diff options
Diffstat (limited to 'Documentation')
10 files changed, 430 insertions, 3 deletions
diff --git a/Documentation/devicetree/bindings/sound/cs4270.txt b/Documentation/devicetree/bindings/sound/cs4270.txt new file mode 100644 index 000000000000..6b222f9b8ef5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4270.txt @@ -0,0 +1,21 @@ +CS4270 audio CODEC + +The driver for this device currently only supports I2C. + +Required properties: + + - compatible : "cirrus,cs4270" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset-gpio : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + +Example: + +codec: cs4270@48 { + compatible = "cirrus,cs4270"; + reg = <0x48>; +}; diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt new file mode 100644 index 000000000000..c81b5fd5a5bc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -0,0 +1,36 @@ +Cirrus Logic CS4271 DT bindings + +This driver supports both the I2C and the SPI bus. + +Required properties: + + - compatible: "cirrus,cs4271" + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Required properties on I2C: + + - reg: the i2c address + + +Optional properties: + + - reset-gpio: a GPIO spec to define which pin is connected to the chip's + !RESET pin + +Examples: + + codec_i2c: cs4271@10 { + compatible = "cirrus,cs4271"; + reg = <0x10>; + reset-gpio = <&gpio 23 0>; + }; + + codec_spi: cs4271@0 { + compatible = "cirrus,cs4271"; + reg = <0x0>; + reset-gpio = <&gpio 23 0>; + spi-max-frequency = <6000000>; + }; + diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt new file mode 100644 index 000000000000..374e145c2ef1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -0,0 +1,45 @@ +Texas Instruments McASP controller + +Required properties: +- compatible : + "ti,dm646x-mcasp-audio" : for DM646x platforms + "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms + "ti,omap2-mcasp-audio" : for OMAP2 platforms (TI81xx, AM33xx) + +- reg : Should contain McASP registers offset and length +- interrupts : Interrupt number for McASP +- op-mode : I2S/DIT ops mode. +- tdm-slots : Slots for TDM operation. +- num-serializer : Serializers used by McASP. +- serial-dir : A list of serializer pin mode. The list number should be equal + to "num-serializer" parameter. Each entry is a number indication + serializer pin direction. (0 - INACTIVE, 1 - TX, 2 - RX) + + +Optional properties: + +- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0 +- tx-num-evt : FIFO levels. +- rx-num-evt : FIFO levels. +- sram-size-playback : size of sram to be allocated during playback +- sram-size-capture : size of sram to be allocated during capture + +Example: + +mcasp0: mcasp0@1d00000 { + compatible = "ti,da830-mcasp-audio"; + #address-cells = <1>; + #size-cells = <0>; + reg = <0x100000 0x3000>; + interrupts = <82 83>; + op-mode = <0>; /* MCASP_IIS_MODE */ + tdm-slots = <2>; + num-serializer = <16>; + serial-dir = < + 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ + 0 0 0 0 + 0 0 0 1 + 2 0 0 0 >; + tx-num-evt = <1>; + rx-num-evt = <1>; +}; diff --git a/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt new file mode 100644 index 000000000000..65dec876cb2d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt @@ -0,0 +1,91 @@ +* Texas Instruments OMAP4+ and twl6040 based audio setups + +Required properties: +- compatible: "ti,abe-twl6040" +- ti,model: Name of the sound card ( for example "SDP4430") +- ti,mclk-freq: MCLK frequency for HPPLL operation +- ti,mcpdm: phandle for the McPDM node +- ti,twl6040: phandle for the twl6040 core node +- ti,audio-routing: List of 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. + +Optional properties: +- ti,dmic: phandle for the OMAP dmic node if the machine have it connected +- ti,jack_detection: Need to be set to <1> if the board capable to detect jack + insertion, removal. + +Available audio endpoints for the audio-routing table: + +Board connectors: + * Headset Stereophone + * Earphone Spk + * Ext Spk + * Line Out + * Vibrator + * Headset Mic + * Main Handset Mic + * Sub Handset Mic + * Line In + * Digital Mic + +twl6040 pins: + * HSOL + * HSOR + * EP + * HFL + * HFR + * AUXL + * AUXR + * VIBRAL + * VIBRAR + * HSMIC + * MAINMIC + * SUBMIC + * AFML + * AFMR + + * Headset Mic Bias + * Main Mic Bias + * Digital Mic1 Bias + * Digital Mic2 Bias + +Digital mic pins: + * DMic + +Example: + +sound { + compatible = "ti,abe-twl6040"; + ti,model = "SDP4430"; + + ti,jack-detection = <1>; + ti,mclk-freq = <38400000>; + + ti,mcpdm = <&mcpdm>; + ti,dmic = <&dmic>; + + ti,twl6040 = <&twl6040>; + + /* Audio routing */ + ti,audio-routing = + "Headset Stereophone", "HSOL", + "Headset Stereophone", "HSOR", + "Earphone Spk", "EP", + "Ext Spk", "HFL", + "Ext Spk", "HFR", + "Line Out", "AUXL", + "Line Out", "AUXR", + "Vibrator", "VIBRAL", + "Vibrator", "VIBRAR", + "HSMIC", "Headset Mic", + "Headset Mic", "Headset Mic Bias", + "MAINMIC", "Main Handset Mic", + "Main Handset Mic", "Main Mic Bias", + "SUBMIC", "Sub Handset Mic", + "Sub Handset Mic", "Main Mic Bias", + "AFML", "Line In", + "AFMR", "Line In", + "DMic", "Digital Mic", + "Digital Mic", "Digital Mic1 Bias"; +}; diff --git a/Documentation/devicetree/bindings/sound/omap-mcbsp.txt b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt new file mode 100644 index 000000000000..17cce4490456 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt @@ -0,0 +1,37 @@ +* Texas Instruments OMAP2+ McBSP module + +Required properties: +- compatible: "ti,omap2420-mcbsp" for McBSP on OMAP2420 + "ti,omap2430-mcbsp" for McBSP on OMAP2430 + "ti,omap3-mcbsp" for McBSP on OMAP3 + "ti,omap4-mcbsp" for McBSP on OMAP4 and newer SoC +- reg: Register location and size, for OMAP4+ as an array: + <MPU access base address, size>, + <L3 interconnect address, size>; +- reg-names: Array of strings associated with the address space +- interrupts: Interrupt numbers for the McBSP port, as an array in case the + McBSP IP have more interrupt lines: + <OCP compliant irq>, + <TX irq>, + <RX irq>; +- interrupt-names: Array of strings associated with the interrupt numbers +- interrupt-parent: The parent interrupt controller +- ti,buffer-size: Size of the FIFO on the port (OMAP2430 and newer SoC) +- ti,hwmods: Name of the hwmod associated to the McBSP port + +Example: + +mcbsp2: mcbsp@49022000 { + compatible = "ti,omap3-mcbsp"; + reg = <0x49022000 0xff>, + <0x49028000 0xff>; + reg-names = "mpu", "sidetone"; + interrupts = <0 17 0x4>, /* OCP compliant interrupt */ + <0 62 0x4>, /* TX interrupt */ + <0 63 0x4>, /* RX interrupt */ + <0 4 0x4>; /* Sidetone */ + interrupt-names = "common", "tx", "rx", "sidetone"; + interrupt-parent = <&intc>; + ti,buffer-size = <1280>; + ti,hwmods = "mcbsp2"; +}; diff --git a/Documentation/devicetree/bindings/sound/omap-twl4030.txt b/Documentation/devicetree/bindings/sound/omap-twl4030.txt new file mode 100644 index 000000000000..6fae51c7f766 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-twl4030.txt @@ -0,0 +1,17 @@ +* Texas Instruments SoC with twl4030 based audio setups + +Required properties: +- compatible: "ti,omap-twl4030" +- ti,model: Name of the sound card (for example "omap3beagle") +- ti,mcbsp: phandle for the McBSP node +- ti,codec: phandle for the twl4030 audio node + +Example: + +sound { + compatible = "ti,omap-twl4030"; + ti,model = "omap3beagle"; + + ti,mcbsp = <&mcbsp2>; + ti,codec = <&twl_audio>; +}; diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt new file mode 100644 index 000000000000..e7b98f41fa5f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -0,0 +1,20 @@ +Texas Instruments - tlv320aic3x Codec module + +The tlv320aic3x serial control bus communicates through I2C protocols + +Required properties: +- compatible - "string" - "ti,tlv320aic3x" +- reg - <int> - I2C slave address + + +Optional properties: + +- gpio-reset - gpio pin number used for codec reset +- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality + +Example: + +tlv320aic3x: tlv320aic3x@1b { + compatible = "ti,tlv320aic3x"; + reg = <0x1b>; +}; diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 4e4d0bc9816f..d90d8ec2853d 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -860,8 +860,14 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. [Multiple options for each card instance] model - force the model name - position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF, - 3 = VIACOMBO, 4 = COMBO) + position_fix - Fix DMA pointer + -1 = system default: choose appropriate one per controller + hardware + 0 = auto: falls back to LPIB when POSBUF doesn't work + 1 = use LPIB + 2 = POSBUF: use position buffer + 3 = VIACOMBO: VIA-specific workaround for capture + 4 = COMBO: use LPIB for playback, auto for capture stream probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) When the bit 8 (0x100) is set, the lower 8 bits are used as the "fixed" codec slots; i.e. the driver probes the diff --git a/Documentation/sound/alsa/Channel-Mapping-API.txt b/Documentation/sound/alsa/Channel-Mapping-API.txt new file mode 100644 index 000000000000..3c43d1a4ca0e --- /dev/null +++ b/Documentation/sound/alsa/Channel-Mapping-API.txt @@ -0,0 +1,153 @@ +ALSA PCM channel-mapping API +============================ + Takashi Iwai <tiwai@suse.de> + +GENERAL +------- + +The channel mapping API allows user to query the possible channel maps +and the current channel map, also optionally to modify the channel map +of the current stream. + +A channel map is an array of position for each PCM channel. +Typically, a stereo PCM stream has a channel map of + { front_left, front_right } +while a 4.0 surround PCM stream has a channel map of + { front left, front right, rear left, rear right }. + +The problem, so far, was that we had no standard channel map +explicitly, and applications had no way to know which channel +corresponds to which (speaker) position. Thus, applications applied +wrong channels for 5.1 outputs, and you hear suddenly strange sound +from rear. Or, some devices secretly assume that center/LFE is the +third/fourth channels while others that C/LFE as 5th/6th channels. + +Also, some devices such as HDMI are configurable for different speaker +positions even with the same number of total channels. However, there +was no way to specify this because of lack of channel map +specification. These are the main motivations for the new channel +mapping API. + + +DESIGN +------ + +Actually, "the channel mapping API" doesn't introduce anything new in +the kernel/user-space ABI perspective. It uses only the existing +control element features. + +As a ground design, each PCM substream may contain a control element +providing the channel mapping information and configuration. This +element is specified by: + iface = SNDRV_CTL_ELEM_IFACE_PCM + name = "Playback Channel Map" or "Capture Channel Map" + device = the same device number for the assigned PCM substream + index = the same index number for the assigned PCM substream + +Note the name is different depending on the PCM substream direction. + +Each control element provides at least the TLV read operation and the +read operation. Optionally, the write operation can be provided to +allow user to change the channel map dynamically. + +* TLV + +The TLV operation gives the list of available channel +maps. A list item of a channel map is usually a TLV of + type data-bytes ch0 ch1 ch2... +where type is the TLV type value, the second argument is the total +bytes (not the numbers) of channel values, and the rest are the +position value for each channel. + +As a TLV type, either SNDRV_CTL_TLVT_CHMAP_FIXED, +SNDRV_CTL_TLV_CHMAP_VAR or SNDRV_CTL_TLVT_CHMAP_PAIRED can be used. +The _FIXED type is for a channel map with the fixed channel position +while the latter two are for flexible channel positions. _VAR type is +for a channel map where all channels are freely swappable and _PAIRED +type is where pair-wise channels are swappable. For example, when you +have {FL/FR/RL/RR} channel map, _PAIRED type would allow you to swap +only {RL/RR/FL/FR} while _VAR type would allow even swapping FL and +RR. + +These new TLV types are defined in sound/tlv.h. + +The available channel position values are defined in sound/asound.h, +here is a cut: + +/* channel positions */ +enum { + SNDRV_CHMAP_UNKNOWN = 0, + SNDRV_CHMAP_NA, /* N/A, silent */ + SNDRV_CHMAP_MONO, /* mono stream */ + /* this follows the alsa-lib mixer channel value + 3 */ + SNDRV_CHMAP_FL, /* front left */ + SNDRV_CHMAP_FR, /* front right */ + SNDRV_CHMAP_RL, /* rear left */ + SNDRV_CHMAP_RR, /* rear right */ + SNDRV_CHMAP_FC, /* front center */ + SNDRV_CHMAP_LFE, /* LFE */ + SNDRV_CHMAP_SL, /* side left */ + SNDRV_CHMAP_SR, /* side right */ + SNDRV_CHMAP_RC, /* rear center */ + /* new definitions */ + SNDRV_CHMAP_FLC, /* front left center */ + SNDRV_CHMAP_FRC, /* front right center */ + SNDRV_CHMAP_RLC, /* rear left center */ + SNDRV_CHMAP_RRC, /* rear right center */ + SNDRV_CHMAP_FLW, /* front left wide */ + SNDRV_CHMAP_FRW, /* front right wide */ + SNDRV_CHMAP_FLH, /* front left high */ + SNDRV_CHMAP_FCH, /* front center high */ + SNDRV_CHMAP_FRH, /* front right high */ + SNDRV_CHMAP_TC, /* top center */ + SNDRV_CHMAP_TFL, /* top front left */ + SNDRV_CHMAP_TFR, /* top front right */ + SNDRV_CHMAP_TFC, /* top front center */ + SNDRV_CHMAP_TRL, /* top rear left */ + SNDRV_CHMAP_TRR, /* top rear right */ + SNDRV_CHMAP_TRC, /* top rear center */ + SNDRV_CHMAP_LAST = SNDRV_CHMAP_TRC, +}; + +When a PCM stream can provide more than one channel map, you can +provide multiple channel maps in a TLV container type. The TLV data +to be returned will contain such as: + SNDRV_CTL_TLVT_CONTAINER 96 + SNDRV_CTL_TLVT_CHMAP_FIXED 4 SNDRV_CHMAP_FC + SNDRV_CTL_TLVT_CHMAP_FIXED 8 SNDRV_CHMAP_FL SNDRV_CHMAP_FR + SNDRV_CTL_TLVT_CHMAP_FIXED 16 NDRV_CHMAP_FL SNDRV_CHMAP_FR \ + SNDRV_CHMAP_RL SNDRV_CHMAP_RR + +The channel position is provided in LSB 16bits. The upper bits are +used for bit flags. + +#define SNDRV_CHMAP_POSITION_MASK 0xffff +#define SNDRV_CHMAP_PHASE_INVERSE (0x01 << 16) +#define SNDRV_CHMAP_DRIVER_SPEC (0x02 << 16) + +SNDRV_CHMAP_PHASE_INVERSE indicates the channel is phase inverted, +(thus summing left and right channels would result in almost silence). +Some digital mic devices have this. + +When SNDRV_CHMAP_DRIVER_SPEC is set, all the channel position values +don't follow the standard definition above but driver-specific. + +* READ OPERATION + +The control read operation is for providing the current channel map of +the given stream. The control element returns an integer array +containing the position of each channel. + +When this is performed before the number of the channel is specified +(i.e. hw_params is set), it should return all channels set to +UNKNOWN. + +* WRITE OPERATION + +The control write operation is optional, and only for devices that can +change the channel configuration on the fly, such as HDMI. User needs +to pass an integer value containing the valid channel positions for +all channels of the assigned PCM substream. + +This operation is allowed only at PCM PREPARED state. When called in +other states, it shall return an error. diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index a92bba816843..16dfe57f1731 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -74,7 +74,8 @@ CMI9880 AD1882 / AD1882A ================ - 3stack 3-stack mode (default) + 3stack 3-stack mode + 3stack-automute 3-stack with automute front HP (default) 6stack 6-stack mode AD1884A / AD1883 / AD1984A / AD1984B |