diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2015-06-25 17:15:18 -0700 |
---|---|---|
committer | Linus Torvalds <torvalds@linux-foundation.org> | 2015-06-25 17:15:18 -0700 |
commit | 4570a37169d4b44d316f40b2ccc681dc93fedc7b (patch) | |
tree | cafffb586c60dddfb04b8619fa1ae0e859600de7 /Documentation | |
parent | f7b08217c755e88a29b5bd53b4a1d10cd8b3c5f8 (diff) | |
parent | 60b93030b44a8c2cd015cebe5624fd7552ec67ec (diff) |
Merge tag 'sound-4.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"It was a busy development cycle at this time, as you can see a wide
range of changes in diffstat. There are no big changes but many
refactoring and improvements. Here we go some highlights:
ALSA core:
- Procfs codes were cleaned up to use seq_file
- Procfs can be opt out via Kconfig (only for EXPERT)
- Two types of jack API were unified finally; now both kctl and input
jack devs are handled via a single function call.
HD-audio:
- Continued code restructuring for the future ASoC driver; now HDA
controller driver is split to a core helper module.
- Preliminary codes for Skylake audio support in HDA core.
- Proper i915 gfx power well management for SKL & co
- Enabled runtime PM as default for Intel HDMI/DP codecs
- Newer Tegra chip supports
- More quirks for Dell headsets, Alienware (with CA0132), etc.
- A couple of DRM ELD helper API functions
ASoC:
- Support for loading ASoC topology maps from firmware, intended to
be used to allow self-describing DSP firmware images to be built
which can map controls added by the DSP to userspace without the
kernel needing to know about individual DSP firmwares
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring
- Big refactoring, cleanup and enhancement for the Wolfson ADSP
driver
- Cleanup series for TI TAS2552 and R-CAR drivers
- Fixes and improvements on RT56xx codecs
- Support for TI TAS571x power amplifiers
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs
- Support for x86 systems with RT5650 and Qualcomm Storm
- Support for Mediatek AFE (Audio Front End) unit
- Other various small fixes to ASoC codec drivers
Firewire:
- Enhanced to allow non-blocking streams to use timestamp
synchronization
- Improve support for DM1500 and BeBoBv3
Misc:
- Cleanup of old pci API functions over all PCI sound drivers
- Fix long-standing regression of the old powermac i2c setup"
* tag 'sound-4.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (533 commits)
ALSA: pcm: Fix pcm_class sysfs output
ALSA: hda-beep: Update authors dead email address
ASoC: wm_adsp: Move DSP Rate controls into the codec
ASoC: wm8995: Fix setting sysclk for WM8995_SYSCLK_MCLK2 case
ALSA: hda: provide default bus io ops extended hdac
ALSA: hda: add hda link cleanup routine
ALSA: hda: add hdac_ext stream creation and cleanup routines
ASoC: rsrc-card: remove unused ret
ALSA: HDAC: move SND_HDA_PREALLOC_SIZE to core
ASoC: mediatek: Add machine driver for rt5650 rt5676 codec
ASoC: mediatek: Add machine driver for MAX98090 codec
ASoC: mediatek: Add AFE platform driver
ASoC: rsnd: remove io from rsnd_mod
ASoC: rsnd: move rsnd_mod_is_working() to rsnd_io_is_working()
ASoC: rsnd: don't use rsnd_mod_to_io() on snd_kcontrol
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_src_xxx()
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_ssi_xxx()
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_dma_xxx()
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_get_adinr()
ASoC: rsnd: add common interrupt handler for SSI/SRC/DMA
...
Diffstat (limited to 'Documentation')
24 files changed, 449 insertions, 8 deletions
diff --git a/Documentation/devicetree/bindings/sound/adi,adau1701.txt b/Documentation/devicetree/bindings/sound/adi,adau1701.txt index 547a49b56a62..0d1128ce2ea7 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau1701.txt +++ b/Documentation/devicetree/bindings/sound/adi,adau1701.txt @@ -20,6 +20,8 @@ Optional properties: pin configurations as described in the datasheet, table 53. Note that the value of this property has to be prefixed with '/bits/ 8'. + - avdd-supply: Power supply for AVDD, providing 3.3V + - dvdd-supply: Power supply for DVDD, providing 3.3V Examples: @@ -28,6 +30,8 @@ Examples: compatible = "adi,adau1701"; reg = <0x34>; reset-gpio = <&gpio 23 0>; + avdd-supply = <&vdd_3v3_reg>; + dvdd-supply = <&vdd_3v3_reg>; adi,pll-mode-gpios = <&gpio 24 0 &gpio 25 0>; adi,pin-config = /bits/ 8 <0x4 0x7 0x5 0x5 0x4 0x4 0x4 0x4 0x4 0x4 0x4 0x4>; diff --git a/Documentation/devicetree/bindings/sound/bt-sco.txt b/Documentation/devicetree/bindings/sound/bt-sco.txt new file mode 100644 index 000000000000..29b8e5d40203 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/bt-sco.txt @@ -0,0 +1,13 @@ +Bluetooth-SCO audio CODEC + +This device support generic Bluetooth SCO link. + +Required properties: + + - compatible : "delta,dfbmcs320" + +Example: + +codec: bt_sco { + compatible = "delta,dfbmcs320"; +}; diff --git a/Documentation/devicetree/bindings/sound/gtm601.txt b/Documentation/devicetree/bindings/sound/gtm601.txt new file mode 100644 index 000000000000..5efc8c068de0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/gtm601.txt @@ -0,0 +1,13 @@ +GTM601 UMTS modem audio interface CODEC + +This device has no configuration interface. Sample rate is fixed - 8kHz. + +Required properties: + + - compatible : "option,gtm601" + +Example: + +codec: gtm601_codec { + compatible = "option,gtm601"; +}; diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index aa802a274520..4e3be6682c98 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -18,6 +18,12 @@ Optional properties: - maxim,dmic-freq: Frequency at which to clock DMIC +- maxim,micbias: Micbias voltage applies to the analog mic, valid voltages value are: + 0 - 2.2v + 1 - 2.55v + 2 - 2.4v + 3 - 2.8v + Pins on the device (for linking into audio routes): * MIC1 diff --git a/Documentation/devicetree/bindings/sound/mt8173-max98090.txt b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt new file mode 100644 index 000000000000..829bd26d17f8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt @@ -0,0 +1,13 @@ +MT8173 with MAX98090 CODEC + +Required properties: +- compatible : "mediatek,mt8173-max98090" +- mediatek,audio-codec: the phandle of the MAX98090 audio codec + +Example: + + sound { + compatible = "mediatek,mt8173-max98090"; + mediatek,audio-codec = <&max98090>; + }; + diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt new file mode 100644 index 000000000000..61e98c976bd4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt @@ -0,0 +1,13 @@ +MT8173 with RT5650 RT5676 CODECS + +Required properties: +- compatible : "mediatek,mt8173-rt5650-rt5676" +- mediatek,audio-codec: the phandles of rt5650 and rt5676 codecs + +Example: + + sound { + compatible = "mediatek,mt8173-rt5650-rt5676"; + mediatek,audio-codec = <&rt5650 &rt5676>; + }; + diff --git a/Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt new file mode 100644 index 000000000000..e302c7f43b95 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt @@ -0,0 +1,45 @@ +Mediatek AFE PCM controller + +Required properties: +- compatible = "mediatek,mt8173-afe-pcm"; +- reg: register location and size +- interrupts: Should contain AFE interrupt +- clock-names: should have these clock names: + "infra_sys_audio_clk", + "top_pdn_audio", + "top_pdn_aud_intbus", + "bck0", + "bck1", + "i2s0_m", + "i2s1_m", + "i2s2_m", + "i2s3_m", + "i2s3_b"; + +Example: + + afe: mt8173-afe-pcm@11220000 { + compatible = "mediatek,mt8173-afe-pcm"; + reg = <0 0x11220000 0 0x1000>; + interrupts = <GIC_SPI 134 IRQ_TYPE_EDGE_FALLING>; + clocks = <&infracfg INFRA_AUDIO>, + <&topckgen TOP_AUDIO_SEL>, + <&topckgen TOP_AUD_INTBUS_SEL>, + <&topckgen TOP_APLL1_DIV0>, + <&topckgen TOP_APLL2_DIV0>, + <&topckgen TOP_I2S0_M_CK_SEL>, + <&topckgen TOP_I2S1_M_CK_SEL>, + <&topckgen TOP_I2S2_M_CK_SEL>, + <&topckgen TOP_I2S3_M_CK_SEL>, + <&topckgen TOP_I2S3_B_CK_SEL>; + clock-names = "infra_sys_audio_clk", + "top_pdn_audio", + "top_pdn_aud_intbus", + "bck0", + "bck1", + "i2s0_m", + "i2s1_m", + "i2s2_m", + "i2s3_m", + "i2s3_b"; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt new file mode 100644 index 000000000000..48129368d4d9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt @@ -0,0 +1,60 @@ +* Qualcomm Technologies APQ8016 SBC ASoC machine driver + +This node models the Qualcomm Technologies APQ8016 SBC ASoC machine driver + +Required properties: + +- compatible : "qcom,apq8016-sbc-sndcard" + +- pinctrl-N : One property must exist for each entry in + pinctrl-names. See ../pinctrl/pinctrl-bindings.txt + for details of the property values. +- pinctrl-names : Must contain a "default" entry. +- reg : Must contain an address for each entry in reg-names. +- reg-names : A list which must include the following entries: + * "mic-iomux" + * "spkr-iomux" +- qcom,model : Name of the sound card. + +Dai-link subnode properties and subnodes: + +Required dai-link subnodes: + +- cpu : CPU sub-node +- codec : CODEC sub-node + +Required CPU/CODEC subnodes properties: + +-link-name : Name of the dai link. +-sound-dai : phandle and port of CPU/CODEC +-capture-dai : phandle and port of CPU/CODEC + +Example: + +sound: sound { + compatible = "qcom,apq8016-sbc-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + qcom,model = "DB410c"; + + /* I2S - Internal codec */ + internal-dai-link@0 { + cpu { /* PRIMARY */ + sound-dai = <&lpass MI2S_PRIMARY>; + }; + codec { + sound-dai = <&wcd_codec 0>; + }; + }; + + /* External Primary or External Secondary -ADV7533 HDMI */ + external-dai-link@0 { + link-name = "ADV7533"; + cpu { /* QUAT */ + sound-dai = <&lpass MI2S_QUATERNARY>; + }; + codec { + sound-dai = <&adv_bridge 0>; + }; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt index e00732dac939..21c648328be9 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt @@ -4,12 +4,21 @@ This node models the Qualcomm Technologies Low-Power Audio SubSystem (LPASS). Required properties: -- compatible : "qcom,lpass-cpu" +- compatible : "qcom,lpass-cpu" or "qcom,apq8016-lpass-cpu" - clocks : Must contain an entry for each entry in clock-names. - clock-names : A list which must include the following entries: * "ahbix-clk" * "mi2s-osr-clk" * "mi2s-bit-clk" + : required clocks for "qcom,lpass-cpu-apq8016" + * "ahbix-clk" + * "mi2s-bit-clk0" + * "mi2s-bit-clk1" + * "mi2s-bit-clk2" + * "mi2s-bit-clk3" + * "pcnoc-mport-clk" + * "pcnoc-sway-clk" + - interrupts : Must contain an entry for each entry in interrupt-names. - interrupt-names : A list which must include the following entries: @@ -22,6 +31,8 @@ Required properties: - reg-names : A list which must include the following entries: * "lpass-lpaif" + + Optional properties: - qcom,adsp : Phandle for the audio DSP node diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index f316ce1f214a..b6b3a786855f 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -5,6 +5,7 @@ Required properties: "renesas,rcar_sound-gen1" if generation1, and "renesas,rcar_sound-gen2" if generation2 Examples with soctypes are: + - "renesas,rcar_sound-r8a7778" (R-Car M1A) - "renesas,rcar_sound-r8a7790" (R-Car H2) - "renesas,rcar_sound-r8a7791" (R-Car M2-W) - reg : Should contain the register physical address. @@ -47,7 +48,7 @@ DAI subnode properties: Example: -rcar_sound: rcar_sound@ec500000 { +rcar_sound: sound@ec500000 { #sound-dai-cells = <1>; compatible = "renesas,rcar_sound-r8a7791", "renesas,rcar_sound-gen2"; reg = <0 0xec500000 0 0x1000>, /* SCU */ diff --git a/Documentation/devicetree/bindings/sound/rt5645.txt b/Documentation/devicetree/bindings/sound/rt5645.txt new file mode 100644 index 000000000000..7cee1f518f59 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5645.txt @@ -0,0 +1,72 @@ +RT5650/RT5645 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : One of "realtek,rt5645" or "realtek,rt5650". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +- hp-detect-gpios: + a GPIO spec for the external headphone detect pin. If jd-mode = 0, + we will get the JD status by getting the value of hp-detect-gpios. + +- realtek,in2-differential + Boolean. Indicate MIC2 input are differential, rather than single-ended. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using IN2P pin as dmic1 data pin + 2: using GPIO6 pin as dmic1 data pin + 3: using GPIO10 pin as dmic1 data pin + 4: using GPIO12 pin as dmic1 data pin + +- realtek,dmic2-data-pin + 0: dmic2 is not used + 1: using IN2N pin as dmic2 data pin + 2: using GPIO5 pin as dmic2 data pin + 3: using GPIO11 pin as dmic2 data pin + +-- realtek,jd-mode : The JD mode of rt5645/rt5650 + 0 : rt5645/rt5650 JD function is not used + 1 : Mode-0 (VDD=3.3V), two port jack detection + 2 : Mode-1 (VDD=3.3V), one port jack detection + 3 : Mode-2 (VDD=1.8V), one port jack detection + +Pins on the device (for linking into audio routes) for RT5645/RT5650: + + * DMIC L1 + * DMIC R1 + * DMIC L2 + * DMIC R2 + * IN1P + * IN1N + * IN2P + * IN2N + * Haptic Generator + * HPOL + * HPOR + * LOUTL + * LOUTR + * PDM1L + * PDM1R + * SPOL + * SPOR + +Example: + +codec: rt5650@1a { + compatible = "realtek,rt5650"; + reg = <0x1a>; + hp-detect-gpios = <&gpio 19 0>; + interrupt-parent = <&gpio>; + interrupts = <7 IRQ_TYPE_EDGE_FALLING>; + realtek,dmic-en = "true"; + realtek,en-jd-func = "true"; + realtek,jd-mode = <3>; +};
\ No newline at end of file diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index 740ff771aa8b..f07078997f87 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -18,6 +18,7 @@ Required properties: Optional properties: - realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin. +- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin. - realtek,in1-differential - realtek,in2-differential @@ -70,6 +71,7 @@ rt5677 { realtek,pow-ldo2-gpio = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; + realtek,reset-gpio = <&gpio TEGRA_GPIO(BB, 3) GPIO_ACTIVE_LOW>; realtek,in1-differential = "true"; realtek,gpio-config = /bits/ 8 <0 0 0 0 0 2>; /* pull up GPIO6 */ realtek,jd2-gpio = <3>; /* Enables Jack detection for GPIO6 */ diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index 73bf314f7240..cf3979eb3578 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -16,7 +16,8 @@ Optional properties: connection's sink, the second being the connection's source. - simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec - mclk. + mclk. When defined, mclk-fs property defined in + dai-link sub nodes are ignored. - simple-audio-card,hp-det-gpio : Reference to GPIO that signals when headphones are attached. - simple-audio-card,mic-det-gpio : Reference to GPIO that signals when @@ -55,6 +56,9 @@ Optional dai-link subnode properties: dai-link uses bit clock inversion. - frame-inversion : bool property. Add this if the dai-link uses frame clock inversion. +- mclk-fs : Multiplication factor between stream + rate and codec mclk, applied only for + the dai-link. For backward compatibility the frame-master and bitclock-master properties can be used as booleans in codec subnode to indicate if the diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt index 55e2a0af5645..c49992c0b62a 100644 --- a/Documentation/devicetree/bindings/sound/tas2552.txt +++ b/Documentation/devicetree/bindings/sound/tas2552.txt @@ -14,6 +14,12 @@ Required properties: Optional properties: - enable-gpio - gpio pin to enable/disable the device +tas2552 can receive it's reference clock via MCLK, BCLK, IVCLKIN pin or use the +internal 1.8MHz. This CLKIN is used by the PLL. In addition to PLL, the PDM +reference clock is also selectable: PLL, IVCLKIN, BCLK or MCLK. +For system integration the dt-bindings/sound/tas2552.h header file provides +defined values to selct and configure the PLL and PDM reference clocks. + Example: tas2552: tas2552@41 { diff --git a/Documentation/devicetree/bindings/sound/tas571x.txt b/Documentation/devicetree/bindings/sound/tas571x.txt new file mode 100644 index 000000000000..0ac31d8d5ac4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas571x.txt @@ -0,0 +1,41 @@ +Texas Instruments TAS5711/TAS5717/TAS5719 stereo power amplifiers + +The codec is controlled through an I2C interface. It also has two other +signals that can be wired up to GPIOs: reset (strongly recommended), and +powerdown (optional). + +Required properties: + +- compatible: "ti,tas5711", "ti,tas5717", or "ti,tas5719" +- reg: The I2C address of the device +- #sound-dai-cells: must be equal to 0 + +Optional properties: + +- reset-gpios: GPIO specifier for the TAS571x's active low reset line +- pdn-gpios: GPIO specifier for the TAS571x's active low powerdown line +- clocks: clock phandle for the MCLK input +- clock-names: should be "mclk" +- AVDD-supply: regulator phandle for the AVDD supply (all chips) +- DVDD-supply: regulator phandle for the DVDD supply (all chips) +- HPVDD-supply: regulator phandle for the HPVDD supply (5717/5719) +- PVDD_AB-supply: regulator phandle for the PVDD_AB supply (5717/5719) +- PVDD_CD-supply: regulator phandle for the PVDD_CD supply (5717/5719) +- PVDD_A-supply: regulator phandle for the PVDD_A supply (5711) +- PVDD_B-supply: regulator phandle for the PVDD_B supply (5711) +- PVDD_C-supply: regulator phandle for the PVDD_C supply (5711) +- PVDD_D-supply: regulator phandle for the PVDD_D supply (5711) + +Example: + + tas5717: audio-codec@2a { + compatible = "ti,tas5717"; + reg = <0x2a>; + #sound-dai-cells = <0>; + + reset-gpios = <&gpio5 1 GPIO_ACTIVE_LOW>; + pdn-gpios = <&gpio5 2 GPIO_ACTIVE_LOW>; + + clocks = <&clk_core CLK_I2S>; + clock-names = "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8741.txt b/Documentation/devicetree/bindings/sound/wm8741.txt index 74bda58c1bcf..a13315408719 100644 --- a/Documentation/devicetree/bindings/sound/wm8741.txt +++ b/Documentation/devicetree/bindings/sound/wm8741.txt @@ -10,9 +10,20 @@ Required properties: - reg : the I2C address of the device for I2C, the chip select number for SPI. +Optional properties: + + - diff-mode: Differential output mode configuration. Default value for field + DIFF in register R8 (MODE_CONTROL_2). If absent, the default is 0, shall be: + 0 = stereo + 1 = mono left + 2 = stereo reversed + 3 = mono right + Example: codec: wm8741@1a { compatible = "wlf,wm8741"; reg = <0x1a>; + + diff-mode = <3>; }; diff --git a/Documentation/devicetree/bindings/sound/zte,zx-i2s.txt b/Documentation/devicetree/bindings/sound/zte,zx-i2s.txt new file mode 100644 index 000000000000..7e5aa6f6b5a1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/zte,zx-i2s.txt @@ -0,0 +1,44 @@ +ZTE ZX296702 I2S controller + +Required properties: + - compatible : Must be "zte,zx296702-i2s" + - reg : Must contain I2S core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + - clock-names: "tx" for the clock to the I2S interface. + - dmas: Pairs of phandle and specifier for the DMA channel that is used by + the core. The core expects two dma channels for transmit. + - dma-names : Must be "tx" and "rx" + +For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties +please check: + * resource-names.txt + * clock/clock-bindings.txt + * dma/dma.txt + +Example: + i2s0: i2s0@0b005000 { + #sound-dai-cells = <0>; + compatible = "zte,zx296702-i2s"; + reg = <0x0b005000 0x1000>; + clocks = <&lsp0clk ZX296702_I2S0_DIV>; + clock-names = "tx"; + interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma 5>, <&dma 6>; + dma-names = "tx", "rx"; + status = "okay"; + }; + + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "zx296702_snd"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + sndcpu: simple-audio-card,cpu { + sound-dai = <&i2s0>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&acodec>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/zte,zx-spdif.txt b/Documentation/devicetree/bindings/sound/zte,zx-spdif.txt new file mode 100644 index 000000000000..989544ea6eb5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/zte,zx-spdif.txt @@ -0,0 +1,28 @@ +ZTE ZX296702 SPDIF controller + +Required properties: + - compatible : Must be "zte,zx296702-spdif" + - reg : Must contain SPDIF core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + - clock-names: "tx" for the clock to the SPDIF interface. + - dmas: Pairs of phandle and specifier for the DMA channel that is used by + the core. The core expects one dma channel for transmit. + - dma-names : Must be "tx" + +For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties +please check: + * resource-names.txt + * clock/clock-bindings.txt + * dma/dma.txt + +Example: + spdif0: spdif0@0b004000 { + compatible = "zte,zx296702-spdif"; + reg = <0x0b004000 0x1000>; + clocks = <&lsp0clk ZX296702_SPDIF0_DIV>; + clock-names = "tx"; + interrupts = <GIC_SPI 21 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma 4>; + dma-names = "tx"; + status = "okay"; + }; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index 53d87bad0adc..7d91d12acdff 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -54,6 +54,7 @@ cosmic Cosmic Circuits crystalfontz Crystalfontz America, Inc. dallas Maxim Integrated Products (formerly Dallas Semiconductor) davicom DAVICOM Semiconductor, Inc. +delta Delta Electronics, Inc. denx Denx Software Engineering digi Digi International Inc. digilent Diglent, Inc. diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 5a3163cac6c3..ec099d4343f2 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -11,7 +11,10 @@ ALC880 ALC260 ====== - N/A + gpio1 Enable GPIO1 + coef Enable EAPD via COEF table + fujitsu Quirk for FSC S7020 + fujitsu-jwse Quirk for FSC S7020 with jack modes and HP mic support ALC262 ====== @@ -20,8 +23,9 @@ ALC262 ALC267/268 ========== inv-dmic Inverted internal mic workaround + hp-eapd Disable HP EAPD on NID 0x15 -ALC269/270/275/276/28x/29x +ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models) ====== laptop-amic Laptops with analog-mic input laptop-dmic Laptops with digital-mic input @@ -29,9 +33,15 @@ ALC269/270/275/276/28x/29x alc271-dmic Enable ALC271X digital mic workaround inv-dmic Inverted internal mic workaround headset-mic Indicates a combined headset (headphone+mic) jack + headset-mode More comprehensive headset support for ALC269 & co + headset-mode-no-hp-mic Headset mode support without headphone mic lenovo-dock Enables docking station I/O for some Lenovos + hp-gpio-led GPIO LED support on HP laptops dell-headset-multi Headset jack, which can also be used as mic-in dell-headset-dock Headset jack (without mic-in), and also dock I/O + alc283-dac-wcaps Fixups for Chromebook with ALC283 + alc283-sense-combo Combo jack sensing on ALC283 + tpt440-dock Pin configs for Lenovo Thinkpad Dock support ALC66x/67x/892 ============== diff --git a/Documentation/sound/alsa/Jack-Controls.txt b/Documentation/sound/alsa/Jack-Controls.txt new file mode 100644 index 000000000000..fe1c5e0c8555 --- /dev/null +++ b/Documentation/sound/alsa/Jack-Controls.txt @@ -0,0 +1,43 @@ +Why we need Jack kcontrols +========================== + +ALSA uses kcontrols to export audio controls(switch, volume, Mux, ...) +to user space. This means userspace applications like pulseaudio can +switch off headphones and switch on speakers when no headphones are +pluged in. + +The old ALSA jack code only created input devices for each registered +jack. These jack input devices are not readable by userspace devices +that run as non root. + +The new jack code creates embedded jack kcontrols for each jack that +can be read by any process. + +This can be combined with UCM to allow userspace to route audio more +intelligently based on jack insertion or removal events. + +Jack Kcontrol Internals +======================= + +Each jack will have a kcontrol list, so that we can create a kcontrol +and attach it to the jack, at jack creation stage. We can also add a +kcontrol to an existing jack, at anytime when required. + +Those kcontrols will be freed automatically when the Jack is freed. + +How to use jack kcontrols +========================= + +In order to keep compatibility, snd_jack_new() has been modified by +adding two params :- + + - @initial_kctl: if true, create a kcontrol and add it to the jack + list. + - @phantom_jack: Don't create a input device for phantom jacks. + +HDA jacks can set phantom_jack to true in order to create a phantom +jack and set initial_kctl to true to create an initial kcontrol with +the correct id. + +ASoC jacks should set initial_kctl as false. The pin name will be +assigned as the jack kcontrol name. diff --git a/Documentation/sound/oss/PSS-updates b/Documentation/sound/oss/PSS-updates index c84dd7597e64..11914a1dc7e7 100644 --- a/Documentation/sound/oss/PSS-updates +++ b/Documentation/sound/oss/PSS-updates @@ -41,7 +41,7 @@ pss_no_sound This module parameter is a flag that can be used to tell the driver to just configure non-sound components. 0 configures all components, a non-0 -value will only attept to configure the CDROM and joystick ports. This +value will only attempt to configure the CDROM and joystick ports. This parameter can be used by a user who only wished to use the builtin joystick and/or CDROM port(s) of his PSS sound card. If this driver is loaded with this parameter and with the parameter below set to true then a user can safely unload diff --git a/Documentation/sound/oss/README.OSS b/Documentation/sound/oss/README.OSS index 4be259428a1c..a085ea3611a1 100644 --- a/Documentation/sound/oss/README.OSS +++ b/Documentation/sound/oss/README.OSS @@ -1346,7 +1346,7 @@ implement nice real-time signal processing audio effect software and network telephones. The ACI mixer has to be switched into the "solo" mode for duplex operation in order to avoid feedback caused by the mixer (input hears output signal). You can de-/activate this mode -through toggleing the record button for the wave controller with an +through toggling the record button for the wave controller with an OSS-mixer. The PCM20 contains a radio tuner, which is also controlled by diff --git a/Documentation/sound/oss/btaudio b/Documentation/sound/oss/btaudio index 1a693e69d44b..effdb9a3f898 100644 --- a/Documentation/sound/oss/btaudio +++ b/Documentation/sound/oss/btaudio @@ -29,7 +29,7 @@ Driver Status Still somewhat experimental. The driver should work stable, i.e. it should'nt crash your box. It might not work as expected, have bugs, -not being fully OSS API compilant, ... +not being fully OSS API compliant, ... Latest versions are available from http://bytesex.org/bttv/, the driver is in the bttv tarball. Kernel patches might be available too, |