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author | Linus Torvalds <torvalds@linux-foundation.org> | 2015-04-15 15:41:41 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2015-04-15 15:41:41 -0700 |
commit | d0a3997c0c3f9351e24029349dee65dd1d9e8d84 (patch) | |
tree | 7a04fe282b0c7b329cd87cdb891f0f3879dc71a6 /Documentation | |
parent | 6d50ff91d9780263160262daeb6adfdda8ddbc6c (diff) | |
parent | d6eb9e3ec78c98324097bab8eea266c3bb0d0ac7 (diff) | |
download | op-kernel-dev-d0a3997c0c3f9351e24029349dee65dd1d9e8d84.zip op-kernel-dev-d0a3997c0c3f9351e24029349dee65dd1d9e8d84.tar.gz |
Merge tag 'sound-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There have been major modernization with the standard bus: in ALSA
sequencer core and HD-audio. Also, HD-audio receives the regmap
support replacing the in-house cache register cache code. These
changes shouldn't impact the existing behavior, but rather
refactoring.
In addition, HD-audio got the code split to a core library part and
the "legacy" driver parts. This is a preliminary work for adapting
the upcoming ASoC HD-audio driver, and the whole transition is still
work in progress, likely finished in 4.1.
Along with them, there are many updates in ASoC area as usual, too:
lots of cleanups, Intel code shuffling, etc.
Here are some highlights:
ALSA core:
- PCM: the audio timestamp / wallclock enhancement
- PCM: fixes in DPCM management
- Fixes / cleanups of user-space control element management
- Sequencer: modernization using the standard bus
HD-audio:
- Modernization using the standard bus
- Regmap support
- Use standard runtime PM for codec power saving
- Widget-path based power-saving for IDT, VIA and Realtek codecs
- Reorganized sysfs entries for each codec object
- More Dell headset support
ASoC:
- Move of jack registration to the card level
- Lots of ASoC cleanups, mainly moving things from the CODEC level to
the card level
- Support for DAPM routes specified by both the machine driver and DT
- Continuing improvements to rcar
- pcm512x enhacements
- Intel platforms updates
- rt5670 updates / fixes
- New platforms / devices: some non-DSP Qualcomm platforms, Google's
Storm platform, Maxmim MAX98925 CODECs and the Ingenic JZ4780 SoC
Misc:
- ice1724: Improved ESI W192M support
- emu10k1: Emu 1010 fixes/enhancement"
* tag 'sound-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (411 commits)
ALSA: hda - set GET bit when adding a vendor verb to the codec regmap
ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T450
ALSA: hda - Fix another race in runtime PM refcounting
ALSA: hda - Expose codec type sysfs
ALSA: ctl: fix to handle several elements added by one operation for userspace element
ASoC: Intel: fix array_size.cocci warnings
ASoC: n810: Automatically disconnect non-connected pins
ASoC: n810: Consistently pass the card DAPM context to n810_ext_control()
ASoC: davinci-evm: Use card DAPM context to access widgets
ASoC: mop500_ab8500: Use card DAPM context to access widgets
ASoC: wm1133-ev1: Use card DAPM context to access widgets
ASoC: atmel: Improve machine driver compile test coverage
ASoC: atmel: Add dependency to SND_SOC_I2C_AND_SPI where necessary
ALSA: control: Fix a typo of SNDRV_CTL_ELEM_ACCESS_TLV_* with SNDRV_CTL_TLV_OP_*
ALSA: usb-audio: Don't attempt to get Microsoft Lifecam Cinema sample rate
ASoC: rnsd: fix build regression without CONFIG_OF
ALSA: emu10k1: add toggles for E-mu 1010 optical ports
ALSA: ctl: fill identical information to return value when adding userspace elements
ALSA: ctl: fix a bug to return no identical information in info operation for userspace controls
ALSA: ctl: confirm to return all identical information in 'activate' event
...
Diffstat (limited to 'Documentation')
11 files changed, 485 insertions, 21 deletions
diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt index b414333..b623d50 100644 --- a/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt +++ b/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt @@ -1,7 +1,7 @@ Ingenic JZ4740 I2S controller Required properties: -- compatible : "ingenic,jz4740-i2s" +- compatible : "ingenic,jz4740-i2s" or "ingenic,jz4780-i2s" - reg : I2S registers location and length - clocks : AIC and I2S PLL clock specifiers. - clock-names: "aic" and "i2s" diff --git a/Documentation/devicetree/bindings/sound/max98925.txt b/Documentation/devicetree/bindings/sound/max98925.txt new file mode 100644 index 0000000..27be63e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98925.txt @@ -0,0 +1,22 @@ +max98925 audio CODEC + +This device supports I2C. + +Required properties: + + - compatible : "maxim,max98925" + + - vmon-slot-no : slot number used to send voltage information + + - imon-slot-no : slot number used to send current information + + - reg : the I2C address of the device for I2C + +Example: + +codec: max98925@1a { + compatible = "maxim,max98925"; + vmon-slot-no = <0>; + imon-slot-no = <2>; + reg = <0x1a>; +}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt index c949abc..c3495be 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt @@ -18,6 +18,7 @@ Required properties: * Headphones * Speakers * Mic Jack + * Int Mic - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's connected to the CODEC. diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt new file mode 100644 index 0000000..e00732d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt @@ -0,0 +1,43 @@ +* Qualcomm Technologies LPASS CPU DAI + +This node models the Qualcomm Technologies Low-Power Audio SubSystem (LPASS). + +Required properties: + +- compatible : "qcom,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" +- interrupts : Must contain an entry for each entry in + interrupt-names. +- interrupt-names : A list which must include the following entries: + * "lpass-irq-lpaif" +- 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: + * "lpass-lpaif" + +Optional properties: + +- qcom,adsp : Phandle for the audio DSP node + +Example: + +lpass@28100000 { + compatible = "qcom,lpass-cpu"; + clocks = <&lcc AHBIX_CLK>, <&lcc MI2S_OSR_CLK>, <&lcc MI2S_BIT_CLK>; + clock-names = "ahbix-clk", "mi2s-osr-clk", "mi2s-bit-clk"; + interrupts = <0 85 1>; + interrupt-names = "lpass-irq-lpaif"; + pinctrl-names = "default", "idle"; + pinctrl-0 = <&mi2s_default>; + pinctrl-1 = <&mi2s_idle>; + reg = <0x28100000 0x10000>; + reg-names = "lpass-lpaif"; + qcom,adsp = <&adsp>; +}; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 2dd690b..f316ce1 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -29,9 +29,17 @@ SSI subnode properties: - shared-pin : if shared clock pin - pio-transfer : use PIO transfer mode - no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case +- dma : Should contain Audio DMAC entry +- dma-names : SSI case "rx" (=playback), "tx" (=capture) + SSIU case "rxu" (=playback), "txu" (=capture) SRC subnode properties: -no properties at this point +- dma : Should contain Audio DMAC entry +- dma-names : "rx" (=playback), "tx" (=capture) + +DVC subnode properties: +- dma : Should contain Audio DMAC entry +- dma-names : "tx" (=playback/capture) DAI subnode properties: - playback : list of playback modules @@ -45,56 +53,145 @@ rcar_sound: rcar_sound@ec500000 { reg = <0 0xec500000 0 0x1000>, /* SCU */ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ - <0 0xec541000 0 0x1280>; /* SSI */ + <0 0xec541000 0 0x1280>, /* SSI */ + <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; + + clocks = <&mstp10_clks R8A7790_CLK_SSI_ALL>, + <&mstp10_clks R8A7790_CLK_SSI9>, <&mstp10_clks R8A7790_CLK_SSI8>, + <&mstp10_clks R8A7790_CLK_SSI7>, <&mstp10_clks R8A7790_CLK_SSI6>, + <&mstp10_clks R8A7790_CLK_SSI5>, <&mstp10_clks R8A7790_CLK_SSI4>, + <&mstp10_clks R8A7790_CLK_SSI3>, <&mstp10_clks R8A7790_CLK_SSI2>, + <&mstp10_clks R8A7790_CLK_SSI1>, <&mstp10_clks R8A7790_CLK_SSI0>, + <&mstp10_clks R8A7790_CLK_SCU_SRC9>, <&mstp10_clks R8A7790_CLK_SCU_SRC8>, + <&mstp10_clks R8A7790_CLK_SCU_SRC7>, <&mstp10_clks R8A7790_CLK_SCU_SRC6>, + <&mstp10_clks R8A7790_CLK_SCU_SRC5>, <&mstp10_clks R8A7790_CLK_SCU_SRC4>, + <&mstp10_clks R8A7790_CLK_SCU_SRC3>, <&mstp10_clks R8A7790_CLK_SCU_SRC2>, + <&mstp10_clks R8A7790_CLK_SCU_SRC1>, <&mstp10_clks R8A7790_CLK_SCU_SRC0>, + <&mstp10_clks R8A7790_CLK_SCU_DVC0>, <&mstp10_clks R8A7790_CLK_SCU_DVC1>, + <&audio_clk_a>, <&audio_clk_b>, <&audio_clk_c>, <&m2_clk>; + clock-names = "ssi-all", + "ssi.9", "ssi.8", "ssi.7", "ssi.6", "ssi.5", + "ssi.4", "ssi.3", "ssi.2", "ssi.1", "ssi.0", + "src.9", "src.8", "src.7", "src.6", "src.5", + "src.4", "src.3", "src.2", "src.1", "src.0", + "dvc.0", "dvc.1", + "clk_a", "clk_b", "clk_c", "clk_i"; rcar_sound,dvc { - dvc0: dvc@0 { }; - dvc1: dvc@1 { }; + dvc0: dvc@0 { + dmas = <&audma0 0xbc>; + dma-names = "tx"; + }; + dvc1: dvc@1 { + dmas = <&audma0 0xbe>; + dma-names = "tx"; + }; }; rcar_sound,src { - src0: src@0 { }; - src1: src@1 { }; - src2: src@2 { }; - src3: src@3 { }; - src4: src@4 { }; - src5: src@5 { }; - src6: src@6 { }; - src7: src@7 { }; - src8: src@8 { }; - src9: src@9 { }; + src0: src@0 { + interrupts = <0 352 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x85>, <&audma1 0x9a>; + dma-names = "rx", "tx"; + }; + src1: src@1 { + interrupts = <0 353 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x87>, <&audma1 0x9c>; + dma-names = "rx", "tx"; + }; + src2: src@2 { + interrupts = <0 354 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x89>, <&audma1 0x9e>; + dma-names = "rx", "tx"; + }; + src3: src@3 { + interrupts = <0 355 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x8b>, <&audma1 0xa0>; + dma-names = "rx", "tx"; + }; + src4: src@4 { + interrupts = <0 356 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x8d>, <&audma1 0xb0>; + dma-names = "rx", "tx"; + }; + src5: src@5 { + interrupts = <0 357 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x8f>, <&audma1 0xb2>; + dma-names = "rx", "tx"; + }; + src6: src@6 { + interrupts = <0 358 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x91>, <&audma1 0xb4>; + dma-names = "rx", "tx"; + }; + src7: src@7 { + interrupts = <0 359 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x93>, <&audma1 0xb6>; + dma-names = "rx", "tx"; + }; + src8: src@8 { + interrupts = <0 360 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x95>, <&audma1 0xb8>; + dma-names = "rx", "tx"; + }; + src9: src@9 { + interrupts = <0 361 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x97>, <&audma1 0xba>; + dma-names = "rx", "tx"; + }; }; rcar_sound,ssi { ssi0: ssi@0 { interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x01>, <&audma1 0x02>, <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi1: ssi@1 { interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x03>, <&audma1 0x04>, <&audma0 0x49>, <&audma1 0x4a>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi2: ssi@2 { interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x05>, <&audma1 0x06>, <&audma0 0x63>, <&audma1 0x64>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi3: ssi@3 { interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x07>, <&audma1 0x08>, <&audma0 0x6f>, <&audma1 0x70>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi4: ssi@4 { interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x09>, <&audma1 0x0a>, <&audma0 0x71>, <&audma1 0x72>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi5: ssi@5 { interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x0b>, <&audma1 0x0c>, <&audma0 0x73>, <&audma1 0x74>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi6: ssi@6 { interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x0d>, <&audma1 0x0e>, <&audma0 0x75>, <&audma1 0x76>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi7: ssi@7 { interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x0f>, <&audma1 0x10>, <&audma0 0x79>, <&audma1 0x7a>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi8: ssi@8 { interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x11>, <&audma1 0x12>, <&audma0 0x7b>, <&audma1 0x7c>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi9: ssi@9 { interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x13>, <&audma1 0x14>, <&audma0 0x7d>, <&audma1 0x7e>; + dma-names = "rx", "tx", "rxu", "txu"; }; }; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt new file mode 100644 index 0000000..c641550 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt @@ -0,0 +1,67 @@ +Renesas Sampling Rate Convert Sound Card: + +Renesas Sampling Rate Convert Sound Card specifies audio DAI connections of SoC <-> codec. + +Required properties: + +- compatible : "renesas,rsrc-card,<board>" + Examples with soctypes are: + - "renesas,rsrc-card,lager" + - "renesas,rsrc-card,koelsch" +Optional properties: + +- card_name : User specified audio sound card name, one string + property. +- cpu : CPU sub-node +- codec : CODEC sub-node + +Optional subnode properties: + +- format : CPU/CODEC common audio format. + "i2s", "right_j", "left_j" , "dsp_a" + "dsp_b", "ac97", "pdm", "msb", "lsb" +- frame-master : Indicates dai-link frame master. + phandle to a cpu or codec subnode. +- bitclock-master : Indicates dai-link bit clock master. + phandle to a cpu or codec subnode. +- bitclock-inversion : bool property. Add this if the + dai-link uses bit clock inversion. +- frame-inversion : bool property. Add this if the + dai-link uses frame clock inversion. +- convert-rate : platform specified sampling rate convert + +Required CPU/CODEC subnodes properties: + +- sound-dai : phandle and port of CPU/CODEC + +Optional CPU/CODEC subnodes properties: + +- clocks / system-clock-frequency : specify subnode's clock if needed. + it can be specified via "clocks" if system has + clock node (= common clock), or "system-clock-frequency" + (if system doens't support common clock) + If a clock is specified, it is + enabled with clk_prepare_enable() + in dai startup() and disabled with + clk_disable_unprepare() in dai + shutdown(). + +Example + +sound { + compatible = "renesas,rsrc-card,lager"; + + card-name = "rsnd-ak4643"; + format = "left_j"; + bitclock-master = <&sndcodec>; + frame-master = <&sndcodec>; + + sndcpu: cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: codec { + sound-dai = <&ak4643>; + system-clock-frequency = <11289600>; + }; +}; diff --git a/Documentation/devicetree/bindings/sound/storm.txt b/Documentation/devicetree/bindings/sound/storm.txt new file mode 100644 index 0000000..062a4c1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/storm.txt @@ -0,0 +1,23 @@ +* Sound complex for Storm boards + +Models a soundcard for Storm boards with the Qualcomm Technologies IPQ806x SOC +connected to a MAX98357A DAC via I2S. + +Required properties: + +- compatible : "google,storm-audio" +- cpu : Phandle of the CPU DAI +- codec : Phandle of the codec DAI + +Optional properties: + +- qcom,model : The user-visible name of this sound card. + +Example: + +sound { + compatible = "google,storm-audio"; + qcom,model = "ipq806x-storm"; + cpu = <&lpass_cpu>; + codec = <&max98357a>; +}; diff --git a/Documentation/devicetree/bindings/sound/wm8804.txt b/Documentation/devicetree/bindings/sound/wm8804.txt index 4d3a56f..6fd124b 100644 --- a/Documentation/devicetree/bindings/sound/wm8804.txt +++ b/Documentation/devicetree/bindings/sound/wm8804.txt @@ -10,6 +10,13 @@ Required properties: - reg : the I2C address of the device for I2C, the chip select number for SPI. + - PVDD-supply, DVDD-supply : Power supplies for the device, as covered + in Documentation/devicetree/bindings/regulator/regulator.txt + +Optional properties: + + - wlf,reset-gpio: A GPIO specifier for the GPIO controlling the reset pin + Example: codec: wm8804@1a { diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt index 79a6127..3fc1cf50 100644 --- a/Documentation/sound/alsa/ControlNames.txt +++ b/Documentation/sound/alsa/ControlNames.txt @@ -71,11 +71,11 @@ SOURCE: HDMI/DP (either HDMI or DisplayPort) Exceptions (deprecated): - [Digital] Capture Source - [Digital] Capture Switch (aka input gain switch) - [Digital] Capture Volume (aka input gain volume) - [Digital] Playback Switch (aka output gain switch) - [Digital] Playback Volume (aka output gain volume) + [Analogue|Digital] Capture Source + [Analogue|Digital] Capture Switch (aka input gain switch) + [Analogue|Digital] Capture Volume (aka input gain volume) + [Analogue|Digital] Playback Switch (aka output gain switch) + [Analogue|Digital] Playback Volume (aka output gain volume) Tone Control - Switch Tone Control - Bass Tone Control - Treble diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 42a0a39..e7193aa 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -466,7 +466,11 @@ The generic parser supports the following hints: - add_jack_modes (bool): add "xxx Jack Mode" enum controls to each I/O jack for allowing to change the headphone amp and mic bias VREF capabilities -- power_down_unused (bool): power down the unused widgets +- power_save_node (bool): advanced power management for each widget, + controlling the power sate (D0/D3) of each widget node depending on + the actual pin and stream states +- power_down_unused (bool): power down the unused widgets, a subset of + power_save_node, and will be dropped in future - add_hp_mic (bool): add the headphone to capture source if possible - hp_mic_detect (bool): enable/disable the hp/mic shared input for a single built-in mic case; default true diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt new file mode 100644 index 0000000..0b191a2 --- /dev/null +++ b/Documentation/sound/alsa/timestamping.txt @@ -0,0 +1,200 @@ +The ALSA API can provide two different system timestamps: + +- Trigger_tstamp is the system time snapshot taken when the .trigger +callback is invoked. This snapshot is taken by the ALSA core in the +general case, but specific hardware may have synchronization +capabilities or conversely may only be able to provide a correct +estimate with a delay. In the latter two cases, the low-level driver +is responsible for updating the trigger_tstamp at the most appropriate +and precise moment. Applications should not rely solely on the first +trigger_tstamp but update their internal calculations if the driver +provides a refined estimate with a delay. + +- tstamp is the current system timestamp updated during the last +event or application query. +The difference (tstamp - trigger_tstamp) defines the elapsed time. + +The ALSA API provides reports two basic pieces of information, avail +and delay, which combined with the trigger and current system +timestamps allow for applications to keep track of the 'fullness' of +the ring buffer and the amount of queued samples. + +The use of these different pointers and time information depends on +the application needs: + +- 'avail' reports how much can be written in the ring buffer +- 'delay' reports the time it will take to hear a new sample after all +queued samples have been played out. + +When timestamps are enabled, the avail/delay information is reported +along with a snapshot of system time. Applications can select from +CLOCK_REALTIME (NTP corrections including going backwards), +CLOCK_MONOTONIC (NTP corrections but never going backwards), +CLOCK_MONOTIC_RAW (without NTP corrections) and change the mode +dynamically with sw_params + + +The ALSA API also provide an audio_tstamp which reflects the passage +of time as measured by different components of audio hardware. In +ascii-art, this could be represented as follows (for the playback +case): + + +--------------------------------------------------------------> time + ^ ^ ^ ^ ^ + | | | | | + analog link dma app FullBuffer + time time time time time + | | | | | + |< codec delay >|<--hw delay-->|<queued samples>|<---avail->| + |<----------------- delay---------------------->| | + |<----ring buffer length---->| + +The analog time is taken at the last stage of the playback, as close +as possible to the actual transducer + +The link time is taken at the output of the SOC/chipset as the samples +are pushed on a link. The link time can be directly measured if +supported in hardware by sample counters or wallclocks (e.g. with +HDAudio 24MHz or PTP clock for networked solutions) or indirectly +estimated (e.g. with the frame counter in USB). + +The DMA time is measured using counters - typically the least reliable +of all measurements due to the bursty natured of DMA transfers. + +The app time corresponds to the time tracked by an application after +writing in the ring buffer. + +The application can query what the hardware supports, define which +audio time it wants reported by selecting the relevant settings in +audio_tstamp_config fields, get an estimate of the timestamp +accuracy. It can also request the delay-to-analog be included in the +measurement. Direct access to the link time is very interesting on +platforms that provide an embedded DSP; measuring directly the link +time with dedicated hardware, possibly synchronized with system time, +removes the need to keep track of internal DSP processing times and +latency. + +In case the application requests an audio tstamp that is not supported +in hardware/low-level driver, the type is overridden as DEFAULT and the +timestamp will report the DMA time based on the hw_pointer value. + +For backwards compatibility with previous implementations that did not +provide timestamp selection, with a zero-valued COMPAT timestamp type +the results will default to the HDAudio wall clock for playback +streams and to the DMA time (hw_ptr) in all other cases. + +The audio timestamp accuracy can be returned to user-space, so that +appropriate decisions are made: + +- for dma time (default), the granularity of the transfers can be + inferred from the steps between updates and in turn provide + information on how much the application pointer can be rewound + safely. + +- the link time can be used to track long-term drifts between audio + and system time using the (tstamp-trigger_tstamp)/audio_tstamp + ratio, the precision helps define how much smoothing/low-pass + filtering is required. The link time can be either reset on startup + or reported as is (the latter being useful to compare progress of + different streams - but may require the wallclock to be always + running and not wrap-around during idle periods). If supported in + hardware, the absolute link time could also be used to define a + precise start time (patches WIP) + +- including the delay in the audio timestamp may + counter-intuitively not increase the precision of timestamps, e.g. if a + codec includes variable-latency DSP processing or a chain of + hardware components the delay is typically not known with precision. + +The accuracy is reported in nanosecond units (using an unsigned 32-bit +word), which gives a max precision of 4.29s, more than enough for +audio applications... + +Due to the varied nature of timestamping needs, even for a single +application, the audio_tstamp_config can be changed dynamically. In +the STATUS ioctl, the parameters are read-only and do not allow for +any application selection. To work around this limitation without +impacting legacy applications, a new STATUS_EXT ioctl is introduced +with read/write parameters. ALSA-lib will be modified to make use of +STATUS_EXT and effectively deprecate STATUS. + +The ALSA API only allows for a single audio timestamp to be reported +at a time. This is a conscious design decision, reading the audio +timestamps from hardware registers or from IPC takes time, the more +timestamps are read the more imprecise the combined measurements +are. To avoid any interpretation issues, a single (system, audio) +timestamp is reported. Applications that need different timestamps +will be required to issue multiple queries and perform an +interpolation of the results + +In some hardware-specific configuration, the system timestamp is +latched by a low-level audio subsytem, and the information provided +back to the driver. Due to potential delays in the communication with +the hardware, there is a risk of misalignment with the avail and delay +information. To make sure applications are not confused, a +driver_timestamp field is added in the snd_pcm_status structure; this +timestamp shows when the information is put together by the driver +before returning from the STATUS and STATUS_EXT ioctl. in most cases +this driver_timestamp will be identical to the regular system tstamp. + +Examples of typestamping with HDaudio: + +1. DMA timestamp, no compensation for DMA+analog delay +$ ./audio_time -p --ts_type=1 +playback: systime: 341121338 nsec, audio time 342000000 nsec, systime delta -878662 +playback: systime: 426236663 nsec, audio time 427187500 nsec, systime delta -950837 +playback: systime: 597080580 nsec, audio time 598000000 nsec, systime delta -919420 +playback: systime: 682059782 nsec, audio time 683020833 nsec, systime delta -961051 +playback: systime: 852896415 nsec, audio time 853854166 nsec, systime delta -957751 +playback: systime: 937903344 nsec, audio time 938854166 nsec, systime delta -950822 + +2. DMA timestamp, compensation for DMA+analog delay +$ ./audio_time -p --ts_type=1 -d +playback: systime: 341053347 nsec, audio time 341062500 nsec, systime delta -9153 +playback: systime: 426072447 nsec, audio time 426062500 nsec, systime delta 9947 +playback: systime: 596899518 nsec, audio time 596895833 nsec, systime delta 3685 +playback: systime: 681915317 nsec, audio time 681916666 nsec, systime delta -1349 +playback: systime: 852741306 nsec, audio time 852750000 nsec, systime delta -8694 + +3. link timestamp, compensation for DMA+analog delay +$ ./audio_time -p --ts_type=2 -d +playback: systime: 341060004 nsec, audio time 341062791 nsec, systime delta -2787 +playback: systime: 426242074 nsec, audio time 426244875 nsec, systime delta -2801 +playback: systime: 597080992 nsec, audio time 597084583 nsec, systime delta -3591 +playback: systime: 682084512 nsec, audio time 682088291 nsec, systime delta -3779 +playback: systime: 852936229 nsec, audio time 852940916 nsec, systime delta -4687 +playback: systime: 938107562 nsec, audio time 938112708 nsec, systime delta -5146 + +Example 1 shows that the timestamp at the DMA level is close to 1ms +ahead of the actual playback time (as a side time this sort of +measurement can help define rewind safeguards). Compensating for the +DMA-link delay in example 2 helps remove the hardware buffering abut +the information is still very jittery, with up to one sample of +error. In example 3 where the timestamps are measured with the link +wallclock, the timestamps show a monotonic behavior and a lower +dispersion. + +Example 3 and 4 are with USB audio class. Example 3 shows a high +offset between audio time and system time due to buffering. Example 4 +shows how compensating for the delay exposes a 1ms accuracy (due to +the use of the frame counter by the driver) + +Example 3: DMA timestamp, no compensation for delay, delta of ~5ms +$ ./audio_time -p -Dhw:1 -t1 +playback: systime: 120174019 nsec, audio time 125000000 nsec, systime delta -4825981 +playback: systime: 245041136 nsec, audio time 250000000 nsec, systime delta -4958864 +playback: systime: 370106088 nsec, audio time 375000000 nsec, systime delta -4893912 +playback: systime: 495040065 nsec, audio time 500000000 nsec, systime delta -4959935 +playback: systime: 620038179 nsec, audio time 625000000 nsec, systime delta -4961821 +playback: systime: 745087741 nsec, audio time 750000000 nsec, systime delta -4912259 +playback: systime: 870037336 nsec, audio time 875000000 nsec, systime delta -4962664 + +Example 4: DMA timestamp, compensation for delay, delay of ~1ms +$ ./audio_time -p -Dhw:1 -t1 -d +playback: systime: 120190520 nsec, audio time 120000000 nsec, systime delta 190520 +playback: systime: 245036740 nsec, audio time 244000000 nsec, systime delta 1036740 +playback: systime: 370034081 nsec, audio time 369000000 nsec, systime delta 1034081 +playback: systime: 495159907 nsec, audio time 494000000 nsec, systime delta 1159907 +playback: systime: 620098824 nsec, audio time 619000000 nsec, systime delta 1098824 +playback: systime: 745031847 nsec, audio time 744000000 nsec, systime delta 1031847 |