diff options
Diffstat (limited to 'Documentation')
15 files changed, 453 insertions, 57 deletions
diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl index fb32aead5a0b..bd6fee22c4dd 100644 --- a/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl @@ -871,9 +871,8 @@ <para> This function itself doesn't allocate the data space. The data must be allocated manually beforehand, and its pointer is passed - as the argument. This pointer is used as the - (<parameter>chip</parameter> identifier in the above example) - for the instance. + as the argument. This pointer (<parameter>chip</parameter> in the + above example) is used as the identifier for the instance. </para> <para> @@ -2304,7 +2303,7 @@ struct _snd_pcm_runtime { <constant>SNDRV_PCM_INFO_XXX</constant>. Here, at least, you have to specify whether the mmap is supported and which interleaved format is supported. - When the is supported, add the + When the hardware supports mmap, add the <constant>SNDRV_PCM_INFO_MMAP</constant> flag here. When the hardware supports the interleaved or the non-interleaved formats, <constant>SNDRV_PCM_INFO_INTERLEAVED</constant> or @@ -2898,7 +2897,7 @@ struct _snd_pcm_runtime { <para> When the pcm supports the pause operation (given in the info - field of the hardware table), the <constant>PAUSE_PUSE</constant> + field of the hardware table), the <constant>PAUSE_PUSH</constant> and <constant>PAUSE_RELEASE</constant> commands must be handled here, too. The former is the command to pause the pcm, and the latter to restart the pcm again. @@ -3085,7 +3084,7 @@ struct _snd_pcm_runtime { <section id="pcm-interface-interrupt-handler-timer"> <title>High frequency timer interrupts</title> <para> - This happense when the hardware doesn't generate interrupts + This happens when the hardware doesn't generate interrupts at the period boundary but issues timer interrupts at a fixed timer rate (e.g. es1968 or ymfpci drivers). In this case, you need to check the current hardware @@ -3251,18 +3250,19 @@ struct _snd_pcm_runtime { <title>Example of Hardware Constraints for Channels</title> <programlisting> <![CDATA[ - static int hw_rule_format_by_channels(struct snd_pcm_hw_params *params, + static int hw_rule_channels_by_format(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval *c = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); + SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_mask fmt; + struct snd_interval ch; - snd_mask_any(&fmt); /* Init the struct */ - if (c->min < 2) { - fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE; - return snd_mask_refine(f, &fmt); + snd_interval_any(&ch); + if (f->bits[0] == SNDRV_PCM_FMTBIT_S16_LE) { + ch.min = ch.max = 1; + ch.integer = 1; + return snd_interval_refine(c, &ch); } return 0; } @@ -3278,35 +3278,35 @@ struct _snd_pcm_runtime { <programlisting> <![CDATA[ snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_FORMAT, - -1); + hw_rule_channels_by_format, NULL, + SNDRV_PCM_HW_PARAM_FORMAT, -1); ]]> </programlisting> </informalexample> </para> <para> - The rule function is called when an application sets the number of - channels. But an application can set the format before the number of - channels. Thus you also need to define the inverse rule: + The rule function is called when an application sets the PCM + format, and it refines the number of channels accordingly. + But an application may set the number of channels before + setting the format. Thus you also need to define the inverse rule: <example> - <title>Example of Hardware Constraints for Channels</title> + <title>Example of Hardware Constraints for Formats</title> <programlisting> <![CDATA[ - static int hw_rule_channels_by_format(struct snd_pcm_hw_params *params, + static int hw_rule_format_by_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval *c = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); + SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_interval ch; + struct snd_mask fmt; - snd_interval_any(&ch); - if (f->bits[0] == SNDRV_PCM_FMTBIT_S16_LE) { - ch.min = ch.max = 1; - ch.integer = 1; - return snd_interval_refine(c, &ch); + snd_mask_any(&fmt); /* Init the struct */ + if (c->min < 2) { + fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE; + return snd_mask_refine(f, &fmt); } return 0; } @@ -3321,8 +3321,8 @@ struct _snd_pcm_runtime { <programlisting> <![CDATA[ snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - hw_rule_format_by_channels, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - -1); + hw_rule_format_by_channels, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); ]]> </programlisting> </informalexample> diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt new file mode 100644 index 000000000000..623d4e70ae11 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4642.txt @@ -0,0 +1,17 @@ +AK4642 I2C transmitter + +This device supports I2C mode only. + +Required properties: + + - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648" + - reg : The chip select number on the I2C bus + +Example: + +&i2c { + ak4648: ak4648@0x12 { + compatible = "asahi-kasei,ak4642"; + reg = <0x12>; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt index a850fb9c88ea..e2cd1d7539e5 100644 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -20,6 +20,18 @@ Optional properties: !RESET pin - cirrus,amuteb-eq-bmutec: When given, the Codec's AMUTEB=BMUTEC flag is enabled. + - cirrus,enable-soft-reset: + The CS4271 requires its LRCLK and MCLK to be stable before its RESET + line is de-asserted. That also means that clocks cannot be changed + without putting the chip back into hardware reset, which also requires + a complete re-initialization of all registers. + + One (undocumented) workaround is to assert and de-assert the PDN bit + in the MODE2 register. This workaround can be enabled with this DT + property. + + Note that this is not needed in case the clocks are stable + throughout the entire runtime of the codec. Examples: diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt new file mode 100644 index 000000000000..be35d34e8b26 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt @@ -0,0 +1,51 @@ +NVIDIA Tegra audio complex + +Required properties: +- compatible : "nvidia,tegra-audio-wm9712" +- nvidia,model : The user-visible name of this sound complex. +- nvidia,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 WM9712's pins, and the jacks on the board: + + WM9712 pins: + + * MONOOUT + * HPOUTL + * HPOUTR + * LOUT2 + * ROUT2 + * OUT3 + * LINEINL + * LINEINR + * PHONE + * PCBEEP + * MIC1 + * MIC2 + * Mic Bias + + Board connectors: + + * Headphone + * LineIn + * Mic + +- nvidia,ac97-controller : The phandle of the Tegra AC97 controller + + +Example: + +sound { + compatible = "nvidia,tegra-audio-wm9712-colibri_t20", + "nvidia,tegra-audio-wm9712"; + nvidia,model = "Toradex Colibri T20"; + + nvidia,audio-routing = + "Headphone", "HPOUTL", + "Headphone", "HPOUTR", + "LineIn", "LINEINL", + "LineIn", "LINEINR", + "Mic", "MIC1"; + + nvidia,ac97-controller = <&ac97>; +}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt new file mode 100644 index 000000000000..c1454979c1ef --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt @@ -0,0 +1,22 @@ +NVIDIA Tegra 20 AC97 controller + +Required properties: +- compatible : "nvidia,tegra20-ac97" +- reg : Should contain AC97 controller registers location and length +- interrupts : Should contain AC97 interrupt +- nvidia,dma-request-selector : The Tegra DMA controller's phandle and + request selector for the AC97 controller +- nvidia,codec-reset-gpio : The Tegra GPIO controller's phandle and the number + of the GPIO used to reset the external AC97 codec +- nvidia,codec-sync-gpio : The Tegra GPIO controller's phandle and the number + of the GPIO corresponding with the AC97 DAP _FS line +Example: + +ac97@70002000 { + compatible = "nvidia,tegra20-ac97"; + reg = <0x70002000 0x200>; + interrupts = <0 81 0x04>; + nvidia,dma-request-selector = <&apbdma 12>; + nvidia,codec-reset-gpio = <&gpio 170 0>; + nvidia,codec-sync-gpio = <&gpio 120 0>; +}; diff --git a/Documentation/devicetree/bindings/sound/omap-twl4030.txt b/Documentation/devicetree/bindings/sound/omap-twl4030.txt index 6fae51c7f766..1ab6bc8404d5 100644 --- a/Documentation/devicetree/bindings/sound/omap-twl4030.txt +++ b/Documentation/devicetree/bindings/sound/omap-twl4030.txt @@ -6,6 +6,52 @@ Required properties: - ti,mcbsp: phandle for the McBSP node - ti,codec: phandle for the twl4030 audio node +Optional properties: +- ti,mcbsp-voice: phandle for the McBSP node connected to the voice port of twl +- ti, jack-det-gpio: Jack detect GPIO +- 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. + If the routing is not provided all possible connection will be available + +Available audio endpoints for the audio-routing table: + +Board connectors: + * Headset Stereophone + * Earpiece Spk + * Handsfree Spk + * Ext Spk + * Main Mic + * Sub Mic + * Headset Mic + * Carkit Mic + * Digital0 Mic + * Digital1 Mic + * Line In + +twl4030 pins: + * HSOL + * HSOR + * EARPIECE + * HFL + * HFR + * PREDRIVEL + * PREDRIVER + * CARKITL + * CARKITR + * MAINMIC + * SUBMIC + * HSMIC + * DIGIMIC0 + * DIGIMIC1 + * CARKITMIC + * AUXL + * AUXR + + * Headset Mic Bias + * Mic Bias 1 /* Used for Main Mic or Digimic0 */ + * Mic Bias 2 /* Used for Sub Mic or Digimic1 */ + Example: sound { diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.txt b/Documentation/devicetree/bindings/sound/renesas,fsi.txt new file mode 100644 index 000000000000..c5be003f413e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,fsi.txt @@ -0,0 +1,26 @@ +Renesas FSI + +Required properties: +- compatible : "renesas,sh_fsi2" or "renesas,sh_fsi" +- reg : Should contain the register physical address and length +- interrupts : Should contain FSI interrupt + +- fsia,spdif-connection : FSI is connected by S/PDFI +- fsia,stream-mode-support : FSI supports 16bit stream mode. +- fsia,use-internal-clock : FSI uses internal clock when master mode. + +- fsib,spdif-connection : same as fsia +- fsib,stream-mode-support : same as fsia +- fsib,use-internal-clock : same as fsia + +Example: + +sh_fsi2: sh_fsi2@0xec230000 { + compatible = "renesas,sh_fsi2"; + reg = <0xec230000 0x400>; + interrupts = <0 146 0x4>; + + fsia,spdif-connection; + fsia,stream-mode-support; + fsia,use-internal-clock; +}; diff --git a/Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt b/Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt new file mode 100644 index 000000000000..4686646fb122 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt @@ -0,0 +1,14 @@ +Samsung SMDK audio complex + +Required properties: +- compatible : "samsung,smdk-wm8994" +- samsung,i2s-controller: The phandle of the Samsung I2S0 controller +- samsung,audio-codec: The phandle of the WM8994 audio codec +Example: + +sound { + compatible = "samsung,smdk-wm8994"; + + samsung,i2s-controller = <&i2s0>; + samsung,audio-codec = <&wm8994>; +}; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt new file mode 100644 index 000000000000..3070046da2e5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt @@ -0,0 +1,63 @@ +* Samsung I2S controller + +Required SoC Specific Properties: + +- compatible : "samsung,i2s-v5" +- reg: physical base address of the controller and length of memory mapped + region. +- 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. + +Optional SoC Specific Properties: + +- samsung,supports-6ch: If the I2S Primary sound source has 5.1 Channel + support, this flag is enabled. +- samsung,supports-rstclr: This flag should be set if I2S software reset bit + control is required. When this flag is set I2S software reset bit will be + enabled or disabled based on need. +- samsung,supports-secdai:If I2S block has a secondary FIFO and internal DMA, + then this flag is enabled. +- samsung,idma-addr: Internal DMA register base address of the audio + sub system(used in secondary sound source). + +Required Board Specific Properties: + +- gpios: The gpio specifier for data out,data in, LRCLK, CDCLK and SCLK + interface lines. The format of the gpio specifier depends on the gpio + controller. + The syntax of samsung gpio specifier is + <[phandle of the gpio controller node] + [pin number within the gpio controller] + [mux function] + [flags and pull up/down] + [drive strength]> + +Example: + +- SoC Specific Portion: + +i2s@03830000 { + compatible = "samsung,i2s-v5"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10 + &pdma0 9 + &pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + samsung,supports-6ch; + samsung,supports-rstclr; + samsung,supports-secdai; + samsung,idma-addr = <0x03000000>; +}; + +- Board Specific Portion: + +i2s@03830000 { + gpios = <&gpz 0 2 0 0>, /* I2S_0_SCLK */ + <&gpz 1 2 0 0>, /* I2S_0_CDCLK */ + <&gpz 2 2 0 0>, /* I2S_0_LRCK */ + <&gpz 3 2 0 0>, /* I2S_0_SDI */ + <&gpz 4 2 0 0>, /* I2S_0_SDO[1] */ + <&gpz 5 2 0 0>, /* I2S_0_SDO[2] */ + <&gpz 6 2 0 0>; /* I2S_0_SDO[3] */ +}; diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index e7b98f41fa5f..f47c3f589fd0 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -11,6 +11,12 @@ Optional properties: - gpio-reset - gpio pin number used for codec reset - ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality +- ai3x-micbias-vg - MicBias Voltage required. + 1 - MICBIAS output is powered to 2.0V, + 2 - MICBIAS output is powered to 2.5V, + 3 - MICBIAS output is connected to AVDD, + If this node is not mentioned or if the value is incorrect, then MicBias + is powered down. Example: diff --git a/Documentation/devicetree/bindings/sound/wm8962.txt b/Documentation/devicetree/bindings/sound/wm8962.txt new file mode 100644 index 000000000000..dceb3b1c2bb7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8962.txt @@ -0,0 +1,16 @@ +WM8962 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "wlf,wm8962" + + - reg : the I2C address of the device. + +Example: + +codec: wm8962@1a { + compatible = "wlf,wm8962"; + reg = <0x1a>; +}; diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index b9cfd339a6fa..ce6581c8ca26 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -890,8 +890,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. enable_msi - Enable Message Signaled Interrupt (MSI) (default = off) power_save - Automatic power-saving timeout (in second, 0 = disable) - power_save_controller - Reset HD-audio controller in power-saving mode - (default = on) + power_save_controller - Support runtime D3 of HD-audio controller + (-1 = on for supported chip (default), false = off, + true = force to on even for unsupported hardware) align_buffer_size - Force rounding of buffer/period sizes to multiples of 128 bytes. This is more efficient in terms of memory access but isn't required by the HDA spec and prevents diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 16dfe57f1731..bb8b0dc532b8 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -53,7 +53,7 @@ ALC882/883/885/888/889 acer-aspire-8930g Acer Aspire 8330G/6935G acer-aspire Acer Aspire others inv-dmic Inverted internal mic workaround - no-primary-hp VAIO Z workaround (for fixed speaker DAC) + no-primary-hp VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC) ALC861/660 ========== diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 7813c06a5c71..d4faa63ff352 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -176,14 +176,14 @@ support the automatic probing (yet as of 2.6.28). And, BIOS is often, yes, pretty often broken. It sets up wrong values and screws up the driver. -The preset model is provided basically to overcome such a situation. -When the matching preset model is found in the white-list, the driver -assumes the static configuration of that preset and builds the mixer -elements and PCM streams based on the static information. Thus, if -you have a newer machine with a slightly different PCI SSID from the -existing one, you may have a good chance to re-use the same model. -You can pass the `model` option to specify the preset model instead of -PCI SSID look-up. +The preset model (or recently called as "fix-up") is provided +basically to overcome such a situation. When the matching preset +model is found in the white-list, the driver assumes the static +configuration of that preset with the correct pin setup, etc. +Thus, if you have a newer machine with a slightly different PCI SSID +(or codec SSID) from the existing one, you may have a good chance to +re-use the same model. You can pass the `model` option to specify the +preset model instead of PCI (and codec-) SSID look-up. What `model` option values are available depends on the codec chip. Check your codec chip from the codec proc file (see "Codec Proc-File" @@ -199,17 +199,12 @@ non-working HD-audio hardware is to check HD-audio codec and several different `model` option values. If you have any luck, some of them might suit with your device well. -Some codecs such as ALC880 have a special model option `model=test`. -This configures the driver to provide as many mixer controls as -possible for every single pin feature except for the unsolicited -events (and maybe some other specials). Adjust each mixer element and -try the I/O in the way of trial-and-error until figuring out the whole -I/O pin mappings. +There are a few special model option values: +- when 'nofixup' is passed, the device-specific fixups in the codec + parser are skipped. +- when `generic` is passed, the codec-specific parser is skipped and + only the generic parser is used. -Note that `model=generic` has a special meaning. It means to use the -generic parser regardless of the codec. Usually the codec-specific -parser is much better than the generic parser (as now). Thus this -option is more about the debugging purpose. Speaker and Headphone Output ~~~~~~~~~~~~~~~~~~~~~~~~~~~~ @@ -387,9 +382,8 @@ init_verbs:: (separated with a space). hints:: Shows / stores hint strings for codec parsers for any use. - Its format is `key = value`. For example, passing `hp_detect = yes` - to IDT/STAC codec parser will result in the disablement of the - headphone detection. + Its format is `key = value`. For example, passing `jack_detect = no` + will disable the jack detection of the machine completely. init_pin_configs:: Shows the initial pin default config values set by BIOS. driver_pin_configs:: @@ -421,6 +415,61 @@ re-configure based on that state, run like below: ------------------------------------------------------------------------ +Hint Strings +~~~~~~~~~~~~ +The codec parser have several switches and adjustment knobs for +matching better with the actual codec or device behavior. Many of +them can be adjusted dynamically via "hints" strings as mentioned in +the section above. For example, by passing `jack_detect = no` string +via sysfs or a patch file, you can disable the jack detection, thus +the codec parser will skip the features like auto-mute or mic +auto-switch. As a boolean value, either `yes`, `no`, `true`, `false`, +`1` or `0` can be passed. + +The generic parser supports the following hints: + +- jack_detect (bool): specify whether the jack detection is available + at all on this machine; default true +- inv_jack_detect (bool): indicates that the jack detection logic is + inverted +- trigger_sense (bool): indicates that the jack detection needs the + explicit call of AC_VERB_SET_PIN_SENSE verb +- inv_eapd (bool): indicates that the EAPD is implemented in the + inverted logic +- pcm_format_first (bool): sets the PCM format before the stream tag + and channel ID +- sticky_stream (bool): keep the PCM format, stream tag and ID as long + as possible; default true +- spdif_status_reset (bool): reset the SPDIF status bits at each time + the SPDIF stream is set up +- pin_amp_workaround (bool): the output pin may have multiple amp + values +- single_adc_amp (bool): ADCs can have only single input amps +- auto_mute (bool): enable/disable the headphone auto-mute feature; + default true +- auto_mic (bool): enable/disable the mic auto-switch feature; default + true +- line_in_auto_switch (bool): enable/disable the line-in auto-switch + feature; default false +- need_dac_fix (bool): limits the DACs depending on the channel count +- primary_hp (bool): probe headphone jacks as the primary outputs; + default true +- multi_cap_vol (bool): provide multiple capture volumes +- inv_dmic_split (bool): provide split internal mic volume/switch for + phase-inverted digital mics +- indep_hp (bool): provide the independent headphone PCM stream and + the corresponding mixer control, if available +- add_stereo_mix_input (bool): add the stereo mix (analog-loopback + mix) to the input mux if available +- add_out_jack_modes (bool): add "xxx Jack Mode" enum controls to each + output jack for allowing to change the headphone amp capability +- add_in_jack_modes (bool): add "xxx Jack Mode" enum controls to each + input jack for allowing to change the mic bias vref +- power_down_unused (bool): power down the unused widgets +- mixer_nid (int): specifies the widget NID of the analog-loopback + mixer + + Early Patching ~~~~~~~~~~~~~~ When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a @@ -445,7 +494,7 @@ A patch file is a plain text file which looks like below: 0x20 0x400 0xff [hint] - hp_detect = yes + jack_detect = no ------------------------------------------------------------------------ The file needs to have a line `[codec]`. The next line should contain @@ -531,6 +580,13 @@ cable is unplugged. Thus, if you hear noises, suspect first the power-saving. See /sys/module/snd_hda_intel/parameters/power_save to check the current value. If it's non-zero, the feature is turned on. +The recent kernel supports the runtime PM for the HD-audio controller +chip, too. It means that the HD-audio controller is also powered up / +down dynamically. The feature is enabled only for certain controller +chips like Intel LynxPoint. You can enable/disable this feature +forcibly by setting `power_save_controller` option, which is also +available at /sys/module/snd_hda_intel/parameters directory. + Tracepoints ~~~~~~~~~~~ @@ -587,8 +643,9 @@ The latest development codes for HD-audio are found on sound git tree: - git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git The master branch or for-next branches can be used as the main -development branches in general while the HD-audio specific patches -are committed in topic/hda branch. +development branches in general while the development for the current +and next kernels are found in for-linus and for-next branches, +respectively. If you are using the latest Linus tree, it'd be better to pull the above GIT tree onto it. If you are using the older kernels, an easy @@ -699,7 +756,11 @@ won't be always updated. For example, the volume values are usually cached in the driver, and thus changing the widget amp value directly via hda-verb won't change the mixer value. -The hda-verb program is found in the ftp directory: +The hda-verb program is included now in alsa-tools: + +- git://git.alsa-project.org/alsa-tools.git + +Also, the old stand-alone package is found in the ftp directory: - ftp://ftp.suse.com/pub/people/tiwai/misc/ @@ -777,3 +838,18 @@ A git repository is available: See README file in the tarball for more details about hda-emu program. + + +hda-jack-retask +~~~~~~~~~~~~~~~ +hda-jack-retask is a user-friendly GUI program to manipulate the +HD-audio pin control for jack retasking. If you have a problem about +the jack assignment, try this program and check whether you can get +useful results. Once when you figure out the proper pin assignment, +it can be fixed either in the driver code statically or via passing a +firmware patch file (see "Early Patching" section). + +The program is included in alsa-tools now: + +- git://git.alsa-project.org/alsa-tools.git + diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a11abc..0bcc55155911 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,52 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +Also, there might be low-intensity noises due to encoding. Perfect gapless is +difficult to reach with all types of compressed data, but works fine with most +music content. The decoder needs to know the encoder delay and encoder padding. +So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers +and are not present by default in the bitstream, hence the need for a new +interface to pass this information to the DSP. Also DSP and userspace needs to +switch from one track to another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence. This needs to be set before the data in the track +is written. + +- set_next_track +This routine tells DSP that metadata and write operation sent after this would +correspond to subsequent track + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + +Sequence flow for gapless would be: +- Open +- Get caps / codec caps +- Set params +- Set metadata of the first track +- Fill data of the first track +- Trigger start +- User-space finished sending all, +- Indicaite next track data by sending set_next_track +- Set metadata of the next track +- then call partial_drain to flush most of buffer in DSP +- Fill data of the next track +- DSP switches to second track +(note: order for partial_drain and write for next track can be reversed as well) + Not supported: - Support for VoIP/circuit-switched calls is not the target of this |
