number of DAIs * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the * parent's name. */ static int snd_soc_register_dais(struct snd_soc_component *component, - struct snd_soc_codec *codec, struct snd_soc_dai_driver *dai_drv, - size_t count, bool legacy_dai_naming) + struct snd_soc_dai_driver *dai_drv, size_t count, + bool legacy_dai_naming) { struct device *dev = component->dev; struct snd_soc_dai *dai; @@ -3939,6 +4034,9 @@ static int snd_soc_register_dais(struct snd_soc_component *component, dev_dbg(dev, "ASoC: dai register %s #%Zu\n", dev_name(dev), count); + component->dai_drv = dai_drv; + component->num_dai = count; + for (i = 0; i < count; i++) { dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL); @@ -3971,16 +4069,11 @@ static int snd_soc_register_dais(struct snd_soc_component *component, } dai->component = component; - dai->codec = codec; dai->dev = dev; dai->driver = &dai_drv[i]; - dai->dapm.dev = dev; if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; - if (!dai->codec) - dai->dapm.idle_bias_off = 1; - list_add(&dai->list, &component->dai_list); dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); @@ -3994,60 +4087,82 @@ err: return ret; } -/** - * snd_soc_register_component - Register a component with the ASoC core - * - */ -static int -__snd_soc_register_component(struct device *dev, - struct snd_soc_component *cmpnt, - const struct snd_soc_component_driver *cmpnt_drv, - struct snd_soc_codec *codec, - struct snd_soc_dai_driver *dai_drv, - int num_dai, bool allow_single_dai) +static void snd_soc_component_seq_notifier(struct snd_soc_dapm_context *dapm, + enum snd_soc_dapm_type type, int subseq) { - int ret; + struct snd_soc_component *component = dapm->component; - dev_dbg(dev, "component register %s\n", dev_name(dev)); + component->driver->seq_notifier(component, type, subseq); +} - if (!cmpnt) { - dev_err(dev, "ASoC: Failed to connecting component\n"); - return -ENOMEM; - } +static int snd_soc_component_stream_event(struct snd_soc_dapm_context *dapm, + int event) +{ + struct snd_soc_component *component = dapm->component; - mutex_init(&cmpnt->io_mutex); + return component->driver->stream_event(component, event); +} + +static int snd_soc_component_initialize(struct snd_soc_component *component, + const struct snd_soc_component_driver *driver, struct device *dev) +{ + struct snd_soc_dapm_context *dapm; - cmpnt->name = fmt_single_name(dev, &cmpnt->id); - if (!cmpnt->name) { - dev_err(dev, "ASoC: Failed to simplifying name\n"); + component->name = fmt_single_name(dev, &component->id); + if (!component->name) { + dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; } - cmpnt->dev = dev; - cmpnt->driver = cmpnt_drv; - cmpnt->dai_drv = dai_drv; - cmpnt->num_dai = num_dai; - INIT_LIST_HEAD(&cmpnt->dai_list); + component->dev = dev; + component->driver = driver; - ret = snd_soc_register_dais(cmpnt, codec, dai_drv, num_dai, - allow_single_dai); - if (ret < 0) { - dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); - goto error_component_name; - } + if (!component->dapm_ptr) + component->dapm_ptr = &component->dapm; + + dapm = component->dapm_ptr; + dapm->dev = dev; + dapm->component = component; + dapm->bias_level = SND_SOC_BIAS_OFF; + if (driver->seq_notifier) + dapm->seq_notifier = snd_soc_component_seq_notifier; + if (driver->stream_event) + dapm->stream_event = snd_soc_component_stream_event; + + INIT_LIST_HEAD(&component->dai_list); + mutex_init(&component->io_mutex); + return 0; +} + +static void snd_soc_component_add_unlocked(struct snd_soc_component *component) +{ + list_add(&component->list, &component_list); +} + +static void snd_soc_component_add(struct snd_soc_component *component) +{ mutex_lock(&client_mutex); - list_add(&cmpnt->list, &component_list); + snd_soc_component_add_unlocked(component); mutex_unlock(&client_mutex); +} - dev_dbg(cmpnt->dev, "ASoC: Registered component '%s'\n", cmpnt->name); - - return ret; +static void snd_soc_component_cleanup(struct snd_soc_component *component) +{ + snd_soc_unregister_dais(component); + kfree(component->name); +} -error_component_name: - kfree(cmpnt->name); +static void snd_soc_component_del_unlocked(struct snd_soc_component *component) +{ + list_del(&component->list); +} - return ret; +static void snd_soc_component_del(struct snd_soc_component *component) +{ + mutex_lock(&client_mutex); + snd_soc_component_del_unlocked(component); + mutex_unlock(&client_mutex); } int snd_soc_register_component(struct device *dev, @@ -4056,32 +4171,38 @@ int snd_soc_register_component(struct device *dev, int num_dai) { struct snd_soc_component *cmpnt; + int ret; - cmpnt = devm_kzalloc(dev, sizeof(*cmpnt), GFP_KERNEL); + cmpnt = kzalloc(sizeof(*cmpnt), GFP_KERNEL); if (!cmpnt) { dev_err(dev, "ASoC: Failed to allocate memory\n"); return -ENOMEM; } + ret = snd_soc_component_initialize(cmpnt, cmpnt_drv, dev); + if (ret) + goto err_free; + cmpnt->ignore_pmdown_time = true; cmpnt->registered_as_component = true; - return __snd_soc_register_component(dev, cmpnt, cmpnt_drv, NULL, - dai_drv, num_dai, true); -} -EXPORT_SYMBOL_GPL(snd_soc_register_component); + ret = snd_soc_register_dais(cmpnt, dai_drv, num_dai, true); + if (ret < 0) { + dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); + goto err_cleanup; + } -static void __snd_soc_unregister_component(struct snd_soc_component *cmpnt) -{ - snd_soc_unregister_dais(cmpnt); + snd_soc_component_add(cmpnt); - mutex_lock(&client_mutex); - list_del(&cmpnt->list); - mutex_unlock(&client_mutex); + return 0; - dev_dbg(cmpnt->dev, "ASoC: Unregistered component '%s'\n", cmpnt->name); - kfree(cmpnt->name); +err_cleanup: + snd_soc_component_cleanup(cmpnt); +err_free: + kfree(cmpnt); + return ret; } +EXPORT_SYMBOL_GPL(snd_soc_register_component); /** * snd_soc_unregister_component - Unregister a component from the ASoC core @@ -4098,7 +4219,9 @@ void snd_soc_unregister_component(struct device *dev) return; found: - __snd_soc_unregister_component(cmpnt); + snd_soc_component_del(cmpnt); + snd_soc_component_cleanup(cmpnt); + kfree(cmpnt); } EXPORT_SYMBOL_GPL(snd_soc_unregister_component); @@ -4131,37 +4254,25 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, { int ret; - /* create platform component name */ - platform->name = fmt_single_name(dev, &platform->id); - if (platform->name == NULL) - return -ENOMEM; + ret = snd_soc_component_initialize(&platform->component, + &platform_drv->component_driver, dev); + if (ret) + return ret; platform->dev = dev; platform->driver = platform_drv; - platform->dapm.dev = dev; - platform->dapm.platform = platform; - platform->dapm.component = &platform->component; - platform->dapm.stream_event = platform_drv->stream_event; if (platform_drv->write) platform->component.write = snd_soc_platform_drv_write; if (platform_drv->read) platform->component.read = snd_soc_platform_drv_read; - /* register component */ - ret = __snd_soc_register_component(dev, &platform->component, - &platform_drv->component_driver, - NULL, NULL, 0, false); - if (ret < 0) { - dev_err(platform->component.dev, - "ASoC: Failed to register component: %d\n", ret); - return ret; - } - mutex_lock(&client_mutex); + snd_soc_component_add_unlocked(&platform->component); list_add(&platform->list, &platform_list); mutex_unlock(&client_mutex); - dev_dbg(dev, "ASoC: Registered platform '%s'\n", platform->name); + dev_dbg(dev, "ASoC: Registered platform '%s'\n", + platform->component.name); return 0; } @@ -4198,15 +4309,16 @@ EXPORT_SYMBOL_GPL(snd_soc_register_platform); */ void snd_soc_remove_platform(struct snd_soc_platform *platform) { - __snd_soc_unregister_component(&platform->component); mutex_lock(&client_mutex); list_del(&platform->list); + snd_soc_component_del_unlocked(&platform->component); mutex_unlock(&client_mutex); + snd_soc_component_cleanup(&platform->component); + dev_dbg(platform->dev, "ASoC: Unregistered platform '%s'\n", - platform->name); - kfree(platform->name); + platform->component.name); } EXPORT_SYMBOL_GPL(snd_soc_remove_platform); @@ -4292,6 +4404,14 @@ static int snd_soc_codec_drv_read(struct snd_soc_component *component, return 0; } +static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); + + return codec->driver->set_bias_level(codec, level); +} + /** * snd_soc_register_codec - Register a codec with the ASoC core * @@ -4303,6 +4423,7 @@ int snd_soc_register_codec(struct device *dev, int num_dai) { struct snd_soc_codec *codec; + struct snd_soc_dai *dai; struct regmap *regmap; int ret, i; @@ -4312,24 +4433,23 @@ int snd_soc_register_codec(struct device *dev, if (codec == NULL) return -ENOMEM; - /* create CODEC component name */ - codec->name = fmt_single_name(dev, &codec->id); - if (codec->name == NULL) { - ret = -ENOMEM; - goto fail_codec; - } + codec->component.dapm_ptr = &codec->dapm; + + ret = snd_soc_component_initialize(&codec->component, + &codec_drv->component_driver, dev); + if (ret) + goto err_free; if (codec_drv->write) codec->component.write = snd_soc_codec_drv_write; if (codec_drv->read) codec->component.read = snd_soc_codec_drv_read; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; - codec->dapm.bias_level = SND_SOC_BIAS_OFF; - codec->dapm.dev = dev; codec->dapm.codec = codec; - codec->dapm.component = &codec->component; - codec->dapm.seq_notifier = codec_drv->seq_notifier; - codec->dapm.stream_event = codec_drv->stream_event; + if (codec_drv->seq_notifier) + codec->dapm.seq_notifier = codec_drv->seq_notifier; + if (codec_drv->set_bias_level) + codec->dapm.set_bias_level = snd_soc_codec_set_bias_level; codec->dev = dev; codec->driver = codec_drv; codec->component.val_bytes = codec_drv->reg_word_size; @@ -4348,7 +4468,7 @@ int snd_soc_register_codec(struct device *dev, dev_err(codec->dev, "Failed to set cache I/O:%d\n", ret); - return ret; + goto err_cleanup; } } } @@ -4358,29 +4478,27 @@ int snd_soc_register_codec(struct device *dev, fixup_codec_formats(&dai_drv[i].capture); } - mutex_lock(&client_mutex); - list_add(&codec->list, &codec_list); - mutex_unlock(&client_mutex); - - /* register component */ - ret = __snd_soc_register_component(dev, &codec->component, - &codec_drv->component_driver, - codec, dai_drv, num_dai, false); + ret = snd_soc_register_dais(&codec->component, dai_drv, num_dai, false); if (ret < 0) { - dev_err(codec->dev, "ASoC: Failed to regster component: %d\n", ret); - goto fail_codec_name; + dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); + goto err_cleanup; } - dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->name); - return 0; + list_for_each_entry(dai, &codec->component.dai_list, list) + dai->codec = codec; -fail_codec_name: mutex_lock(&client_mutex); - list_del(&codec->list); + snd_soc_component_add_unlocked(&codec->component); + list_add(&codec->list, &codec_list); mutex_unlock(&client_mutex); - kfree(codec->name); -fail_codec: + dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", + codec->component.name); + return 0; + +err_cleanup: + snd_soc_component_cleanup(&codec->component); +err_free: kfree(codec); return ret; } @@ -4402,16 +4520,17 @@ void snd_soc_unregister_codec(struct device *dev) return; found: - __snd_soc_unregister_component(&codec->component); mutex_lock(&client_mutex); list_del(&codec->list); + snd_soc_component_del_unlocked(&codec->component); mutex_unlock(&client_mutex); - dev_dbg(codec->dev, "ASoC: Unregistered codec '%s'\n", codec->name); + dev_dbg(codec->dev, "ASoC: Unregistered codec '%s'\n", + codec->component.name); + snd_soc_component_cleanup(&codec->component); snd_soc_cache_exit(codec); - kfree(codec->name); kfree(codec); } EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); @@ -4420,9 +4539,16 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname) { - struct device_node *np = card->dev->of_node; + struct device_node *np; int ret; + if (!card->dev) { + pr_err("card->dev is not set before calling %s\n", __func__); + return -EINVAL; + } + + np = card->dev->of_node; + ret = of_property_read_string_index(np, propname, 0, &card->name); /* * EINVAL means the property does not exist. This is fine providing diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index cdc837ed144d..8348352dc2c6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -350,12 +350,27 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, } /** + * snd_soc_dapm_kcontrol_dapm() - Returns the dapm context associated to a + * kcontrol + * @kcontrol: The kcontrol + * + * Note: This function must only be used on kcontrols that are known to have + * been registered for a CODEC. Otherwise the behaviour is undefined. + */ +struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( + struct snd_kcontrol *kcontrol) +{ + return dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->dapm; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_dapm); + +/** * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol * @kcontrol: The kcontrol */ struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol) { - return dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->codec; + return snd_soc_dapm_to_codec(snd_soc_dapm_kcontrol_dapm(kcontrol)); } EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_codec); @@ -375,23 +390,38 @@ static void dapm_reset(struct snd_soc_card *card) } } -static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg, +static const char *soc_dapm_prefix(struct snd_soc_dapm_context *dapm) +{ + if (!dapm->component) + return NULL; + return dapm->component->name_prefix; +} + +static int soc_dapm_read(struct snd_soc_dapm_context *dapm, int reg, unsigned int *value) { - if (!w->dapm->component) + if (!dapm->component) return -EIO; - return snd_soc_component_read(w->dapm->component, reg, value); + return snd_soc_component_read(dapm->component, reg, value); } -static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, +static int soc_dapm_update_bits(struct snd_soc_dapm_context *dapm, int reg, unsigned int mask, unsigned int value) { - if (!w->dapm->component) + if (!dapm->component) return -EIO; - return snd_soc_component_update_bits_async(w->dapm->component, reg, + return snd_soc_component_update_bits_async(dapm->component, reg, mask, value); } +static int soc_dapm_test_bits(struct snd_soc_dapm_context *dapm, + int reg, unsigned int mask, unsigned int value) +{ + if (!dapm->component) + return -EIO; + return snd_soc_component_test_bits(dapm->component, reg, mask, value); +} + static void soc_dapm_async_complete(struct snd_soc_dapm_context *dapm) { if (dapm->component) @@ -420,15 +450,10 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, if (ret != 0) goto out; - if (dapm->codec) { - if (dapm->codec->driver->set_bias_level) - ret = dapm->codec->driver->set_bias_level(dapm->codec, - level); - else - dapm->bias_level = level; - } else if (!card || dapm != &card->dapm) { + if (dapm->set_bias_level) + ret = dapm->set_bias_level(dapm, level); + else if (!card || dapm != &card->dapm) dapm->bias_level = level; - } if (ret != 0) goto out; @@ -452,7 +477,7 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, int i; if (e->reg != SND_SOC_NOPM) { - soc_widget_read(dest, e->reg, &val); + soc_dapm_read(dapm, e->reg, &val); val = (val >> e->shift_l) & e->mask; item = snd_soc_enum_val_to_item(e, val); } else { @@ -496,7 +521,7 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_widget *w, unsigned int val; if (reg != SND_SOC_NOPM) { - soc_widget_read(w, reg, &val); + soc_dapm_read(w->dapm, reg, &val); val = (val >> shift) & mask; if (invert) val = max - val; @@ -570,11 +595,7 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, const char *name; int ret; - if (dapm->codec) - prefix = dapm->codec->name_prefix; - else - prefix = NULL; - + prefix = soc_dapm_prefix(dapm); if (prefix) prefix_len = strlen(prefix) + 1; else @@ -1308,16 +1329,18 @@ static void dapm_seq_check_event(struct snd_soc_card *card, static void dapm_seq_run_coalesced(struct snd_soc_card *card, struct list_head *pending) { + struct snd_soc_dapm_context *dapm; struct snd_soc_dapm_widget *w; int reg; unsigned int value = 0; unsigned int mask = 0; - reg = list_first_entry(pending, struct snd_soc_dapm_widget, - power_list)->reg; + w = list_first_entry(pending, struct snd_soc_dapm_widget, power_list); + reg = w->reg; + dapm = w->dapm; list_for_each_entry(w, pending, power_list) { - WARN_ON(reg != w->reg); + WARN_ON(reg != w->reg || dapm != w->dapm); w->power = w->new_power; mask |= w->mask << w->shift; @@ -1326,7 +1349,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, else value |= w->off_val << w->shift; - pop_dbg(w->dapm->dev, card->pop_time, + pop_dbg(dapm->dev, card->pop_time, "pop test : Queue %s: reg=0x%x, 0x%x/0x%x\n", w->name, reg, value, mask); @@ -1339,14 +1362,12 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, /* Any widget will do, they should all be updating the * same register. */ - w = list_first_entry(pending, struct snd_soc_dapm_widget, - power_list); - pop_dbg(w->dapm->dev, card->pop_time, + pop_dbg(dapm->dev, card->pop_time, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); pop_wait(card->pop_time); - soc_widget_update_bits(w, reg, mask, value); + soc_dapm_update_bits(dapm, reg, mask, value); } list_for_each_entry(w, pending, power_list) { @@ -1492,7 +1513,8 @@ static void dapm_widget_update(struct snd_soc_card *card) if (!w) return; - ret = soc_widget_update_bits(w, update->reg, update->mask, update->val); + ret = soc_dapm_update_bits(w->dapm, update->reg, update->mask, + update->val); if (ret < 0) dev_err(w->dapm->dev, "ASoC: %s DAPM update failed: %d\n", w->name, ret); @@ -2062,17 +2084,13 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, } EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); -/* show dapm widget status in sys fs */ -static ssize_t dapm_widget_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t dapm_widget_show_codec(struct snd_soc_codec *codec, char *buf) { - struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); - struct snd_soc_codec *codec =rtd->codec; struct snd_soc_dapm_widget *w; int count = 0; char *state = "not set"; - list_for_each_entry(w, &codec->card->widgets, list) { + list_for_each_entry(w, &codec->component.card->widgets, list) { if (w->dapm != &codec->dapm) continue; @@ -2120,6 +2138,21 @@ static ssize_t dapm_widget_show(struct device *dev, return count; } +/* show dapm widget status in sys fs */ +static ssize_t dapm_widget_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int i, count = 0; + + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_codec *codec = rtd->codec_dais[i]->codec; + count += dapm_widget_show_codec(codec, buf + count); + } + + return count; +} + static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); int snd_soc_dapm_sys_add(struct device *dev) @@ -2371,14 +2404,16 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, const char *source; char prefixed_sink[80]; char prefixed_source[80]; + const char *prefix; int ret; - if (dapm->codec && dapm->codec->name_prefix) { + prefix = soc_dapm_prefix(dapm); + if (prefix) { snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", - dapm->codec->name_prefix, route->sink); + prefix, route->sink); sink = prefixed_sink; snprintf(prefixed_source, sizeof(prefixed_source), "%s %s", - dapm->codec->name_prefix, route->source); + prefix, route->source); source = prefixed_source; } else { sink = route->sink; @@ -2439,6 +2474,7 @@ static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm, const char *source; char prefixed_sink[80]; char prefixed_source[80]; + const char *prefix; if (route->control) { dev_err(dapm->dev, @@ -2446,12 +2482,13 @@ static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm, return -EINVAL; } - if (dapm->codec && dapm->codec->name_prefix) { + prefix = soc_dapm_prefix(dapm); + if (prefix) { snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", - dapm->codec->name_prefix, route->sink); + prefix, route->sink); sink = prefixed_sink; snprintf(prefixed_source, sizeof(prefixed_source), "%s %s", - dapm->codec->name_prefix, route->source); + prefix, route->source); source = prefixed_source; } else { sink = route->sink; @@ -2670,7 +2707,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) /* Read the initial power state from the device */ if (w->reg >= 0) { - soc_widget_read(w, w->reg, &val); + soc_dapm_read(w->dapm, w->reg, &val); val = val >> w->shift; val &= w->mask; if (val == w->on_val) @@ -2701,8 +2738,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct snd_soc_card *card = codec->card; + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); + struct snd_soc_card *card = dapm->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; int reg = mc->reg; @@ -2711,17 +2748,20 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val; + int ret = 0; if (snd_soc_volsw_is_stereo(mc)) - dev_warn(codec->dapm.dev, + dev_warn(dapm->dev, "ASoC: Control '%s' is stereo, which is not supported\n", kcontrol->id.name); mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (dapm_kcontrol_is_powered(kcontrol) && reg != SND_SOC_NOPM) - val = (snd_soc_read(codec, reg) >> shift) & mask; - else + if (dapm_kcontrol_is_powered(kcontrol) && reg != SND_SOC_NOPM) { + ret = soc_dapm_read(dapm, reg, &val); + val = (val >> shift) & mask; + } else { val = dapm_kcontrol_get_value(kcontrol); + } mutex_unlock(&card->dapm_mutex); if (invert) @@ -2729,7 +2769,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, else ucontrol->value.integer.value[0] = val; - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw); @@ -2745,8 +2785,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw); int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct snd_soc_card *card = codec->card; + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); + struct snd_soc_card *card = dapm->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; int reg = mc->reg; @@ -2760,7 +2800,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int ret = 0; if (snd_soc_volsw_is_stereo(mc)) - dev_warn(codec->dapm.dev, + dev_warn(dapm->dev, "ASoC: Control '%s' is stereo, which is not supported\n", kcontrol->id.name); @@ -2778,7 +2818,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mask = mask << shift; val = val << shift; - reg_change = snd_soc_test_bits(codec, reg, mask, val); + reg_change = soc_dapm_test_bits(dapm, reg, mask, val); } if (change || reg_change) { @@ -2817,12 +2857,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val; + int ret = 0; if (e->reg != SND_SOC_NOPM) - reg_val = snd_soc_read(codec, e->reg); + ret = soc_dapm_read(dapm, e->reg, ®_val); else reg_val = dapm_kcontrol_get_value(kcontrol); @@ -2834,7 +2875,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, ucontrol->value.enumerated.item[1] = val; } - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); @@ -2850,8 +2891,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct snd_soc_card *card = codec->card; + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); + struct snd_soc_card *card = dapm->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int *item = ucontrol->value.enumerated.item; unsigned int val, change; @@ -2874,7 +2915,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); if (e->reg != SND_SOC_NOPM) - change = snd_soc_test_bits(codec, e->reg, mask, val); + change = soc_dapm_test_bits(dapm, e->reg, mask, val); else change = dapm_kcontrol_set_value(kcontrol, val); @@ -2971,6 +3012,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) { struct snd_soc_dapm_widget *w; + const char *prefix; int ret; if ((w = dapm_cnew_widget(widget)) == NULL) @@ -3011,9 +3053,9 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; } - if (dapm->codec && dapm->codec->name_prefix) - w->name = kasprintf(GFP_KERNEL, "%s %s", - dapm->codec->name_prefix, widget->name); + prefix = soc_dapm_prefix(dapm); + if (prefix) + w->name = kasprintf(GFP_KERNEL, "%s %s", prefix, widget->name); else w->name = kasprintf(GFP_KERNEL, "%s", widget->name); @@ -3066,7 +3108,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, w->dapm = dapm; w->codec = dapm->codec; - w->platform = dapm->platform; INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); @@ -3173,27 +3214,15 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - if (source->driver->ops && source->driver->ops->hw_params) { - substream.stream = SNDRV_PCM_STREAM_CAPTURE; - ret = source->driver->ops->hw_params(&substream, - params, source); - if (ret != 0) { - dev_err(source->dev, - "ASoC: hw_params() failed: %d\n", ret); - goto out; - } - } + substream.stream = SNDRV_PCM_STREAM_CAPTURE; + ret = soc_dai_hw_params(&substream, params, source); + if (ret < 0) + goto out; - if (sink->driver->ops && sink->driver->ops->hw_params) { - substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - ret = sink->driver->ops->hw_params(&substream, params, - sink); - if (ret != 0) { - dev_err(sink->dev, - "ASoC: hw_params() failed: %d\n", ret); - goto out; - } - } + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + ret = soc_dai_hw_params(&substream, params, sink); + if (ret < 0) + goto out; break; case SND_SOC_DAPM_POST_PMU: @@ -3365,25 +3394,15 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } -void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) +static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = card->rtd; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dapm_widget *sink, *source; - struct snd_soc_dai *cpu_dai, *codec_dai; int i; - /* for each BE DAI link... */ - for (i = 0; i < card->num_rtd; i++) { - rtd = &card->rtd[i]; - cpu_dai = rtd->cpu_dai; - codec_dai = rtd->codec_dai; - - /* - * dynamic FE links have no fixed DAI mapping. - * CODEC<->CODEC links have no direct connection. - */ - if (rtd->dai_link->dynamic || rtd->dai_link->params) - continue; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; /* there is no point in connecting BE DAI links with dummies */ if (snd_soc_dai_is_dummy(codec_dai) || @@ -3395,8 +3414,8 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) source = cpu_dai->playback_widget; sink = codec_dai->playback_widget; dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - cpu_dai->codec->name, source->name, - codec_dai->platform->name, sink->name); + cpu_dai->component->name, source->name, + codec_dai->component->name, sink->name); snd_soc_dapm_add_path(&card->dapm, source, sink, NULL, NULL); @@ -3407,8 +3426,8 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) source = codec_dai->capture_widget; sink = cpu_dai->capture_widget; dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - codec_dai->codec->name, source->name, - cpu_dai->platform->name, sink->name); + codec_dai->component->name, source->name, + cpu_dai->component->name, sink->name); snd_soc_dapm_add_path(&card->dapm, source, sink, NULL, NULL); @@ -3445,11 +3464,34 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, } } +void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd = card->rtd; + int i; + + /* for each BE DAI link... */ + for (i = 0; i < card->num_rtd; i++) { + rtd = &card->rtd[i]; + + /* + * dynamic FE links have no fixed DAI mapping. + * CODEC<->CODEC links have no direct connection. + */ + if (rtd->dai_link->dynamic || rtd->dai_link->params) + continue; + + dapm_connect_dai_link_widgets(card, rtd); + } +} + static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int event) { + int i; + soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event); - soc_dapm_dai_stream_event(rtd->codec_dai, stream, event); + for (i = 0; i < rtd->num_codecs; i++) + soc_dapm_dai_stream_event(rtd->codec_dais[i], stream, event); dapm_power_widgets(rtd->card, event); } @@ -3758,36 +3800,31 @@ static bool snd_soc_dapm_widget_in_card_paths(struct snd_soc_card *card, } /** - * snd_soc_dapm_auto_nc_codec_pins - call snd_soc_dapm_nc_pin for unused pins - * @codec: The codec whose pins should be processed + * snd_soc_dapm_auto_nc_pins - call snd_soc_dapm_nc_pin for unused pins + * @card: The card whose pins should be processed * - * Automatically call snd_soc_dapm_nc_pin() for any external pins in the codec - * which are unused. Pins are used if they are connected externally to the - * codec, whether that be to some other device, or a loop-back connection to - * the codec itself. + * Automatically call snd_soc_dapm_nc_pin() for any external pins in the card + * which are unused. Pins are used if they are connected externally to a + * component, whether that be to some other device, or a loop-back connection to + * the component itself. */ -void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec) +void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card) { - struct snd_soc_card *card = codec->card; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_dapm_widget *w; - dev_dbg(codec->dev, "ASoC: Auto NC: DAPMs: card:%p codec:%p\n", - &card->dapm, &codec->dapm); + dev_dbg(card->dev, "ASoC: Auto NC: DAPMs: card:%p\n", &card->dapm); list_for_each_entry(w, &card->widgets, list) { - if (w->dapm != dapm) - continue; switch (w->id) { case snd_soc_dapm_input: case snd_soc_dapm_output: case snd_soc_dapm_micbias: - dev_dbg(codec->dev, "ASoC: Auto NC: Checking widget %s\n", + dev_dbg(card->dev, "ASoC: Auto NC: Checking widget %s\n", w->name); if (!snd_soc_dapm_widget_in_card_paths(card, w)) { - dev_dbg(codec->dev, + dev_dbg(card->dev, "... Not in map; disabling\n"); - snd_soc_dapm_nc_pin(dapm, w->name); + snd_soc_dapm_nc_pin(w->dapm, w->name); } break; default: diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 5bace124ef43..6307f85e871b 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -119,7 +119,10 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea struct snd_dmaengine_dai_dma_data *dma_data; struct dma_slave_caps dma_caps; struct snd_pcm_hardware hw; - int ret; + u32 addr_widths = BIT(DMA_SLAVE_BUSWIDTH_1_BYTE) | + BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | + BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + int i, ret; if (pcm->config && pcm->config->pcm_hardware) return snd_soc_set_runtime_hwparams(substream, @@ -146,6 +149,38 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT) hw.info |= SNDRV_PCM_INFO_BATCH; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + addr_widths = dma_caps.dstn_addr_widths; + else + addr_widths = dma_caps.src_addr_widths; + } + + /* + * Prepare formats mask for valid/allowed sample types. If the dma does + * not have support for the given physical word size, it needs to be + * masked out so user space can not use the format which produces + * corrupted audio. + * In case the dma driver does not implement the slave_caps the default + * assumption is that it supports 1, 2 and 4 bytes widths. + */ + for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { + int bits = snd_pcm_format_physical_width(i); + + /* Enable only samples with DMA supported physical widths */ + switch (bits) { + case 8: + case 16: + case 24: + case 32: + case 64: + if (addr_widths & (1 << (bits / 8))) + hw.formats |= (1LL << i); + break; + default: + /* Unsupported types */ + break; + } } return snd_soc_set_runtime_hwparams(substream, &hw); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index d0d98810af91..ab47fea997a3 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -43,7 +43,7 @@ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, INIT_LIST_HEAD(&jack->jack_zones); BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); - return snd_jack_new(codec->card->snd_card, id, type, &jack->jack); + return snd_jack_new(codec->component.card->snd_card, id, type, &jack->jack); } EXPORT_SYMBOL_GPL(snd_soc_jack_new); @@ -260,7 +260,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) static irqreturn_t gpio_handler(int irq, void *data) { struct snd_soc_jack_gpio *gpio = data; - struct device *dev = gpio->jack->codec->card->dev; + struct device *dev = gpio->jack->codec->component.card->dev; trace_snd_soc_jack_irq(gpio->name); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 54d18f22a33e..731fdb5b5f9b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -7,7 +7,7 @@ * Copyright (C) 2010 Texas Instruments Inc. * * Authors: Liam Girdwood <lrg@ti.com> - * Mark Brown <broonie@opensource.wolfsonmicro.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 @@ -47,22 +47,26 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + int i; lockdep_assert_held(&rtd->pcm_mutex); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active++; - codec_dai->playback_active++; + for (i = 0; i < rtd->num_codecs; i++) + rtd->codec_dais[i]->playback_active++; } else { cpu_dai->capture_active++; - codec_dai->capture_active++; + for (i = 0; i < rtd->num_codecs; i++) + rtd->codec_dais[i]->capture_active++; } cpu_dai->active++; - codec_dai->active++; cpu_dai->component->active++; - codec_dai->component->active++; + for (i = 0; i < rtd->num_codecs; i++) { + rtd->codec_dais[i]->active++; + rtd->codec_dais[i]->component->active++; + } } /** @@ -78,22 +82,26 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + int i; lockdep_assert_held(&rtd->pcm_mutex); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active--; - codec_dai->playback_active--; + for (i = 0; i < rtd->num_codecs; i++) + rtd->codec_dais[i]->playback_active--; } else { cpu_dai->capture_active--; - codec_dai->capture_active--; + for (i = 0; i < rtd->num_codecs; i++) + rtd->codec_dais[i]->capture_active--; } cpu_dai->active--; - codec_dai->active--; cpu_dai->component->active--; - codec_dai->component->active--; + for (i = 0; i < rtd->num_codecs; i++) { + rtd->codec_dais[i]->component->active--; + rtd->codec_dais[i]->active--; + } } /** @@ -107,11 +115,16 @@ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) */ bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd) { + int i; + bool ignore = true; + if (!rtd->pmdown_time || rtd->dai_link->ignore_pmdown_time) return true; - return rtd->cpu_dai->component->ignore_pmdown_time && - rtd->codec_dai->component->ignore_pmdown_time; + for (i = 0; i < rtd->num_codecs; i++) + ignore &= rtd->codec_dais[i]->component->ignore_pmdown_time; + + return rtd->cpu_dai->component->ignore_pmdown_time && ignore; } /** @@ -222,8 +235,7 @@ static int soc_pcm_params_symmetry(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 *codec_dai = rtd->codec_dai; - unsigned int rate, channels, sample_bits, symmetry; + unsigned int rate, channels, sample_bits, symmetry, i; rate = params_rate(params); channels = params_channels(params); @@ -231,8 +243,11 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, /* reject unmatched parameters when applying symmetry */ symmetry = cpu_dai->driver->symmetric_rates || - codec_dai->driver->symmetric_rates || rtd->dai_link->symmetric_rates; + + for (i = 0; i < rtd->num_codecs; i++) + symmetry |= rtd->codec_dais[i]->driver->symmetric_rates; + if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) { dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n", cpu_dai->rate, rate); @@ -240,8 +255,11 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, } symmetry = cpu_dai->driver->symmetric_channels || - codec_dai->driver->symmetric_channels || rtd->dai_link->symmetric_channels; + + for (i = 0; i < rtd->num_codecs; i++) + symmetry |= rtd->codec_dais[i]->driver->symmetric_channels; + if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) { dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n", cpu_dai->channels, channels); @@ -249,8 +267,11 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, } symmetry = cpu_dai->driver->symmetric_samplebits || - codec_dai->driver->symmetric_samplebits || rtd->dai_link->symmetric_samplebits; + + for (i = 0; i < rtd->num_codecs; i++) + symmetry |= rtd->codec_dais[i]->driver->symmetric_samplebits; + if (symmetry && cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) { dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n", cpu_dai->sample_bits, sample_bits); @@ -264,15 +285,20 @@ static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver; - struct snd_soc_dai_driver *codec_driver = rtd->codec_dai->driver; struct snd_soc_dai_link *link = rtd->dai_link; + unsigned int symmetry, i; - return cpu_driver->symmetric_rates || codec_driver->symmetric_rates || - link->symmetric_rates || cpu_driver->symmetric_channels || - codec_driver->symmetric_channels || link->symmetric_channels || - cpu_driver->symmetric_samplebits || - codec_driver->symmetric_samplebits || - link->symmetric_samplebits; + symmetry = cpu_driver->symmetric_rates || link->symmetric_rates || + cpu_driver->symmetric_channels || link->symmetric_channels || + cpu_driver->symmetric_samplebits || link->symmetric_samplebits; + + for (i = 0; i < rtd->num_codecs; i++) + symmetry = symmetry || + rtd->codec_dais[i]->driver->symmetric_rates || + rtd->codec_dais[i]->driver->symmetric_channels || + rtd->codec_dais[i]->driver->symmetric_samplebits; + + return symmetry; } /* @@ -284,15 +310,10 @@ static int sample_sizes[] = { 24, 32, }; -static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static void soc_pcm_set_msb(struct snd_pcm_substream *substream, int bits) { - int ret, i, bits; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - bits = dai->driver->playback.sig_bits; - else - bits = dai->driver->capture.sig_bits; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret, i; if (!bits) return; @@ -304,38 +325,105 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, ret = snd_pcm_hw_constraint_msbits(substream->runtime, 0, sample_sizes[i], bits); if (ret != 0) - dev_warn(dai->dev, + dev_warn(rtd->dev, "ASoC: Failed to set MSB %d/%d: %d\n", bits, sample_sizes[i], ret); } } -static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, - struct snd_soc_pcm_stream *codec_stream, - struct snd_soc_pcm_stream *cpu_stream) +static void soc_pcm_apply_msb(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 *codec_dai; + int i; + unsigned int bits = 0, cpu_bits; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->playback.sig_bits == 0) { + bits = 0; + break; + } + bits = max(codec_dai->driver->playback.sig_bits, bits); + } + cpu_bits = cpu_dai->driver->playback.sig_bits; + } else { + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->playback.sig_bits == 0) { + bits = 0; + break; + } + bits = max(codec_dai->driver->capture.sig_bits, bits); + } + cpu_bits = cpu_dai->driver->capture.sig_bits; + } + + soc_pcm_set_msb(substream, bits); + soc_pcm_set_msb(substream, cpu_bits); +} + +static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hardware *hw = &runtime->hw; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_driver *cpu_dai_drv = rtd->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; + unsigned int chan_min = 0, chan_max = UINT_MAX; + unsigned int rate_min = 0, rate_max = UINT_MAX; + unsigned int rates = UINT_MAX; + u64 formats = ULLONG_MAX; + int i; - hw->channels_min = max(codec_stream->channels_min, - cpu_stream->channels_min); - hw->channels_max = min(codec_stream->channels_max, - cpu_stream->channels_max); - if (hw->formats) - hw->formats &= codec_stream->formats & cpu_stream->formats; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_stream = &cpu_dai_drv->playback; else - hw->formats = codec_stream->formats & cpu_stream->formats; - hw->rates = snd_pcm_rate_mask_intersect(codec_stream->rates, - cpu_stream->rates); + cpu_stream = &cpu_dai_drv->capture; - hw->rate_min = 0; - hw->rate_max = UINT_MAX; + /* first calculate min/max only for CODECs in the DAI link */ + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai_drv = rtd->codec_dais[i]->driver; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &codec_dai_drv->playback; + else + codec_stream = &codec_dai_drv->capture; + chan_min = max(chan_min, codec_stream->channels_min); + chan_max = min(chan_max, codec_stream->channels_max); + rate_min = max(rate_min, codec_stream->rate_min); + rate_max = min_not_zero(rate_max, codec_stream->rate_max); + formats &= codec_stream->formats; + rates = snd_pcm_rate_mask_intersect(codec_stream->rates, rates); + } + + /* + * chan min/max cannot be enforced if there are multiple CODEC DAIs + * connected to a single CPU DAI, use CPU DAI's directly and let + * channel allocation be fixed up later + */ + if (rtd->num_codecs > 1) { + chan_min = cpu_stream->channels_min; + chan_max = cpu_stream->channels_max; + } + + hw->channels_min = max(chan_min, cpu_stream->channels_min); + hw->channels_max = min(chan_max, cpu_stream->channels_max); + if (hw->formats) + hw->formats &= formats & cpu_stream->formats; + else + hw->formats = formats & cpu_stream->formats; + hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_stream->rates); snd_pcm_limit_hw_rates(runtime); hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); - hw->rate_min = max(hw->rate_min, codec_stream->rate_min); + hw->rate_min = max(hw->rate_min, rate_min); hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max); - hw->rate_max = min_not_zero(hw->rate_max, codec_stream->rate_max); + hw->rate_max = min_not_zero(hw->rate_max, rate_max); } /* @@ -349,15 +437,16 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; - int ret = 0; + struct snd_soc_dai *codec_dai; + const char *codec_dai_name = "multicodec"; + int i, ret = 0; pinctrl_pm_select_default_state(cpu_dai->dev); - pinctrl_pm_select_default_state(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pinctrl_pm_select_default_state(rtd->codec_dais[i]->dev); pm_runtime_get_sync(cpu_dai->dev); - pm_runtime_get_sync(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pm_runtime_get_sync(rtd->codec_dais[i]->dev); pm_runtime_get_sync(platform->dev); mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -376,18 +465,28 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) ret = platform->driver->ops->open(substream); if (ret < 0) { dev_err(platform->dev, "ASoC: can't open platform" - " %s: %d\n", platform->name, ret); + " %s: %d\n", platform->component.name, ret); goto platform_err; } } - if (codec_dai->driver->ops && codec_dai->driver->ops->startup) { - ret = codec_dai->driver->ops->startup(substream, codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, "ASoC: can't open codec" - " %s: %d\n", codec_dai->name, ret); - goto codec_dai_err; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops && codec_dai->driver->ops->startup) { + ret = codec_dai->driver->ops->startup(substream, + codec_dai); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: can't open codec %s: %d\n", + codec_dai->name, ret); + goto codec_dai_err; + } } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_dai->tx_mask = 0; + else + codec_dai->rx_mask = 0; } if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { @@ -404,13 +503,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto dynamic; /* Check that the codec and cpu DAIs are compatible */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->playback, - &cpu_dai_drv->playback); - } else { - soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->capture, - &cpu_dai_drv->capture); - } + soc_pcm_init_runtime_hw(substream); + + if (rtd->num_codecs == 1) + codec_dai_name = rtd->codec_dai->name; if (soc_pcm_has_symmetry(substream)) runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; @@ -418,23 +514,22 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) ret = -EINVAL; if (!runtime->hw.rates) { printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); goto config_err; } if (!runtime->hw.formats) { printk(KERN_ERR "ASoC: %s <-> %s No matching formats\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); goto config_err; } if (!runtime->hw.channels_min || !runtime->hw.channels_max || runtime->hw.channels_min > runtime->hw.channels_max) { printk(KERN_ERR "ASoC: %s <-> %s No matching channels\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); goto config_err; } - soc_pcm_apply_msb(substream, codec_dai); - soc_pcm_apply_msb(substream, cpu_dai); + soc_pcm_apply_msb(substream); /* Symmetry only applies if we've already got an active stream. */ if (cpu_dai->active) { @@ -443,14 +538,17 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto config_err; } - if (codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream, codec_dai); - if (ret != 0) - goto config_err; + for (i = 0; i < rtd->num_codecs; i++) { + if (rtd->codec_dais[i]->active) { + ret = soc_pcm_apply_symmetry(substream, + rtd->codec_dais[i]); + if (ret != 0) + goto config_err; + } } pr_debug("ASoC: %s <-> %s info:\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); pr_debug("ASoC: rate mask 0x%x\n", runtime->hw.rates); pr_debug("ASoC: min ch %d max ch %d\n", runtime->hw.channels_min, runtime->hw.channels_max); @@ -469,10 +567,15 @@ config_err: rtd->dai_link->ops->shutdown(substream); machine_err: - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); + i = rtd->num_codecs; codec_dai_err: + while (--i >= 0) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + } + if (platform->driver->ops && platform->driver->ops->close) platform->driver->ops->close(substream); @@ -483,10 +586,13 @@ out: mutex_unlock(&rtd->pcm_mutex); pm_runtime_put(platform->dev); - pm_runtime_put(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pm_runtime_put(rtd->codec_dais[i]->dev); pm_runtime_put(cpu_dai->dev); - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) { + if (!rtd->codec_dais[i]->active) + pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev); + } if (!cpu_dai->active) pinctrl_pm_select_sleep_state(cpu_dai->dev); @@ -502,7 +608,7 @@ static void close_delayed_work(struct work_struct *work) { struct snd_soc_pcm_runtime *rtd = container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dais[0]; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -531,7 +637,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai; + int i; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -541,14 +648,20 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) if (!cpu_dai->active) cpu_dai->rate = 0; - if (!codec_dai->active) - codec_dai->rate = 0; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (!codec_dai->active) + codec_dai->rate = 0; + } if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + } if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); @@ -578,10 +691,13 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) mutex_unlock(&rtd->pcm_mutex); pm_runtime_put(platform->dev); - pm_runtime_put(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pm_runtime_put(rtd->codec_dais[i]->dev); pm_runtime_put(cpu_dai->dev); - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) { + if (!rtd->codec_dais[i]->active) + pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev); + } if (!cpu_dai->active) pinctrl_pm_select_sleep_state(cpu_dai->dev); @@ -598,8 +714,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; + struct snd_soc_dai *codec_dai; + int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -621,12 +737,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - if (codec_dai->driver->ops && codec_dai->driver->ops->prepare) { - ret = codec_dai->driver->ops->prepare(substream, codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, "ASoC: DAI prepare error: %d\n", - ret); - goto out; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops && codec_dai->driver->ops->prepare) { + ret = codec_dai->driver->ops->prepare(substream, + codec_dai); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: DAI prepare error: %d\n", ret); + goto out; + } } } @@ -649,13 +769,44 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(rtd, substream->stream, SND_SOC_DAPM_STREAM_START); - snd_soc_dai_digital_mute(codec_dai, 0, substream->stream); + for (i = 0; i < rtd->num_codecs; i++) + snd_soc_dai_digital_mute(rtd->codec_dais[i], 0, + substream->stream); out: mutex_unlock(&rtd->pcm_mutex); return ret; } +static void soc_pcm_codec_params_fixup(struct snd_pcm_hw_params *params, + unsigned int mask) +{ + struct snd_interval *interval; + int channels = hweight_long(mask); + + interval = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + interval->min = channels; + interval->max = channels; +} + +int soc_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int ret; + + if (dai->driver->ops && dai->driver->ops->hw_params) { + ret = dai->driver->ops->hw_params(substream, params, dai); + if (ret < 0) { + dev_err(dai->dev, "ASoC: can't set %s hw params: %d\n", + dai->name, ret); + return ret; + } + } + + 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 @@ -667,8 +818,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; + int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -685,29 +835,40 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_params) { - ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, "ASoC: can't set %s hw params:" - " %d\n", codec_dai->name, ret); + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + struct snd_pcm_hw_params codec_params; + + /* copy params for each codec */ + codec_params = *params; + + /* fixup params based on TDM slot masks */ + if (codec_dai->tx_mask) + soc_pcm_codec_params_fixup(&codec_params, + codec_dai->tx_mask); + if (codec_dai->rx_mask) + soc_pcm_codec_params_fixup(&codec_params, + codec_dai->rx_mask); + + ret = soc_dai_hw_params(substream, &codec_params, codec_dai); + if(ret < 0) goto codec_err; - } - } - if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_params) { - ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); - if (ret < 0) { - dev_err(cpu_dai->dev, "ASoC: %s hw params failed: %d\n", - cpu_dai->name, ret); - goto interface_err; - } + codec_dai->rate = params_rate(&codec_params); + codec_dai->channels = params_channels(&codec_params); + codec_dai->sample_bits = snd_pcm_format_physical_width( + params_format(&codec_params)); } + ret = soc_dai_hw_params(substream, params, cpu_dai); + if (ret < 0) + goto interface_err; + if (platform->driver->ops && platform->driver->ops->hw_params) { ret = platform->driver->ops->hw_params(substream, params); if (ret < 0) { dev_err(platform->dev, "ASoC: %s hw params failed: %d\n", - platform->name, ret); + platform->component.name, ret); goto platform_err; } } @@ -718,11 +879,6 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, cpu_dai->sample_bits = snd_pcm_format_physical_width(params_format(params)); - codec_dai->rate = params_rate(params); - codec_dai->channels = params_channels(params); - codec_dai->sample_bits = - snd_pcm_format_physical_width(params_format(params)); - out: mutex_unlock(&rtd->pcm_mutex); return ret; @@ -732,10 +888,16 @@ platform_err: cpu_dai->driver->ops->hw_free(substream, cpu_dai); interface_err: - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); + i = rtd->num_codecs; codec_err: + while (--i >= 0) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + codec_dai->rate = 0; + } + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); @@ -751,8 +913,9 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai; bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int i; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -763,16 +926,22 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) cpu_dai->sample_bits = 0; } - if (codec_dai->active == 1) { - codec_dai->rate = 0; - codec_dai->channels = 0; - codec_dai->sample_bits = 0; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->active == 1) { + codec_dai->rate = 0; + codec_dai->channels = 0; + codec_dai->sample_bits = 0; + } } /* apply codec digital mute */ - if ((playback && codec_dai->playback_active == 1) || - (!playback && codec_dai->capture_active == 1)) - snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); + for (i = 0; i < rtd->num_codecs; i++) { + if ((playback && rtd->codec_dais[i]->playback_active == 1) || + (!playback && rtd->codec_dais[i]->capture_active == 1)) + snd_soc_dai_digital_mute(rtd->codec_dais[i], 1, + substream->stream); + } /* free any machine hw params */ if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) @@ -783,8 +952,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) platform->driver->ops->hw_free(substream); /* now free hw params for the DAIs */ - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + } if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); @@ -798,13 +970,17 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (codec_dai->driver->ops && codec_dai->driver->ops->trigger) { - ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; + struct snd_soc_dai *codec_dai; + int i, ret; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops && codec_dai->driver->ops->trigger) { + ret = codec_dai->driver->ops->trigger(substream, + cmd, codec_dai); + if (ret < 0) + return ret; + } } if (platform->driver->ops && platform->driver->ops->trigger) { @@ -834,14 +1010,18 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (codec_dai->driver->ops && - codec_dai->driver->ops->bespoke_trigger) { - ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; + struct snd_soc_dai *codec_dai; + int i, ret; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops && + codec_dai->driver->ops->bespoke_trigger) { + ret = codec_dai->driver->ops->bespoke_trigger(substream, + cmd, codec_dai); + if (ret < 0) + return ret; + } } if (platform->driver->bespoke_trigger) { @@ -867,10 +1047,12 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai; struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t offset = 0; snd_pcm_sframes_t delay = 0; + snd_pcm_sframes_t codec_delay = 0; + int i; if (platform->driver->ops && platform->driver->ops->pointer) offset = platform->driver->ops->pointer(substream); @@ -878,11 +1060,21 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops && cpu_dai->driver->ops->delay) delay += cpu_dai->driver->ops->delay(substream, cpu_dai); - if (codec_dai->driver->ops && codec_dai->driver->ops->delay) - delay += codec_dai->driver->ops->delay(substream, codec_dai); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops && codec_dai->driver->ops->delay) + codec_delay = max(codec_delay, + codec_dai->driver->ops->delay(substream, + codec_dai)); + } + delay += codec_delay; + /* + * None of the existing platform drivers implement delay(), so + * for now the codec_dai of first multicodec entry is used + */ if (platform->driver->delay) - delay += platform->driver->delay(substream, codec_dai); + delay += platform->driver->delay(substream, rtd->codec_dais[0]); runtime->delay = delay; @@ -985,7 +1177,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, struct snd_soc_dapm_widget *widget, int stream) { struct snd_soc_pcm_runtime *be; - int i; + int i, j; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { for (i = 0; i < card->num_links; i++) { @@ -994,9 +1186,14 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (!be->dai_link->no_pcm) continue; - if (be->cpu_dai->playback_widget == widget || - be->codec_dai->playback_widget == widget) + if (be->cpu_dai->playback_widget == widget) return be; + + for (j = 0; j < be->num_codecs; j++) { + struct snd_soc_dai *dai = be->codec_dais[j]; + if (dai->playback_widget == widget) + return be; + } } } else { @@ -1006,9 +1203,14 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (!be->dai_link->no_pcm) continue; - if (be->cpu_dai->capture_widget == widget || - be->codec_dai->capture_widget == widget) + if (be->cpu_dai->capture_widget == widget) return be; + + for (j = 0; j < be->num_codecs; j++) { + struct snd_soc_dai *dai = be->codec_dais[j]; + if (dai->capture_widget == widget) + return be; + } } } @@ -1071,6 +1273,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, /* Destroy any old FE <--> BE connections */ list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + unsigned int i; /* is there a valid CPU DAI widget for this BE */ widget = dai_get_widget(dpcm->be->cpu_dai, stream); @@ -1080,11 +1283,14 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* is there a valid CODEC DAI widget for this BE */ - widget = dai_get_widget(dpcm->be->codec_dai, stream); + for (i = 0; i < dpcm->be->num_codecs; i++) { + struct snd_soc_dai *dai = dpcm->be->codec_dais[i]; + widget = dai_get_widget(dai, stream); - /* prune the BE if it's no longer in our active list */ - if (widget && widget_in_list(list, widget)) - continue; + /* prune the BE if it's no longer in our active list */ + if (widget && widget_in_list(list, widget)) + continue; + } dev_dbg(fe->dev, "ASoC: pruning %s BE %s for %s\n", stream ? "capture" : "playback", @@ -2069,6 +2275,7 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) 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) @@ -2113,16 +2320,22 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) list_for_each_entry(dpcm, clients, list_be) { struct snd_soc_pcm_runtime *be = dpcm->be; - struct snd_soc_dai *dai = be->codec_dai; - struct snd_soc_dai_driver *drv = dai->driver; + int i; if (be->dai_link->ignore_suspend) continue; - dev_dbg(be->dev, "ASoC: BE digital mute %s\n", be->dai_link->name); + for (i = 0; i < be->num_codecs; i++) { + struct snd_soc_dai *dai = be->codec_dais[i]; + struct snd_soc_dai_driver *drv = dai->driver; + + dev_dbg(be->dev, "ASoC: BE digital mute %s\n", + be->dai_link->name); - if (drv->ops && drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, mute); + if (drv->ops && drv->ops->digital_mute && + dai->playback_active) + drv->ops->digital_mute(dai, mute); + } } return 0; @@ -2187,22 +2400,27 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + int i; if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { playback = rtd->dai_link->dpcm_playback; capture = rtd->dai_link->dpcm_capture; } else { - if (codec_dai->driver->playback.channels_min && - cpu_dai->driver->playback.channels_min) - playback = 1; - if (codec_dai->driver->capture.channels_min && - cpu_dai->driver->capture.channels_min) - capture = 1; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->playback.channels_min) + playback = 1; + if (codec_dai->driver->capture.channels_min) + capture = 1; + } + + capture = capture && cpu_dai->driver->capture.channels_min; + playback = playback && cpu_dai->driver->playback.channels_min; } if (rtd->dai_link->playback_only) { @@ -2228,7 +2446,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->dai_link->stream_name); else snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); + rtd->dai_link->stream_name, + (rtd->num_codecs > 1) ? + "multicodec" : rtd->codec_dai->name, num); ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback, capture, &pcm); @@ -2301,8 +2521,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) pcm->private_free = platform->driver->pcm_free; out: - dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", codec_dai->name, - cpu_dai->name); + dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", + (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, + cpu_dai->name); return ret; } diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 02734bd4f09b..a83aff09dce2 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -41,8 +41,7 @@ static int tegra_alc5632_asoc_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_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -105,7 +104,7 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card); + struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card); snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &tegra_alc5632_hs_jack); diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index ce73e1f62c4b..b86cd9936ef1 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -49,8 +49,7 @@ static int tegra_max98090_asoc_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_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -127,7 +126,7 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct tegra_max98090 *machine = snd_soc_card_get_drvdata(codec->card); + struct tegra_max98090 *machine = snd_soc_card_get_drvdata(rtd->card); if (gpio_is_valid(machine->gpio_hp_det)) { snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 4feb16a99e02..a6898831fb9f 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -51,8 +51,7 @@ static int tegra_rt5640_asoc_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_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -110,7 +109,7 @@ static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(codec->card); + struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(rtd->card); snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, &tegra_rt5640_hp_jack); diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index 8e774d1a243c..769e28f6642e 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -55,8 +55,7 @@ static int tegra_wm8753_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_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 0939661df60b..86e05e938585 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -60,8 +60,7 @@ static int tegra_wm8903_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_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -173,7 +172,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); if (gpio_is_valid(machine->gpio_hp_det)) { diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 734bfcd21148..589d2d9b553a 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -50,8 +50,7 @@ static int trimslice_asoc_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_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); int srate, mclk; int err; diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index be1b1aa96b7e..b2c3d0d5dca3 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2534,12 +2534,10 @@ static int snd_dbri_create(struct snd_card *card, dbri->op = op; dbri->irq = irq; - dbri->dma = dma_alloc_coherent(&op->dev, - sizeof(struct dbri_dma), - &dbri->dma_dvma, GFP_ATOMIC); + dbri->dma = dma_zalloc_coherent(&op->dev, sizeof(struct dbri_dma), + &dbri->dma_dvma, GFP_ATOMIC); if (!dbri->dma) return -ENOMEM; - memset((void *)dbri->dma, 0, sizeof(struct dbri_dma)); dprintk(D_GEN, "DMA Cmd Block 0x%p (0x%08x)\n", dbri->dma, dbri->dma_dvma); diff --git a/sound/usb/card.c b/sound/usb/card.c index a09e5f3519e3..7ecd0e8a5c51 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -680,6 +680,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) struct snd_usb_audio *chip = usb_get_intfdata(intf); struct snd_usb_stream *as; struct usb_mixer_interface *mixer; + struct list_head *p; if (chip == (void *)-1L) return 0; @@ -692,6 +693,9 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) as->substream[0].need_setup_ep = as->substream[1].need_setup_ep = true; } + list_for_each(p, &chip->midi_list) { + snd_usbmidi_suspend(p); + } } } else { /* @@ -713,6 +717,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) { struct snd_usb_audio *chip = usb_get_intfdata(intf); struct usb_mixer_interface *mixer; + struct list_head *p; int err = 0; if (chip == (void *)-1L) @@ -731,6 +736,10 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) goto err_out; } + list_for_each(p, &chip->midi_list) { + snd_usbmidi_resume(p); + } + if (!chip->autosuspended) snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); chip->autosuspended = 0; diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 9da74d2e8eee..7b166c2be0f7 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -102,8 +102,8 @@ struct usb_protocol_ops { void (*input)(struct snd_usb_midi_in_endpoint*, uint8_t*, int); void (*output)(struct snd_usb_midi_out_endpoint *ep, struct urb *urb); void (*output_packet)(struct urb*, uint8_t, uint8_t, uint8_t, uint8_t); - void (*init_out_endpoint)(struct snd_usb_midi_out_endpoint*); - void (*finish_out_endpoint)(struct snd_usb_midi_out_endpoint*); + void (*init_out_endpoint)(struct snd_usb_midi_out_endpoint *); + void (*finish_out_endpoint)(struct snd_usb_midi_out_endpoint *); }; struct snd_usb_midi { @@ -112,7 +112,7 @@ struct snd_usb_midi { struct usb_interface *iface; const struct snd_usb_audio_quirk *quirk; struct snd_rawmidi *rmidi; - struct usb_protocol_ops* usb_protocol_ops; + struct usb_protocol_ops *usb_protocol_ops; struct list_head list; struct timer_list error_timer; spinlock_t disc_lock; @@ -134,7 +134,7 @@ struct snd_usb_midi { }; struct snd_usb_midi_out_endpoint { - struct snd_usb_midi* umidi; + struct snd_usb_midi *umidi; struct out_urb_context { struct urb *urb; struct snd_usb_midi_out_endpoint *ep; @@ -147,7 +147,7 @@ struct snd_usb_midi_out_endpoint { spinlock_t buffer_lock; struct usbmidi_out_port { - struct snd_usb_midi_out_endpoint* ep; + struct snd_usb_midi_out_endpoint *ep; struct snd_rawmidi_substream *substream; int active; uint8_t cable; /* cable number << 4 */ @@ -167,8 +167,8 @@ struct snd_usb_midi_out_endpoint { }; struct snd_usb_midi_in_endpoint { - struct snd_usb_midi* umidi; - struct urb* urbs[INPUT_URBS]; + struct snd_usb_midi *umidi; + struct urb *urbs[INPUT_URBS]; struct usbmidi_in_port { struct snd_rawmidi_substream *substream; u8 running_status_length; @@ -178,7 +178,7 @@ struct snd_usb_midi_in_endpoint { int current_port; }; -static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep); +static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint *ep); static const uint8_t snd_usbmidi_cin_length[] = { 0, 0, 2, 3, 3, 1, 2, 3, 3, 3, 3, 3, 2, 2, 3, 1 @@ -187,7 +187,7 @@ static const uint8_t snd_usbmidi_cin_length[] = { /* * Submits the URB, with error handling. */ -static int snd_usbmidi_submit_urb(struct urb* urb, gfp_t flags) +static int snd_usbmidi_submit_urb(struct urb *urb, gfp_t flags) { int err = usb_submit_urb(urb, flags); if (err < 0 && err != -ENODEV) @@ -221,10 +221,10 @@ static int snd_usbmidi_urb_error(const struct urb *urb) /* * Receives a chunk of MIDI data. */ -static void snd_usbmidi_input_data(struct snd_usb_midi_in_endpoint* ep, int portidx, - uint8_t* data, int length) +static void snd_usbmidi_input_data(struct snd_usb_midi_in_endpoint *ep, + int portidx, uint8_t *data, int length) { - struct usbmidi_in_port* port = &ep->ports[portidx]; + struct usbmidi_in_port *port = &ep->ports[portidx]; if (!port->substream) { dev_dbg(&ep->umidi->dev->dev, "unexpected port %d!\n", portidx); @@ -250,9 +250,9 @@ static void dump_urb(const char *type, const u8 *data, int length) /* * Processes the data read from the device. */ -static void snd_usbmidi_in_urb_complete(struct urb* urb) +static void snd_usbmidi_in_urb_complete(struct urb *urb) { - struct snd_usb_midi_in_endpoint* ep = urb->context; + struct snd_usb_midi_in_endpoint *ep = urb->context; if (urb->status == 0) { dump_urb("received", urb->transfer_buffer, urb->actual_length); @@ -274,10 +274,10 @@ static void snd_usbmidi_in_urb_complete(struct urb* urb) snd_usbmidi_submit_urb(urb, GFP_ATOMIC); } -static void snd_usbmidi_out_urb_complete(struct urb* urb) +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; + struct snd_usb_midi_out_endpoint *ep = context->ep; unsigned int urb_index; spin_lock(&ep->buffer_lock); @@ -304,10 +304,10 @@ static void snd_usbmidi_out_urb_complete(struct urb* urb) * This is called when some data should be transferred to the device * (from one or more substreams). */ -static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep) +static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint *ep) { unsigned int urb_index; - struct urb* urb; + struct urb *urb; unsigned long flags; spin_lock_irqsave(&ep->buffer_lock, flags); @@ -343,7 +343,8 @@ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep) static void snd_usbmidi_out_tasklet(unsigned long data) { - struct snd_usb_midi_out_endpoint* ep = (struct snd_usb_midi_out_endpoint *) data; + struct snd_usb_midi_out_endpoint *ep = + (struct snd_usb_midi_out_endpoint *) data; snd_usbmidi_do_output(ep); } @@ -375,7 +376,7 @@ static void snd_usbmidi_error_timer(unsigned long data) } /* helper function to send static data that may not DMA-able */ -static int send_bulk_static_data(struct snd_usb_midi_out_endpoint* ep, +static int send_bulk_static_data(struct snd_usb_midi_out_endpoint *ep, const void *data, int len) { int err = 0; @@ -396,8 +397,8 @@ static int send_bulk_static_data(struct snd_usb_midi_out_endpoint* ep, * fourth byte in each packet, and uses length instead of CIN. */ -static void snd_usbmidi_standard_input(struct snd_usb_midi_in_endpoint* ep, - uint8_t* buffer, int buffer_length) +static void snd_usbmidi_standard_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) { int i; @@ -405,12 +406,13 @@ static void snd_usbmidi_standard_input(struct snd_usb_midi_in_endpoint* ep, if (buffer[i] != 0) { int cable = buffer[i] >> 4; int length = snd_usbmidi_cin_length[buffer[i] & 0x0f]; - snd_usbmidi_input_data(ep, cable, &buffer[i + 1], length); + snd_usbmidi_input_data(ep, cable, &buffer[i + 1], + length); } } -static void snd_usbmidi_midiman_input(struct snd_usb_midi_in_endpoint* ep, - uint8_t* buffer, int buffer_length) +static void snd_usbmidi_midiman_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) { int i; @@ -427,8 +429,8 @@ static void snd_usbmidi_midiman_input(struct snd_usb_midi_in_endpoint* ep, * the data bytes but not the status byte and that is marked with CIN 4. */ static void snd_usbmidi_maudio_broken_running_status_input( - struct snd_usb_midi_in_endpoint* ep, - uint8_t* buffer, int buffer_length) + struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) { int i; @@ -458,7 +460,8 @@ static void snd_usbmidi_maudio_broken_running_status_input( * doesn't use this format.) */ port->running_status_length = 0; - snd_usbmidi_input_data(ep, cable, &buffer[i + 1], length); + snd_usbmidi_input_data(ep, cable, &buffer[i + 1], + length); } } @@ -479,11 +482,13 @@ static void snd_usbmidi_cme_input(struct snd_usb_midi_in_endpoint *ep, /* * Adds one USB MIDI packet to the output buffer. */ -static void snd_usbmidi_output_standard_packet(struct urb* urb, uint8_t p0, - uint8_t p1, uint8_t p2, uint8_t p3) +static void snd_usbmidi_output_standard_packet(struct urb *urb, uint8_t p0, + uint8_t p1, uint8_t p2, + uint8_t p3) { - uint8_t* buf = (uint8_t*)urb->transfer_buffer + urb->transfer_buffer_length; + uint8_t *buf = + (uint8_t *)urb->transfer_buffer + urb->transfer_buffer_length; buf[0] = p0; buf[1] = p1; buf[2] = p2; @@ -494,11 +499,13 @@ static void snd_usbmidi_output_standard_packet(struct urb* urb, uint8_t p0, /* * Adds one Midiman packet to the output buffer. */ -static void snd_usbmidi_output_midiman_packet(struct urb* urb, uint8_t p0, - uint8_t p1, uint8_t p2, uint8_t p3) +static void snd_usbmidi_output_midiman_packet(struct urb *urb, uint8_t p0, + uint8_t p1, uint8_t p2, + uint8_t p3) { - uint8_t* buf = (uint8_t*)urb->transfer_buffer + urb->transfer_buffer_length; + uint8_t *buf = + (uint8_t *)urb->transfer_buffer + urb->transfer_buffer_length; buf[0] = p1; buf[1] = p2; buf[2] = p3; @@ -509,8 +516,8 @@ static void snd_usbmidi_output_midiman_packet(struct urb* urb, uint8_t p0, /* * Converts MIDI commands to USB MIDI packets. */ -static void snd_usbmidi_transmit_byte(struct usbmidi_out_port* port, - uint8_t b, struct urb* urb) +static void snd_usbmidi_transmit_byte(struct usbmidi_out_port *port, + uint8_t b, struct urb *urb) { uint8_t p0 = port->cable; void (*output_packet)(struct urb*, uint8_t, uint8_t, uint8_t, uint8_t) = @@ -547,10 +554,12 @@ static void snd_usbmidi_transmit_byte(struct usbmidi_out_port* port, output_packet(urb, p0 | 0x05, 0xf7, 0, 0); break; case STATE_SYSEX_1: - output_packet(urb, p0 | 0x06, port->data[0], 0xf7, 0); + output_packet(urb, p0 | 0x06, port->data[0], + 0xf7, 0); break; case STATE_SYSEX_2: - output_packet(urb, p0 | 0x07, port->data[0], port->data[1], 0xf7); + output_packet(urb, p0 | 0x07, port->data[0], + port->data[1], 0xf7); break; } port->state = STATE_UNKNOWN; @@ -596,21 +605,22 @@ static void snd_usbmidi_transmit_byte(struct usbmidi_out_port* port, port->state = STATE_SYSEX_2; break; case STATE_SYSEX_2: - output_packet(urb, p0 | 0x04, port->data[0], port->data[1], b); + output_packet(urb, p0 | 0x04, port->data[0], + port->data[1], b); port->state = STATE_SYSEX_0; break; } } } -static void snd_usbmidi_standard_output(struct snd_usb_midi_out_endpoint* ep, +static void snd_usbmidi_standard_output(struct snd_usb_midi_out_endpoint *ep, struct urb *urb) { int p; /* FIXME: lower-numbered ports can starve higher-numbered ports */ for (p = 0; p < 0x10; ++p) { - struct usbmidi_out_port* port = &ep->ports[p]; + struct usbmidi_out_port *port = &ep->ports[p]; if (!port->active) continue; while (urb->transfer_buffer_length + 3 < ep->max_transfer) { @@ -753,18 +763,18 @@ static struct usb_protocol_ops snd_usbmidi_akai_ops = { * at the third byte. */ -static void snd_usbmidi_novation_input(struct snd_usb_midi_in_endpoint* ep, - uint8_t* buffer, int buffer_length) +static void snd_usbmidi_novation_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) { if (buffer_length < 2 || !buffer[0] || buffer_length < buffer[0] + 1) return; snd_usbmidi_input_data(ep, 0, &buffer[2], buffer[0] - 1); } -static void snd_usbmidi_novation_output(struct snd_usb_midi_out_endpoint* ep, +static void snd_usbmidi_novation_output(struct snd_usb_midi_out_endpoint *ep, struct urb *urb) { - uint8_t* transfer_buffer; + uint8_t *transfer_buffer; int count; if (!ep->ports[0].active) @@ -791,13 +801,13 @@ static struct usb_protocol_ops snd_usbmidi_novation_ops = { * "raw" protocol: just move raw MIDI bytes from/to the endpoint */ -static void snd_usbmidi_raw_input(struct snd_usb_midi_in_endpoint* ep, - uint8_t* buffer, int buffer_length) +static void snd_usbmidi_raw_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) { snd_usbmidi_input_data(ep, 0, buffer, buffer_length); } -static void snd_usbmidi_raw_output(struct snd_usb_midi_out_endpoint* ep, +static void snd_usbmidi_raw_output(struct snd_usb_midi_out_endpoint *ep, struct urb *urb) { int count; @@ -823,8 +833,8 @@ static struct usb_protocol_ops snd_usbmidi_raw_ops = { * FTDI protocol: raw MIDI bytes, but input packets have two modem status bytes. */ -static void snd_usbmidi_ftdi_input(struct snd_usb_midi_in_endpoint* ep, - uint8_t* buffer, int buffer_length) +static void snd_usbmidi_ftdi_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) { if (buffer_length > 2) snd_usbmidi_input_data(ep, 0, buffer + 2, buffer_length - 2); @@ -883,7 +893,7 @@ static struct usb_protocol_ops snd_usbmidi_122l_ops = { * Emagic USB MIDI protocol: raw MIDI with "F5 xx" port switching. */ -static void snd_usbmidi_emagic_init_out(struct snd_usb_midi_out_endpoint* ep) +static void snd_usbmidi_emagic_init_out(struct snd_usb_midi_out_endpoint *ep) { static const u8 init_data[] = { /* initialization magic: "get version" */ @@ -900,7 +910,7 @@ static void snd_usbmidi_emagic_init_out(struct snd_usb_midi_out_endpoint* ep) send_bulk_static_data(ep, init_data, sizeof(init_data)); } -static void snd_usbmidi_emagic_finish_out(struct snd_usb_midi_out_endpoint* ep) +static void snd_usbmidi_emagic_finish_out(struct snd_usb_midi_out_endpoint *ep) { static const u8 finish_data[] = { /* switch to patch mode with last preset */ @@ -916,8 +926,8 @@ static void snd_usbmidi_emagic_finish_out(struct snd_usb_midi_out_endpoint* ep) send_bulk_static_data(ep, finish_data, sizeof(finish_data)); } -static void snd_usbmidi_emagic_input(struct snd_usb_midi_in_endpoint* ep, - uint8_t* buffer, int buffer_length) +static void snd_usbmidi_emagic_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) { int i; @@ -960,18 +970,18 @@ static void snd_usbmidi_emagic_input(struct snd_usb_midi_in_endpoint* ep, } } -static void snd_usbmidi_emagic_output(struct snd_usb_midi_out_endpoint* ep, +static void snd_usbmidi_emagic_output(struct snd_usb_midi_out_endpoint *ep, struct urb *urb) { int port0 = ep->current_port; - uint8_t* buf = urb->transfer_buffer; + uint8_t *buf = urb->transfer_buffer; int buf_free = ep->max_transfer; int length, i; for (i = 0; i < 0x10; ++i) { /* round-robin, starting at the last current port */ int portnum = (port0 + i) & 15; - struct usbmidi_out_port* port = &ep->ports[portnum]; + struct usbmidi_out_port *port = &ep->ports[portnum]; if (!port->active) continue; @@ -1015,7 +1025,7 @@ static struct usb_protocol_ops snd_usbmidi_emagic_ops = { }; -static void update_roland_altsetting(struct snd_usb_midi* umidi) +static void update_roland_altsetting(struct snd_usb_midi *umidi) { struct usb_interface *intf; struct usb_host_interface *hostif; @@ -1037,7 +1047,7 @@ static void update_roland_altsetting(struct snd_usb_midi* umidi) static int substream_open(struct snd_rawmidi_substream *substream, int dir, int open) { - struct snd_usb_midi* umidi = substream->rmidi->private_data; + struct snd_usb_midi *umidi = substream->rmidi->private_data; struct snd_kcontrol *ctl; down_read(&umidi->disc_rwsem); @@ -1051,7 +1061,8 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir, if (!umidi->opened[0] && !umidi->opened[1]) { if (umidi->roland_load_ctl) { ctl = umidi->roland_load_ctl; - ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + ctl->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_INACTIVE; snd_ctl_notify(umidi->card, SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); update_roland_altsetting(umidi); @@ -1067,7 +1078,8 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir, if (!umidi->opened[0] && !umidi->opened[1]) { if (umidi->roland_load_ctl) { ctl = umidi->roland_load_ctl; - ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + ctl->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; snd_ctl_notify(umidi->card, SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); } @@ -1080,8 +1092,8 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir, static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) { - struct snd_usb_midi* umidi = substream->rmidi->private_data; - struct usbmidi_out_port* port = NULL; + struct snd_usb_midi *umidi = substream->rmidi->private_data; + struct usbmidi_out_port *port = NULL; int i, j; for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) @@ -1106,9 +1118,11 @@ static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream) return substream_open(substream, 0, 0); } -static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream, int up) +static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream, + int up) { - struct usbmidi_out_port* port = (struct usbmidi_out_port*)substream->runtime->private_data; + struct usbmidi_out_port *port = + (struct usbmidi_out_port *)substream->runtime->private_data; port->active = up; if (up) { @@ -1125,7 +1139,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream, static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream) { - struct usbmidi_out_port* port = substream->runtime->private_data; + struct usbmidi_out_port *port = substream->runtime->private_data; struct snd_usb_midi_out_endpoint *ep = port->ep; unsigned int drain_urbs; DEFINE_WAIT(wait); @@ -1164,9 +1178,10 @@ static int snd_usbmidi_input_close(struct snd_rawmidi_substream *substream) return substream_open(substream, 1, 0); } -static void snd_usbmidi_input_trigger(struct snd_rawmidi_substream *substream, int up) +static void snd_usbmidi_input_trigger(struct snd_rawmidi_substream *substream, + int up) { - struct snd_usb_midi* umidi = substream->rmidi->private_data; + struct snd_usb_midi *umidi = substream->rmidi->private_data; if (up) set_bit(substream->number, &umidi->input_triggered); @@ -1199,7 +1214,7 @@ static void free_urb_and_buffer(struct snd_usb_midi *umidi, struct urb *urb, * Frees an input endpoint. * May be called when ep hasn't been initialized completely. */ -static void snd_usbmidi_in_endpoint_delete(struct snd_usb_midi_in_endpoint* ep) +static void snd_usbmidi_in_endpoint_delete(struct snd_usb_midi_in_endpoint *ep) { unsigned int i; @@ -1213,12 +1228,12 @@ static void snd_usbmidi_in_endpoint_delete(struct snd_usb_midi_in_endpoint* ep) /* * Creates an input endpoint. */ -static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, - struct snd_usb_midi_endpoint_info* ep_info, - struct snd_usb_midi_endpoint* rep) +static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi *umidi, + struct snd_usb_midi_endpoint_info *ep_info, + struct snd_usb_midi_endpoint *rep) { - struct snd_usb_midi_in_endpoint* ep; - void* buffer; + struct snd_usb_midi_in_endpoint *ep; + void *buffer; unsigned int pipe; int length; unsigned int i; @@ -1289,14 +1304,14 @@ static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint *ep /* * Creates an output endpoint, and initializes output ports. */ -static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, - struct snd_usb_midi_endpoint_info* ep_info, - struct snd_usb_midi_endpoint* rep) +static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi, + struct snd_usb_midi_endpoint_info *ep_info, + struct snd_usb_midi_endpoint *rep) { - struct snd_usb_midi_out_endpoint* ep; + struct snd_usb_midi_out_endpoint *ep; unsigned int i; unsigned int pipe; - void* buffer; + void *buffer; rep->out = NULL; ep = kzalloc(sizeof(*ep), GFP_KERNEL); @@ -1381,12 +1396,12 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, /* * Frees everything. */ -static void snd_usbmidi_free(struct snd_usb_midi* umidi) +static void snd_usbmidi_free(struct snd_usb_midi *umidi) { int i; for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { - struct snd_usb_midi_endpoint* ep = &umidi->endpoints[i]; + struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i]; if (ep->out) snd_usbmidi_out_endpoint_delete(ep->out); if (ep->in) @@ -1399,9 +1414,9 @@ static void snd_usbmidi_free(struct snd_usb_midi* umidi) /* * Unlinks all URBs (must be done before the usb_device is deleted). */ -void snd_usbmidi_disconnect(struct list_head* p) +void snd_usbmidi_disconnect(struct list_head *p) { - struct snd_usb_midi* umidi; + struct snd_usb_midi *umidi; unsigned int i, j; umidi = list_entry(p, struct snd_usb_midi, list); @@ -1417,7 +1432,7 @@ void snd_usbmidi_disconnect(struct list_head* p) up_write(&umidi->disc_rwsem); for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { - struct snd_usb_midi_endpoint* ep = &umidi->endpoints[i]; + struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i]; if (ep->out) tasklet_kill(&ep->out->tasklet); if (ep->out) { @@ -1448,16 +1463,18 @@ EXPORT_SYMBOL(snd_usbmidi_disconnect); static void snd_usbmidi_rawmidi_free(struct snd_rawmidi *rmidi) { - struct snd_usb_midi* umidi = rmidi->private_data; + struct snd_usb_midi *umidi = rmidi->private_data; snd_usbmidi_free(umidi); } -static struct snd_rawmidi_substream *snd_usbmidi_find_substream(struct snd_usb_midi* umidi, - int stream, int number) +static struct snd_rawmidi_substream *snd_usbmidi_find_substream(struct snd_usb_midi *umidi, + int stream, + int number) { struct snd_rawmidi_substream *substream; - list_for_each_entry(substream, &umidi->rmidi->streams[stream].substreams, list) { + list_for_each_entry(substream, &umidi->rmidi->streams[stream].substreams, + list) { if (substream->number == number) return substream; } @@ -1633,7 +1650,7 @@ static struct port_info { SNDRV_SEQ_PORT_TYPE_SYNTHESIZER), }; -static struct port_info *find_port_info(struct snd_usb_midi* umidi, int number) +static struct port_info *find_port_info(struct snd_usb_midi *umidi, int number) { int i; @@ -1659,16 +1676,18 @@ static void snd_usbmidi_get_port_info(struct snd_rawmidi *rmidi, int number, } } -static void snd_usbmidi_init_substream(struct snd_usb_midi* umidi, +static void snd_usbmidi_init_substream(struct snd_usb_midi *umidi, int stream, int number, - struct snd_rawmidi_substream ** rsubstream) + struct snd_rawmidi_substream **rsubstream) { struct port_info *port_info; const char *name_format; - struct snd_rawmidi_substream *substream = snd_usbmidi_find_substream(umidi, stream, number); + struct snd_rawmidi_substream *substream = + snd_usbmidi_find_substream(umidi, stream, number); if (!substream) { - dev_err(&umidi->dev->dev, "substream %d:%d not found\n", stream, number); + dev_err(&umidi->dev->dev, "substream %d:%d not found\n", stream, + number); return; } @@ -1684,21 +1703,23 @@ static void snd_usbmidi_init_substream(struct snd_usb_midi* umidi, /* * Creates the endpoints and their ports. */ -static int snd_usbmidi_create_endpoints(struct snd_usb_midi* umidi, - struct snd_usb_midi_endpoint_info* endpoints) +static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi, + struct snd_usb_midi_endpoint_info *endpoints) { int i, j, err; int out_ports = 0, in_ports = 0; for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { if (endpoints[i].out_cables) { - err = snd_usbmidi_out_endpoint_create(umidi, &endpoints[i], + err = snd_usbmidi_out_endpoint_create(umidi, + &endpoints[i], &umidi->endpoints[i]); if (err < 0) return err; } if (endpoints[i].in_cables) { - err = snd_usbmidi_in_endpoint_create(umidi, &endpoints[i], + err = snd_usbmidi_in_endpoint_create(umidi, + &endpoints[i], &umidi->endpoints[i]); if (err < 0) return err; @@ -1706,12 +1727,16 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi* umidi, for (j = 0; j < 0x10; ++j) { if (endpoints[i].out_cables & (1 << j)) { - snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_OUTPUT, out_ports, + snd_usbmidi_init_substream(umidi, + SNDRV_RAWMIDI_STREAM_OUTPUT, + out_ports, &umidi->endpoints[i].out->ports[j].substream); ++out_ports; } if (endpoints[i].in_cables & (1 << j)) { - snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_INPUT, in_ports, + snd_usbmidi_init_substream(umidi, + SNDRV_RAWMIDI_STREAM_INPUT, + in_ports, &umidi->endpoints[i].in->ports[j].substream); ++in_ports; } @@ -1725,16 +1750,16 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi* umidi, /* * Returns MIDIStreaming device capabilities. */ -static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, - struct snd_usb_midi_endpoint_info* endpoints) +static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, + struct snd_usb_midi_endpoint_info *endpoints) { - struct usb_interface* intf; + struct usb_interface *intf; struct usb_host_interface *hostif; - struct usb_interface_descriptor* intfd; - struct usb_ms_header_descriptor* ms_header; + struct usb_interface_descriptor *intfd; + struct usb_ms_header_descriptor *ms_header; struct usb_host_endpoint *hostep; - struct usb_endpoint_descriptor* ep; - struct usb_ms_endpoint_descriptor* ms_ep; + struct usb_endpoint_descriptor *ep; + struct usb_ms_endpoint_descriptor *ms_ep; int i, epidx; intf = umidi->iface; @@ -1742,7 +1767,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, return -ENXIO; hostif = &intf->altsetting[0]; intfd = get_iface_desc(hostif); - ms_header = (struct usb_ms_header_descriptor*)hostif->extra; + ms_header = (struct usb_ms_header_descriptor *)hostif->extra; if (hostif->extralen >= 7 && ms_header->bLength >= 7 && ms_header->bDescriptorType == USB_DT_CS_INTERFACE && @@ -1759,7 +1784,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, ep = get_ep_desc(hostep); if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep)) continue; - ms_ep = (struct usb_ms_endpoint_descriptor*)hostep->extra; + ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra; if (hostep->extralen < 4 || ms_ep->bLength < 4 || ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || @@ -1783,9 +1808,10 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, * ESI MIDI Mate that try to use them anyway. */ endpoints[epidx].out_interval = 1; - endpoints[epidx].out_cables = (1 << ms_ep->bNumEmbMIDIJack) - 1; + endpoints[epidx].out_cables = + (1 << ms_ep->bNumEmbMIDIJack) - 1; dev_dbg(&umidi->dev->dev, "EP %02X: %d jack(s)\n", - ep->bEndpointAddress, ms_ep->bNumEmbMIDIJack); + ep->bEndpointAddress, ms_ep->bNumEmbMIDIJack); } else { if (endpoints[epidx].in_ep) { if (++epidx >= MIDI_MAX_ENDPOINTS) { @@ -1799,9 +1825,10 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, endpoints[epidx].in_interval = ep->bInterval; else if (snd_usb_get_speed(umidi->dev) == USB_SPEED_LOW) endpoints[epidx].in_interval = 1; - endpoints[epidx].in_cables = (1 << ms_ep->bNumEmbMIDIJack) - 1; + endpoints[epidx].in_cables = + (1 << ms_ep->bNumEmbMIDIJack) - 1; dev_dbg(&umidi->dev->dev, "EP %02X: %d jack(s)\n", - ep->bEndpointAddress, ms_ep->bNumEmbMIDIJack); + ep->bEndpointAddress, ms_ep->bNumEmbMIDIJack); } } return 0; @@ -1825,7 +1852,7 @@ static int roland_load_get(struct snd_kcontrol *kcontrol, static int roland_load_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *value) { - struct snd_usb_midi* umidi = kcontrol->private_data; + struct snd_usb_midi *umidi = kcontrol->private_data; int changed; if (value->value.enumerated.item[0] > 1) @@ -1851,11 +1878,11 @@ static struct snd_kcontrol_new roland_load_ctl = { * On Roland devices, use the second alternate setting to be able to use * the interrupt input endpoint. */ -static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi* umidi) +static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi *umidi) { - struct usb_interface* intf; + struct usb_interface *intf; struct usb_host_interface *hostif; - struct usb_interface_descriptor* intfd; + struct usb_interface_descriptor *intfd; intf = umidi->iface; if (!intf || intf->num_altsetting != 2) @@ -1864,8 +1891,10 @@ static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi* umidi) hostif = &intf->altsetting[1]; intfd = get_iface_desc(hostif); if (intfd->bNumEndpoints != 2 || - (get_endpoint(hostif, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK || - (get_endpoint(hostif, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT) + (get_endpoint(hostif, 0)->bmAttributes & + USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK || + (get_endpoint(hostif, 1)->bmAttributes & + USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT) return; dev_dbg(&umidi->dev->dev, "switching to altsetting %d with int ep\n", @@ -1881,14 +1910,14 @@ static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi* umidi) /* * Try to find any usable endpoints in the interface. */ -static int snd_usbmidi_detect_endpoints(struct snd_usb_midi* umidi, - struct snd_usb_midi_endpoint_info* endpoint, +static int snd_usbmidi_detect_endpoints(struct snd_usb_midi *umidi, + struct snd_usb_midi_endpoint_info *endpoint, int max_endpoints) { - struct usb_interface* intf; + struct usb_interface *intf; struct usb_host_interface *hostif; - struct usb_interface_descriptor* intfd; - struct usb_endpoint_descriptor* epd; + struct usb_interface_descriptor *intfd; + struct usb_endpoint_descriptor *epd; int i, out_eps = 0, in_eps = 0; if (USB_ID_VENDOR(umidi->usb_id) == 0x0582) @@ -1929,8 +1958,8 @@ static int snd_usbmidi_detect_endpoints(struct snd_usb_midi* umidi, /* * Detects the endpoints for one-port-per-endpoint protocols. */ -static int snd_usbmidi_detect_per_port_endpoints(struct snd_usb_midi* umidi, - struct snd_usb_midi_endpoint_info* endpoints) +static int snd_usbmidi_detect_per_port_endpoints(struct snd_usb_midi *umidi, + struct snd_usb_midi_endpoint_info *endpoints) { int err, i; @@ -1947,13 +1976,13 @@ static int snd_usbmidi_detect_per_port_endpoints(struct snd_usb_midi* umidi, /* * Detects the endpoints and ports of Yamaha devices. */ -static int snd_usbmidi_detect_yamaha(struct snd_usb_midi* umidi, - struct snd_usb_midi_endpoint_info* endpoint) +static int snd_usbmidi_detect_yamaha(struct snd_usb_midi *umidi, + struct snd_usb_midi_endpoint_info *endpoint) { - struct usb_interface* intf; + struct usb_interface *intf; struct usb_host_interface *hostif; - struct usb_interface_descriptor* intfd; - uint8_t* cs_desc; + struct usb_interface_descriptor *intfd; + uint8_t *cs_desc; intf = umidi->iface; if (!intf) @@ -1972,9 +2001,11 @@ static int snd_usbmidi_detect_yamaha(struct snd_usb_midi* umidi, cs_desc += cs_desc[0]) { if (cs_desc[1] == USB_DT_CS_INTERFACE) { if (cs_desc[2] == UAC_MIDI_IN_JACK) - endpoint->in_cables = (endpoint->in_cables << 1) | 1; + endpoint->in_cables = + (endpoint->in_cables << 1) | 1; else if (cs_desc[2] == UAC_MIDI_OUT_JACK) - endpoint->out_cables = (endpoint->out_cables << 1) | 1; + endpoint->out_cables = + (endpoint->out_cables << 1) | 1; } } if (!endpoint->in_cables && !endpoint->out_cables) @@ -1986,12 +2017,12 @@ static int snd_usbmidi_detect_yamaha(struct snd_usb_midi* umidi, /* * Detects the endpoints and ports of Roland devices. */ -static int snd_usbmidi_detect_roland(struct snd_usb_midi* umidi, - struct snd_usb_midi_endpoint_info* endpoint) +static int snd_usbmidi_detect_roland(struct snd_usb_midi *umidi, + struct snd_usb_midi_endpoint_info *endpoint) { - struct usb_interface* intf; + struct usb_interface *intf; struct usb_host_interface *hostif; - u8* cs_desc; + u8 *cs_desc; intf = umidi->iface; if (!intf) @@ -2024,14 +2055,14 @@ static int snd_usbmidi_detect_roland(struct snd_usb_midi* umidi, /* * Creates the endpoints and their ports for Midiman devices. */ -static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi, - struct snd_usb_midi_endpoint_info* endpoint) +static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi *umidi, + struct snd_usb_midi_endpoint_info *endpoint) { struct snd_usb_midi_endpoint_info ep_info; - struct usb_interface* intf; + struct usb_interface *intf; struct usb_host_interface *hostif; - struct usb_interface_descriptor* intfd; - struct usb_endpoint_descriptor* epd; + struct usb_interface_descriptor *intfd; + struct usb_endpoint_descriptor *epd; int cable, err; intf = umidi->iface; @@ -2068,39 +2099,50 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi, epd = get_endpoint(hostif, 4); if (!usb_endpoint_dir_out(epd) || !usb_endpoint_xfer_bulk(epd)) { - dev_dbg(&umidi->dev->dev, "endpoint[4] isn't bulk output\n"); + dev_dbg(&umidi->dev->dev, + "endpoint[4] isn't bulk output\n"); return -ENXIO; } } - ep_info.out_ep = get_endpoint(hostif, 2)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK; + ep_info.out_ep = get_endpoint(hostif, 2)->bEndpointAddress & + USB_ENDPOINT_NUMBER_MASK; ep_info.out_interval = 0; ep_info.out_cables = endpoint->out_cables & 0x5555; - err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]); + err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, + &umidi->endpoints[0]); if (err < 0) return err; - ep_info.in_ep = get_endpoint(hostif, 0)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK; + ep_info.in_ep = get_endpoint(hostif, 0)->bEndpointAddress & + USB_ENDPOINT_NUMBER_MASK; ep_info.in_interval = get_endpoint(hostif, 0)->bInterval; ep_info.in_cables = endpoint->in_cables; - err = snd_usbmidi_in_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]); + err = snd_usbmidi_in_endpoint_create(umidi, &ep_info, + &umidi->endpoints[0]); if (err < 0) return err; if (endpoint->out_cables > 0x0001) { - ep_info.out_ep = get_endpoint(hostif, 4)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK; + ep_info.out_ep = get_endpoint(hostif, 4)->bEndpointAddress & + USB_ENDPOINT_NUMBER_MASK; ep_info.out_cables = endpoint->out_cables & 0xaaaa; - err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[1]); + err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, + &umidi->endpoints[1]); if (err < 0) return err; } for (cable = 0; cable < 0x10; ++cable) { if (endpoint->out_cables & (1 << cable)) - snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_OUTPUT, cable, + snd_usbmidi_init_substream(umidi, + SNDRV_RAWMIDI_STREAM_OUTPUT, + cable, &umidi->endpoints[cable & 1].out->ports[cable].substream); if (endpoint->in_cables & (1 << cable)) - snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_INPUT, cable, + snd_usbmidi_init_substream(umidi, + SNDRV_RAWMIDI_STREAM_INPUT, + cable, &umidi->endpoints[0].in->ports[cable].substream); } return 0; @@ -2110,7 +2152,7 @@ static struct snd_rawmidi_global_ops snd_usbmidi_ops = { .get_port_info = snd_usbmidi_get_port_info, }; -static int snd_usbmidi_create_rawmidi(struct snd_usb_midi* umidi, +static int snd_usbmidi_create_rawmidi(struct snd_usb_midi *umidi, int out_ports, int in_ports) { struct snd_rawmidi *rmidi; @@ -2128,8 +2170,10 @@ static int snd_usbmidi_create_rawmidi(struct snd_usb_midi* umidi, rmidi->ops = &snd_usbmidi_ops; rmidi->private_data = umidi; rmidi->private_free = snd_usbmidi_rawmidi_free; - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_usbmidi_output_ops); - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_usbmidi_input_ops); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &snd_usbmidi_output_ops); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &snd_usbmidi_input_ops); umidi->rmidi = rmidi; return 0; @@ -2138,16 +2182,16 @@ static int snd_usbmidi_create_rawmidi(struct snd_usb_midi* umidi, /* * Temporarily stop input. */ -void snd_usbmidi_input_stop(struct list_head* p) +void snd_usbmidi_input_stop(struct list_head *p) { - struct snd_usb_midi* umidi; + struct snd_usb_midi *umidi; unsigned int i, j; umidi = list_entry(p, struct snd_usb_midi, list); if (!umidi->input_running) return; for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { - struct snd_usb_midi_endpoint* ep = &umidi->endpoints[i]; + struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i]; if (ep->in) for (j = 0; j < INPUT_URBS; ++j) usb_kill_urb(ep->in->urbs[j]); @@ -2156,14 +2200,14 @@ void snd_usbmidi_input_stop(struct list_head* p) } EXPORT_SYMBOL(snd_usbmidi_input_stop); -static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint* ep) +static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint *ep) { unsigned int i; if (!ep) return; for (i = 0; i < INPUT_URBS; ++i) { - struct urb* urb = ep->urbs[i]; + struct urb *urb = ep->urbs[i]; urb->dev = ep->umidi->dev; snd_usbmidi_submit_urb(urb, GFP_KERNEL); } @@ -2172,9 +2216,9 @@ static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint* ep) /* * Resume input after a call to snd_usbmidi_input_stop(). */ -void snd_usbmidi_input_start(struct list_head* p) +void snd_usbmidi_input_start(struct list_head *p) { - struct snd_usb_midi* umidi; + struct snd_usb_midi *umidi; int i; umidi = list_entry(p, struct snd_usb_midi, list); @@ -2187,14 +2231,42 @@ void snd_usbmidi_input_start(struct list_head* p) EXPORT_SYMBOL(snd_usbmidi_input_start); /* + * Prepare for suspend. Typically called from the USB suspend callback. + */ +void snd_usbmidi_suspend(struct list_head *p) +{ + struct snd_usb_midi *umidi; + + umidi = list_entry(p, struct snd_usb_midi, list); + mutex_lock(&umidi->mutex); + snd_usbmidi_input_stop(p); + mutex_unlock(&umidi->mutex); +} +EXPORT_SYMBOL(snd_usbmidi_suspend); + +/* + * Resume. Typically called from the USB resume callback. + */ +void snd_usbmidi_resume(struct list_head *p) +{ + struct snd_usb_midi *umidi; + + umidi = list_entry(p, struct snd_usb_midi, list); + mutex_lock(&umidi->mutex); + snd_usbmidi_input_start(p); + mutex_unlock(&umidi->mutex); +} +EXPORT_SYMBOL(snd_usbmidi_resume); + +/* * Creates and registers everything needed for a MIDI streaming interface. */ int snd_usbmidi_create(struct snd_card *card, - struct usb_interface* iface, + struct usb_interface *iface, struct list_head *midi_list, - const struct snd_usb_audio_quirk* quirk) + const struct snd_usb_audio_quirk *quirk) { - struct snd_usb_midi* umidi; + struct snd_usb_midi *umidi; struct snd_usb_midi_endpoint_info endpoints[MIDI_MAX_ENDPOINTS]; int out_ports, in_ports; int i, err; @@ -2292,7 +2364,8 @@ int snd_usbmidi_create(struct snd_card *card, err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; default: - dev_err(&umidi->dev->dev, "invalid quirk type %d\n", quirk->type); + dev_err(&umidi->dev->dev, "invalid quirk type %d\n", + quirk->type); err = -ENXIO; break; } diff --git a/sound/usb/midi.h b/sound/usb/midi.h index 2fca80b744c0..ad8a3211f8e7 100644 --- a/sound/usb/midi.h +++ b/sound/usb/midi.h @@ -43,8 +43,10 @@ int snd_usbmidi_create(struct snd_card *card, struct usb_interface *iface, struct list_head *midi_list, const struct snd_usb_audio_quirk *quirk); -void snd_usbmidi_input_stop(struct list_head* p); -void snd_usbmidi_input_start(struct list_head* p); +void snd_usbmidi_input_stop(struct list_head *p); +void snd_usbmidi_input_start(struct list_head *p); void snd_usbmidi_disconnect(struct list_head *p); +void snd_usbmidi_suspend(struct list_head *p); +void snd_usbmidi_resume(struct list_head *p); #endif /* __USBMIDI_H */ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 0b728d886f0d..2e4a9dbc51fa 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1340,12 +1340,11 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, */ if (range > 384) { usb_audio_warn(state->chip, - "Warning! Unlikely big volume range (=%u), " - "cval->res is probably wrong.", + "Warning! Unlikely big volume range (=%u), cval->res is probably wrong.", range); - usb_audio_warn(state->chip, "[%d] FU [%s] ch = %d, " - "val = %d/%d/%d", cval->id, - kctl->id.name, cval->channels, + usb_audio_warn(state->chip, + "[%d] FU [%s] ch = %d, val = %d/%d/%d", + cval->id, kctl->id.name, cval->channels, cval->min, cval->max, cval->res); } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7c57f2268dd7..19a921eb75f1 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -670,7 +670,7 @@ static int snd_usb_gamecon780_boot_quirk(struct usb_device *dev) /* set the initial volume and don't change; other values are either * too loud or silent due to firmware bug (bko#65251) */ - u8 buf[2] = { 0x74, 0xdc }; + u8 buf[2] = { 0x74, 0xe3 }; return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, UAC_FU_VOLUME << 8, 9 << 8, buf, 2); |