diff options
author | Linus Torvalds | 2019-09-17 17:43:33 -0700 |
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committer | Linus Torvalds | 2019-09-17 17:43:33 -0700 |
commit | 6ab8ad31601f29470eb895fd95e5c963e125aa1b (patch) | |
tree | 73327fe9fc2ee62e7815fa0a666fdf46aaab7322 /Documentation | |
parent | ea982ba7f79141d86eb7a440fcba6796ed718b9b (diff) | |
parent | 9bf9bf5440b99edfba496388c90b52ebcd9df715 (diff) |
Merge tag 'sound-5.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"As shown in diffstat and logs, it was again a busy development cycle
at this time, too. The most significant changes are still on-going
refactoring / modernization works for ASoC core and drivers, but there
are lots of other changes as well. Here we go, some highlights below:
ASoC:
- Quite a lot of cleanup / refactoring of ASoC core and APIs; most of
them are systematic, but also including cleanups and modernization
- A bulk of updates for some ASoC platforms, Freescale, sunxi and
Intel SST/SOF
- Initial support for Sound Open Firmware on i.MX8
- Removal of deprecated w90x900 and nuc900 drivers
- New support for Cirrus Logic CS47L15 and CS47L92, Freescale i.MX
7ULP and 8MQ, Meson G12A and NXP UDA1334
USB-audio:
- More validations of descriptor units for hardening against bugs
reported by fuzzers
- PCM device assignment workaround for a past call-order change
- Scarlett Gen2 mixer interface, a few more more quirks
HD-audio:
- Support for audio component with AMD/ATI and Nvidia HDMI codecs
- Clean up HD-audio core and remove indirect access ops for Intel SOF
- DMIC detection at probe; it would make systems automatically
falling back to SST/SOF driver on devices that need DMIC handling.
Needs a new Kconfig to set, and beware that it's still new and a
bit experimental
FireWire:
- Lots of code refactoring and cleanups"
* tag 'sound-5.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (521 commits)
ASoC: sdm845: remove unneeded semicolon
ASoC: fsl_sai: Implement set_bclk_ratio
ASoC: dmaengine: Replace strncpy() with strscpy_pad() for pcm->name
ASoC: wcd9335: remove redundant use of ret variable
ALSA: firewire-tascam: check intermediate state of clock status and retry
ALSA: firewire-tascam: handle error code when getting current source of clock
ASoC: hdmi-codec: Add an op to set callback function for plug event
ASoC: rt5677: keep analog power register at SND_SOC_BIAS_OFF
ASoC: rt5677: Remove magic number register writes
ASoC: soc-core: self contained soc_unbind_aux_dev()
ASoC: soc-core: add soc_unbind_aux_dev()
ASoC: soc-core: self contained soc_bind_aux_dev()
ASoC: soc-core: move soc_probe_link_dais() next to soc_remove_link_dais()
ASoC: soc-core: self contained soc_probe_link_dais()
ASoC: soc-core: add new soc_link_init()
ASoC: soc-core: move soc_probe_dai() next to soc_remove_dai()
ASoC: soc-core: self contained soc_remove_link_dais()
ASoC: soc-core: self contained soc_remove_link_components()
ASoC: soc-core: self contained soc_probe_link_components()
ASoC: rt1308: make array pd static const, makes object smaller
...
Diffstat (limited to 'Documentation')
19 files changed, 274 insertions, 88 deletions
diff --git a/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml b/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml new file mode 100644 index 000000000000..3248595dc93c --- /dev/null +++ b/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml @@ -0,0 +1,88 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/dsp/fsl,dsp.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP i.MX8 DSP core + +maintainers: + - Daniel Baluta <daniel.baluta@nxp.com> + +description: | + Some boards from i.MX8 family contain a DSP core used for + advanced pre- and post- audio processing. + +properties: + compatible: + enum: + - fsl,imx8qxp-dsp + + reg: + description: Should contain register location and length + + clocks: + items: + - description: ipg clock + - description: ocram clock + - description: core clock + + clock-names: + items: + - const: ipg + - const: ocram + - const: core + + power-domains: + description: + List of phandle and PM domain specifier as documented in + Documentation/devicetree/bindings/power/power_domain.txt + maxItems: 4 + + mboxes: + description: + List of <&phandle type channel> - 2 channels for TXDB, 2 channels for RXDB + (see mailbox/fsl,mu.txt) + maxItems: 4 + + mbox-names: + items: + - const: txdb0 + - const: txdb1 + - const: rxdb0 + - const: rxdb1 + + memory-region: + description: + phandle to a node describing reserved memory (System RAM memory) + used by DSP (see bindings/reserved-memory/reserved-memory.txt) + maxItems: 1 + +required: + - compatible + - reg + - clocks + - clock-names + - power-domains + - mboxes + - mbox-names + - memory-region + +examples: + - | + #include <dt-bindings/firmware/imx/rsrc.h> + #include <dt-bindings/clock/imx8-clock.h> + dsp@596e8000 { + compatible = "fsl,imx8qxp-dsp"; + reg = <0x596e8000 0x88000>; + clocks = <&adma_lpcg IMX_ADMA_LPCG_DSP_IPG_CLK>, + <&adma_lpcg IMX_ADMA_LPCG_OCRAM_IPG_CLK>, + <&adma_lpcg IMX_ADMA_LPCG_DSP_CORE_CLK>; + clock-names = "ipg", "ocram", "core"; + power-domains = <&pd IMX_SC_R_MU_13A>, + <&pd IMX_SC_R_MU_13B>, + <&pd IMX_SC_R_DSP>, + <&pd IMX_SC_R_DSP_RAM>; + mbox-names = "txdb0", "txdb1", "rxdb0", "rxdb1"; + mboxes = <&lsio_mu13 2 0>, <&lsio_mu13 2 1>, <&lsio_mu13 3 0>, <&lsio_mu13 3 1>; + }; diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml index e0284d8c3b63..38d4cede0860 100644 --- a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml @@ -70,7 +70,9 @@ allOf: properties: compatible: contains: - const: allwinner,sun8i-h3-spdif + enum: + - allwinner,sun8i-h3-spdif + - allwinner,sun50i-h6-spdif then: properties: diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml new file mode 100644 index 000000000000..f290eb72a878 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml @@ -0,0 +1,39 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-a64-codec-analog.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A64 Analog Codec Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens@csie.org> + - Maxime Ripard <maxime.ripard@bootlin.com> + +properties: + compatible: + const: allwinner,sun50i-a64-codec-analog + + reg: + maxItems: 1 + + cpvdd-supply: + description: + Regulator for the headphone amplifier + +required: + - compatible + - reg + - cpvdd-supply + +additionalProperties: false + +examples: + - | + codec_analog: codec-analog@1f015c0 { + compatible = "allwinner,sun50i-a64-codec-analog"; + reg = <0x01f015c0 0x4>; + cpvdd-supply = <®_eldo1>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml new file mode 100644 index 000000000000..5e7cc05bbff1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml @@ -0,0 +1,57 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun8i-a33-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A33 Codec Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens@csie.org> + - Maxime Ripard <maxime.ripard@bootlin.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: allwinner,sun8i-a33-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + audio-codec@1c22e00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun8i-a33-codec"; + reg = <0x01c22e00 0x400>; + interrupts = <0 29 4>; + clocks = <&ccu 47>, <&ccu 92>; + clock-names = "bus", "mod"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt index 4330fc9dca6d..3080979350a0 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt @@ -4,13 +4,18 @@ Required properties: - compatible: 'amlogic,axg-toddr' or 'amlogic,axg-toddr' or 'amlogic,g12a-frddr' or - 'amlogic,g12a-toddr' + 'amlogic,g12a-toddr' or + 'amlogic,sm1-frddr' or + 'amlogic,sm1-toddr' - reg: physical base address of the controller and length of memory mapped region. - interrupts: interrupt specifier for the fifo. - clocks: phandle to the fifo peripheral clock provided by the audio clock controller. -- resets: phandle to memory ARB line provided by the arb reset controller. +- resets: list of reset phandle, one for each entry reset-names. +- reset-names: should contain the following: + * "arb" : memory ARB line (required) + * "rst" : dedicated device reset line (optional) - #sound-dai-cells: must be 0. Example of FRDDR A on the A113 SoC: diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt index 73f473a9365f..716878107a24 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt @@ -2,7 +2,8 @@ Required properties: - compatible: 'amlogic,axg-pdm' or - 'amlogic,g12a-pdm' + 'amlogic,g12a-pdm' or + 'amlogic,sm1-pdm' - reg: physical base address of the controller and length of memory mapped region. - clocks: list of clock phandle, one for each entry clock-names. @@ -12,6 +13,9 @@ Required properties: * "sysclk" : dsp system clock - #sound-dai-cells: must be 0. +Optional property: +- resets: phandle to the dedicated reset line of the pdm input. + Example of PDM on the A113 SoC: pdm: audio-controller@ff632000 { diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt index 0b82504fa419..df92a4ecf288 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt @@ -2,7 +2,8 @@ Required properties: - compatible: 'amlogic,axg-spdifin' or - 'amlogic,g12a-spdifin' + 'amlogic,g12a-spdifin' or + 'amlogic,sm1-spdifin' - interrupts: interrupt specifier for the spdif input. - clocks: list of clock phandle, one for each entry clock-names. - clock-names: should contain the following: @@ -10,6 +11,9 @@ Required properties: * "refclk" : spdif input reference clock - #sound-dai-cells: must be 0. +Optional property: +- resets: phandle to the dedicated reset line of the spdif input. + Example on the A113 SoC: spdifin: audio-controller@400 { diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt index 826152730508..28381dd1f633 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt @@ -2,13 +2,17 @@ Required properties: - compatible: 'amlogic,axg-spdifout' or - 'amlogic,g12a-spdifout' + 'amlogic,g12a-spdifout' or + 'amlogic,sm1-spdifout' - clocks: list of clock phandle, one for each entry clock-names. - clock-names: should contain the following: * "pclk" : peripheral clock. * "mclk" : master clock - #sound-dai-cells: must be 0. +Optional property: +- resets: phandle to the dedicated reset line of the spdif output. + Example on the A113 SoC: spdifout: audio-controller@480 { diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt index 8835a43edfbb..5996c0cd89c2 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt @@ -4,7 +4,9 @@ Required properties: - compatible: 'amlogic,axg-tdmin' or 'amlogic,axg-tdmout' or 'amlogic,g12a-tdmin' or - 'amlogic,g12a-tdmout' + 'amlogic,g12a-tdmout' or + 'amlogic,sm1-tdmin' or + 'amlogic,sm1-tdmout - reg: physical base address of the controller and length of memory mapped region. - clocks: list of clock phandle, one for each entry clock-names. diff --git a/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt b/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt index aa6c35570d31..4e8cd7eb7cec 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt +++ b/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt @@ -1,10 +1,12 @@ * Amlogic HDMI Tx control glue Required properties: -- compatible: "amlogic,g12a-tohdmitx" +- compatible: "amlogic,g12a-tohdmitx" or + "amlogic,sm1-tohdmitx" - reg: physical base address of the controller and length of memory mapped region. - #sound-dai-cells: should be 1. +- resets: phandle to the dedicated reset line of the hdmitx glue. Example on the S905X2 SoC: @@ -12,6 +14,7 @@ tohdmitx: audio-controller@744 { compatible = "amlogic,g12a-tohdmitx"; reg = <0x0 0x744 0x0 0x4>; #sound-dai-cells = <1>; + resets = <&clkc_audio AUD_RESET_TOHDMITX>; }; Example of an 'amlogic,axg-sound-card': diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.txt b/Documentation/devicetree/bindings/sound/everest,es8316.txt new file mode 100644 index 000000000000..1bf03c5f2af4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8316.txt @@ -0,0 +1,23 @@ +Everest ES8316 audio CODEC + +This device supports both I2C and SPI. + +Required properties: + + - compatible : should be "everest,es8316" + - reg : the I2C address of the device for I2C + +Optional properties: + + - clocks : a list of phandle, should contain entries for clock-names + - clock-names : should include as follows: + "mclk" : master clock (MCLK) of the device + +Example: + +es8316: codec@11 { + compatible = "everest,es8316"; + reg = <0x11>; + clocks = <&clks 10>; + clock-names = "mclk"; +}; diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt index 5b9914367610..0e6e2166f76c 100644 --- a/Documentation/devicetree/bindings/sound/fsl,esai.txt +++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt @@ -7,8 +7,11 @@ other DSPs. It has up to six transmitters and four receivers. Required properties: - - compatible : Compatible list, must contain "fsl,imx35-esai" or - "fsl,vf610-esai" + - compatible : Compatible list, should contain one of the following + compatibles: + "fsl,imx35-esai", + "fsl,vf610-esai", + "fsl,imx6ull-esai", - reg : Offset and length of the register set for the device. diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 2e726b983845..0dc83cc4a236 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -8,7 +8,9 @@ codec/DSP interfaces. Required properties: - compatible : Compatible list, contains "fsl,vf610-sai", - "fsl,imx6sx-sai" or "fsl,imx6ul-sai" + "fsl,imx6sx-sai", "fsl,imx6ul-sai", + "fsl,imx7ulp-sai", "fsl,imx8mq-sai" or + "fsl,imx8qm-sai". - reg : Offset and length of the register set for the device. diff --git a/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt b/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt deleted file mode 100644 index 056a098495cc..000000000000 --- a/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt +++ /dev/null @@ -1,14 +0,0 @@ -* Allwinner A64 Codec Analog Controls - -Required properties: -- compatible: must be one of the following compatibles: - - "allwinner,sun50i-a64-codec-analog" -- reg: must contain the registers location and length -- cpvdd-supply: Regulator supply for the headphone amplifier - -Example: - codec_analog: codec-analog@1f015c0 { - compatible = "allwinner,sun50i-a64-codec-analog"; - reg = <0x01f015c0 0x4>; - cpvdd-supply = <®_eldo1>; - }; diff --git a/Documentation/devicetree/bindings/sound/sun8i-a33-codec.txt b/Documentation/devicetree/bindings/sound/sun8i-a33-codec.txt deleted file mode 100644 index 7ecf6bd60d27..000000000000 --- a/Documentation/devicetree/bindings/sound/sun8i-a33-codec.txt +++ /dev/null @@ -1,63 +0,0 @@ -Allwinner SUN8I audio codec ------------------------------------- - -On Sun8i-A33 SoCs, the audio is separated in different parts: - - A DAI driver. It uses the "sun4i-i2s" driver which is - documented here: - Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml - - An analog part of the codec which is handled as PRCM registers. - See Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt - - An digital part of the codec which is documented in this current - binding documentation. - - And finally, an audio card which links all the above components. - The simple-audio card will be used. - See Documentation/devicetree/bindings/sound/simple-card.txt - -This bindings documentation exposes Sun8i codec (digital part). - -Required properties: -- compatible: must be "allwinner,sun8i-a33-codec" -- reg: must contain the registers location and length -- interrupts: must contain the codec interrupt -- clocks: a list of phandle + clock-specifer pairs, one for each entry - in clock-names. -- clock-names: should contain followings: - - "bus": the parent APB clock for this controller - - "mod": the parent module clock - -Here is an example to add a sound card and the codec binding on sun8i SoCs that -are similar to A33 using simple-card: - - sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "sun8i-a33-audio"; - simple-audio-card,format = "i2s"; - simple-audio-card,frame-master = <&link_codec>; - simple-audio-card,bitclock-master = <&link_codec>; - simple-audio-card,mclk-fs = <512>; - simple-audio-card,aux-devs = <&codec_analog>; - simple-audio-card,routing = - "Left DAC", "Digital Left DAC", - "Right DAC", "Digital Right DAC"; - - simple-audio-card,cpu { - sound-dai = <&dai>; - }; - - link_codec: simple-audio-card,codec { - sound-dai = <&codec>; - }; - - soc@1c00000 { - [...] - - audio-codec@1c22e00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun8i-a33-codec"; - reg = <0x01c22e00 0x400>; - interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&ccu CLK_BUS_CODEC>, <&ccu CLK_AC_DIG>; - clock-names = "bus", "mod"; - }; - }; - diff --git a/Documentation/devicetree/bindings/sound/uda1334.txt b/Documentation/devicetree/bindings/sound/uda1334.txt new file mode 100644 index 000000000000..f64071b25e8d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/uda1334.txt @@ -0,0 +1,17 @@ +UDA1334 audio CODEC + +This device uses simple GPIO pins for controlling codec settings. + +Required properties: + + - compatible : "nxp,uda1334" + - nxp,mute-gpios: a GPIO spec for the MUTE pin. + - nxp,deemph-gpios: a GPIO spec for the De-emphasis pin + +Example: + +uda1334: audio-codec { + compatible = "nxp,uda1334"; + nxp,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; + nxp,deemph-gpios = <&gpio3 3 GPIO_ACTIVE_LOW>; +}; diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 4a3cecc8ad38..02aacd69ab96 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -1001,6 +1001,8 @@ position_fix 2 = POSBUF: use position buffer, 3 = VIACOMBO: VIA-specific workaround for capture, 4 = COMBO: use LPIB for playback, auto for capture stream + 5 = SKL+: apply the delay calculation available on recent Intel chips + 6 = FIFO: correct the position with the fixed FIFO size, for recent AMD chips probe_mask Bitmask to probe codecs (default = -1, meaning all slots); When the bit 8 (0x100) is set, the lower 8 bits are used diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index 7d7c191102a7..11298f0ce44d 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -260,6 +260,9 @@ alc295-hp-x360 HP Spectre X360 fixups alc-sense-combo Headset button support for Chrome platform +huawei-mbx-stereo + Enable initialization verbs for Huawei MBX stereo speakers; + might be risky, try this at your own risk ALC66x/67x/892 ============== diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst index 9f7347830ba4..0f3109d9abc8 100644 --- a/Documentation/sound/hd-audio/notes.rst +++ b/Documentation/sound/hd-audio/notes.rst @@ -66,6 +66,11 @@ by comparing both LPIB and position-buffer values. ``position_fix=4`` is another combination available for all controllers, and uses LPIB for the playback and the position-buffer for the capture streams. +``position_fix=5`` is specific to Intel platforms, so far, for Skylake +and onward. It applies the delay calculation for the precise position +reporting. +``position_fix=6`` is to correct the position with the fixed FIFO +size, mainly targeted for the recent AMD controllers. 0 is the default value for all other controllers, the automatic check and fallback to LPIB as described in the above. If you get a problem of repeated sounds, this option might |