March 2020 - Alsa-devel - mailman.alsa-project.org

Corrections for pcm_a52 to work with pulseaudio
by GitHub pull_request - edited 19 Mar '20

19 Mar '20

alsa-project/alsa-plugins pull request #8 was edited from amotzkau: Hi there, First this is my first contact with ALSA programming, so please forgive me for any obvious mistakes I made. On my computer Pulseaudio didn't work with the pcm_a52 plugin (Digital Surround 5.1 profile in Pulseaudio), everything was played with 30x speed (approximately). So I looked into the cause of this. Pulseaudio uses the zero-copy mmap interface of ALSA and expects that snd_pcm_mmap_commit() will accept all frames that snd_pcm_mmap_begin() allowed to send. If snd_pcm_mmap_commit() accepts less frames or none (-EAGAIN) then Pulseaudio didn't remember that or even saved those frames. They were just discarded. I thought that it would be a sound contract to guarantee that snd_pcm_mmap_commit() will always take those frames. Therefore I looked into pcm_a52, why it doesn't promise that, and changed the a52_pointer() function to offer such a guarantee. These are my patches to make it work. Best regards, Alex Request URL : https://github.com/alsa-project/alsa-plugins/pull/8 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/8.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v2 0/2] Add generic FSL CPU DAI driver
by Daniel Baluta 19 Mar '20

19 Mar '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> On platforms with a DSP we need to split the resource handling between Application Processor (AP) and DSP. On platforms where the DSP doesn't have an easy access to resources, the AP will take care of configuring them. Resources handled by this generic driver are: clocks, power domains, pinctrl. Changes since v1: - added dt-bindings patch - add missing signed-off-by - do not hardcode the name of DAI driver but derive it from newly introduced dai_index property. Daniel Baluta (2): dt-bindings: sound: Add FSL CPU DAI bindings ASoC: fsl: Add generic CPU DAI driver .../devicetree/bindings/sound/fsl,dai.yaml | 97 ++++++ sound/soc/fsl/Kconfig | 8 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_dai.c | 288 ++++++++++++++++++ 4 files changed, 395 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,dai.yaml create mode 100644 sound/soc/fsl/fsl_dai.c -- 2.17.1

4 6

Corrections for pcm_a52 to work with pulseaudio
by GitHub pull_request - opened 18 Mar '20

18 Mar '20

alsa-project/alsa-plugins pull request #8 was opened from amotzkau: Hi there, First this is my first contact with ALSA programming, so please excuse any obvious mistakes. On my computer Pulseaudio didn't work with the pcm_a52 plugin (Digital Surround 5.1 profile in Pulseaudio), everything was played with 30x speed (approximately). So I looked into the cause of this. Pulseaudio uses the zero-copy mmap interface of ALSA and expects that snd_pcm_mmap_commit() will except all frames that snd_pcm_mmap_begin() allowed to send. If snd_pcm_mmap_commit() accepts less frames or none (-EAGAIN) then Pulseaudio didn't remember that or even saved those frames. They were just discarded. I thought that it would be a sound contract to guarantee that snd_pcm_mmap_commit() will always take those frames. Therefore I looked into pcm_a52, why it doesn't promise that, and changed the a52_pointer() function to offer such a guarantee. These are my patches to make it work. Best regards, Alex Request URL : https://github.com/alsa-project/alsa-plugins/pull/8 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/8.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH 0/3] ASoC: stm32: manage rebind issue
by Olivier Moysan 18 Mar '20

18 Mar '20

This patchset corrects a rebind issue on STM32 SPDIFRX and I2S drivers. The same correction has already been applied for SAI driver: 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue") The commit e894efef9ac7 ("ASoC: core: add support to card rebind") allows to rebind the sound card after a rebind of one of its component. With this commit, the sound card is actually rebound, but may be no more functional. The following problems have been seen on STM32 drivers. 1) DMA channel is not requested: With the sound card rebind the simplified call sequence is: probe snd_soc_register_component snd_soc_try_rebind_card snd_soc_instantiate_card devm_snd_dmaengine_pcm_register The problem occurs because the pcm must be registered, before snd_soc_instantiate_card() is called. Modify the driver, to change the call sequence as follows: probe devm_snd_dmaengine_pcm_register snd_soc_register_component snd_soc_try_rebind_card 2) DMA channel is not released: dma_release_channel() is not called when devm_dmaengine_pcm_release() is executed. This occurs because SND_DMAENGINE_PCM_DRV_NAME component, has already been released through devm_component_release(). devm_dmaengine_pcm_release() should be called before devm_component_release() to avoid this problem. Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component() explicitly from the driver, to have the right sequence. Olivier Moysan (3): ASoC: stm32: spdifrx: fix regmap status check ASoC: stm32: spdifrx: manage rebind issue ASoC: stm32: i2s: manage rebind issue sound/soc/stm/stm32_i2s.c | 40 ++++++++++++++++------ sound/soc/stm/stm32_spdifrx.c | 64 +++++++++++++++++++---------------- 2 files changed, 63 insertions(+), 41 deletions(-) -- 2.17.1

2 5

Applied "ASoC: codecs: wsa881x: remove soundwire stream handling" to the asoc tree
by Mark Brown 18 Mar '20

18 Mar '20

The patch ASoC: codecs: wsa881x: remove soundwire stream handling has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 16252a8f3af77f69c2193fdc7b2f595b30845a44 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Tue, 17 Mar 2020 15:12:33 +0000 Subject: [PATCH] ASoC: codecs: wsa881x: remove soundwire stream handling There could be multiple instances of this codec on any platform, so handling stream directly in this codec driver can lead to multiple calls to prepare/enable/disable on the same SoundWire stream. Move this stream handling to machine driver to fix this issue. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20200317151233.8763-3-srinivas.kandagatla@linaro.… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wsa881x.c | 44 +------------------------------------- 1 file changed, 1 insertion(+), 43 deletions(-) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index b59f1d0e7f84..3d525297eac9 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -676,7 +676,6 @@ struct wsa881x_priv { int active_ports; bool port_prepared[WSA881X_MAX_SWR_PORTS]; bool port_enable[WSA881X_MAX_SWR_PORTS]; - bool stream_prepared; }; static void wsa881x_init(struct wsa881x_priv *wsa881x) @@ -954,41 +953,6 @@ static const struct snd_soc_dapm_widget wsa881x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("SPKR"), }; -static int wsa881x_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct wsa881x_priv *wsa881x = dev_get_drvdata(dai->dev); - int ret; - - if (wsa881x->stream_prepared) { - sdw_disable_stream(wsa881x->sruntime); - sdw_deprepare_stream(wsa881x->sruntime); - wsa881x->stream_prepared = false; - } - - - ret = sdw_prepare_stream(wsa881x->sruntime); - if (ret) - return ret; - - /** - * NOTE: there is a strict hw requirement about the ordering of port - * enables and actual PA enable. PA enable should only happen after - * soundwire ports are enabled if not DC on the line is accumulated - * resulting in Click/Pop Noise - * PA enable/mute are handled as part of DAPM and digital mute. - */ - - ret = sdw_enable_stream(wsa881x->sruntime); - if (ret) { - sdw_deprepare_stream(wsa881x->sruntime); - return ret; - } - wsa881x->stream_prepared = true; - - return ret; -} - static int wsa881x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1016,12 +980,7 @@ static int wsa881x_hw_free(struct snd_pcm_substream *substream, { struct wsa881x_priv *wsa881x = dev_get_drvdata(dai->dev); - if (wsa881x->stream_prepared) { - sdw_disable_stream(wsa881x->sruntime); - sdw_deprepare_stream(wsa881x->sruntime); - sdw_stream_remove_slave(wsa881x->slave, wsa881x->sruntime); - wsa881x->stream_prepared = false; - } + sdw_stream_remove_slave(wsa881x->slave, wsa881x->sruntime); return 0; } @@ -1052,7 +1011,6 @@ static int wsa881x_digital_mute(struct snd_soc_dai *dai, int mute, int stream) static struct snd_soc_dai_ops wsa881x_dai_ops = { .hw_params = wsa881x_hw_params, - .prepare = wsa881x_prepare, .hw_free = wsa881x_hw_free, .mute_stream = wsa881x_digital_mute, .set_sdw_stream = wsa881x_set_sdw_stream, -- 2.20.1

1 0

[PATCH v2 0/2] ASoC: sdm845: fix soundwire stream handling
by Srinivas Kandagatla 18 Mar '20

18 Mar '20

Recent addition of SoundWire stream state-machine checks in linux-next have shown an existing issue with handling soundwire streams in codec drivers. In general soundwire stream prepare/enable/disable can be called from either codec/machine/controller driver. However calling it in codec driver means that if multiple instances(Left/Right speakers) of the same codec is connected to the same stream then it will endup calling stream prepare/enable/disable more than once. This will mess up the stream state-machine checks in the soundwire core. Moving this stream handling to machine driver would fix this issue and also allow board/platform specfic power sequencing. Changes since v1: - removed false error check while setting sruntime. Srinivas Kandagatla (2): ASoC: qcom: sdm845: handle soundwire stream ASoC: codecs: wsa881x: remove soundwire stream handling sound/soc/codecs/wsa881x.c | 44 +------------------------ sound/soc/qcom/Kconfig | 2 +- sound/soc/qcom/sdm845.c | 67 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 69 insertions(+), 44 deletions(-) -- 2.21.0

2 3

[PATCH 2/2] ASoC: rt5682: remove noisy debug messages
by shumingf＠realtek.com 18 Mar '20

18 Mar '20

From: Shuming Fan <shumingf(a)realtek.com> Some debug messages are too noisy. This patch removes it. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 023bc672274b..2ee01e782fe0 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1197,11 +1197,11 @@ static int rt5682_div_sel(struct rt5682_priv *rt5682, } for (i = 0; i < size - 1; i++) { - pr_info("div[%d]=%d\n", i, div[i]); + dev_dbg(rt5682->component->dev, "div[%d]=%d\n", i, div[i]); if (target * div[i] == rt5682->sysclk) return i; if (target * div[i + 1] > rt5682->sysclk) { - pr_err("can't find div for sysclk %d\n", + dev_dbg(rt5682->component->dev, "can't find div for sysclk %d\n", rt5682->sysclk); return i; } -- 2.25.1

3 3

[PATCH 1/2] ASoC: rt5682: fix the random recording noise of headset
by shumingf＠realtek.com 18 Mar '20

18 Mar '20

From: Shuming Fan <shumingf(a)realtek.com> The cycle time of FIFO clock should increase 2 times to avoid the random recording noise issue. This setting could apply to all known situations in i2s mode. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 2 ++ sound/soc/codecs/rt5682.h | 2 ++ 2 files changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 7ca02a5e52e9..023bc672274b 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -3643,6 +3643,8 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ); regmap_update_bits(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_HV); + regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1, + RT5682_FIFO_CLK_DIV_MASK, RT5682_FIFO_CLK_DIV_2); INIT_DELAYED_WORK(&rt5682->jack_detect_work, rt5682_jack_detect_handler); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 43de6e802309..0baeece84ec4 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -657,6 +657,8 @@ #define RT5682_DMIC_1_EN_SFT 15 #define RT5682_DMIC_1_DIS (0x0 << 15) #define RT5682_DMIC_1_EN (0x1 << 15) +#define RT5682_FIFO_CLK_DIV_MASK (0x7 << 12) +#define RT5682_FIFO_CLK_DIV_2 (0x1 << 12) #define RT5682_DMIC_1_DP_MASK (0x3 << 4) #define RT5682_DMIC_1_DP_SFT 4 #define RT5682_DMIC_1_DP_GPIO2 (0x0 << 4) -- 2.25.1

4 6

[PATCH 0/2] ASoC: sdm845: fix soundwire stream handling
by Srinivas Kandagatla 18 Mar '20

18 Mar '20

Recent addition of SoundWire stream state-machine checks in linux-next have shown an existing issue with handling soundwire streams in codec drivers. In general soundwire stream prepare/enable/disable can be called from either codec/machine/controller driver. However calling it in codec driver means that if multiple instances(Left/Right speakers) of the same codec is connected to the same stream then it will endup calling stream prepare/enable/disable more than once. This will mess up the stream state-machine checks in the soundwire core. Moving this stream handling to machine driver would fix this issue and also allow board/platform specfic power sequencing. Srinivas Kandagatla (2): ASoC: qcom: sdm845: handle soundwire stream ASoC: codecs: wsa881x: remove soundwire stream handling sound/soc/codecs/wsa881x.c | 44 +----------------------- sound/soc/qcom/Kconfig | 2 +- sound/soc/qcom/sdm845.c | 69 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 71 insertions(+), 44 deletions(-) -- 2.21.0

2 8

[PATCH] python: ucm-validator: configs: Add config folder for hda-dsp card and output of alsa-info.sh script from device using it
by Mateusz Gorski 18 Mar '20

18 Mar '20

Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> --- .../hda-dsp/Acer-SwiftSF515-51T-V102.txt | 1153 +++++++++++++++++ 1 file changed, 1153 insertions(+) create mode 100644 python/ucm-validator/configs/hda-dsp/Acer-SwiftSF515-51T-V102.txt diff --git a/python/ucm-validator/configs/hda-dsp/Acer-SwiftSF515-51T-V102.txt b/python/ucm-validator/configs/hda-dsp/Acer-SwiftSF515-51T-V102.txt new file mode 100644 index 0000000..66ad82d --- /dev/null +++ b/python/ucm-validator/configs/hda-dsp/Acer-SwiftSF515-51T-V102.txt @@ -0,0 +1,1153 @@ +upload=true&script=true&cardinfo= +!!################################ +!!ALSA Information Script v 0.4.64 +!!################################ + +!!Script ran on: Wed Mar 18 09:43:34 UTC 2020 + + +!!Linux Distribution +!!------------------ + +Ubuntu 18.04.4 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 18.04.4 LTS" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 18.04.4 LTS" HOME_URL="https://www.ubuntu.com/" SUPPORT_URL="https://help.ubuntu.com/" BUG_REPORT_URL="https://bugs.launchpad.net/ubuntu/" PRIVACY_POLICY_URL="https://www.ubuntu.com/legal/terms-and-policies/privacy-policy" UBUNTU_CODENAME=bionic + + +!!DMI Information +!!--------------- + +Manufacturer: Acer +Product Name: Swift SF515-51T +Product Version: V1.02 +Firmware Version: V1.02 +Board Vendor: WL +Board Name: Guinness_WL + + +!!ACPI Device Status Information +!!--------------- + +/sys/bus/acpi/devices/10250759:00/status 15 +/sys/bus/acpi/devices/ACPI0003:00/status 15 +/sys/bus/acpi/devices/ACPI000C:00/status 15 +/sys/bus/acpi/devices/ACPI000E:00/status 15 +/sys/bus/acpi/devices/INT33A1:00/status 15 +/sys/bus/acpi/devices/INT33D5:00/status 15 +/sys/bus/acpi/devices/INT3400:00/status 15 +/sys/bus/acpi/devices/INT3403:01/status 15 +/sys/bus/acpi/devices/INT3403:03/status 15 +/sys/bus/acpi/devices/INT340E:00/status 15 +/sys/bus/acpi/devices/INT34BB:00/status 15 +/sys/bus/acpi/devices/INT3F0D:00/status 15 +/sys/bus/acpi/devices/LNXPOWER:01/status 1 +/sys/bus/acpi/devices/LNXPOWER:02/status 1 +/sys/bus/acpi/devices/LNXPOWER:03/status 15 +/sys/bus/acpi/devices/LNXPOWER:05/status 1 +/sys/bus/acpi/devices/LNXPOWER:06/status 1 +/sys/bus/acpi/devices/LNXPOWER:07/status 1 +/sys/bus/acpi/devices/LNXPOWER:09/status 1 +/sys/bus/acpi/devices/MELF0411:00/status 15 +/sys/bus/acpi/devices/MSFT0101:00/status 15 +/sys/bus/acpi/devices/PNP0103:00/status 15 +/sys/bus/acpi/devices/PNP0C02:03/status 3 +/sys/bus/acpi/devices/PNP0C02:05/status 3 +/sys/bus/acpi/devices/PNP0C0A:03/status 31 +/sys/bus/acpi/devices/PNP0C0C:00/status 15 +/sys/bus/acpi/devices/PNP0C0E:00/status 11 +/sys/bus/acpi/devices/PNP0C0F:00/status 9 +/sys/bus/acpi/devices/PNP0C0F:01/status 9 +/sys/bus/acpi/devices/PNP0C0F:02/status 9 +/sys/bus/acpi/devices/PNP0C0F:03/status 9 +/sys/bus/acpi/devices/PNP0C0F:04/status 9 +/sys/bus/acpi/devices/PNP0C0F:05/status 9 +/sys/bus/acpi/devices/PNP0C0F:06/status 9 +/sys/bus/acpi/devices/PNP0C0F:07/status 9 +/sys/bus/acpi/devices/PRP00001:00/status 11 +/sys/bus/acpi/devices/SYNA7DAB:00/status 15 +/sys/bus/acpi/devices/USBC000:00/status 15 +/sys/bus/acpi/devices/device:11/status 15 + + +!!Kernel Information +!!------------------ + +Kernel release: 5.5.7+ +Operating System: GNU/Linux +Architecture: x86_64 +Processor: x86_64 +SMP Enabled: Yes + + +!!ALSA Version +!!------------ + +Driver version: k5.5.7+ +Library version: 1.1.3 +Utilities version: 1.1.3 + + +!!Loaded ALSA modules +!!------------------- + +snd_soc_skl_hda_dsp + + +!!Sound Servers on this system +!!---------------------------- + +Pulseaudio: + Installed - Yes (/usr/bin/pulseaudio) + Running - Yes + + +!!Soundcards recognised by ALSA +!!----------------------------- + + 0 [hdadsp ]: hda-dsp - hda-dsp + WL-SwiftSF515_51T-V1.02-Guinness_WL + + +!!PCI Soundcards installed in the system +!!-------------------------------------- + +00:1f.3 Multimedia audio controller [0401]: Intel Corporation Device [8086:9dc8] (rev 30) + Subsystem: Acer Incorporated [ALI] Device [1025:1305] + + +!!Modprobe options (Sound related) +!!-------------------------------- + +snd_pcsp: index=-2 +snd_usb_audio: index=-2 +snd_atiixp_modem: index=-2 +snd_intel8x0m: index=-2 +snd_via82xx_modem: index=-2 +snd_atiixp_modem: index=-2 +snd_intel8x0m: index=-2 +snd_via82xx_modem: index=-2 +snd_usb_audio: index=-2 +snd_usb_caiaq: index=-2 +snd_usb_ua101: index=-2 +snd_usb_us122l: index=-2 +snd_usb_usx2y: index=-2 +snd_cmipci: mpu_port=0x330 fm_port=0x388 +snd_pcsp: index=-2 +snd_usb_audio: index=-2 + + +!!Loaded sound module options +!!--------------------------- + +!!Module: snd_soc_skl_hda_dsp + * : + + +!!HDA-Intel Codec information +!!--------------------------- +--startcollapse-- + +Codec: Realtek ALC256 +Address: 0 +AFG Function Id: 0x1 (unsol 1) +Vendor Id: 0x10ec0256 +Subsystem Id: 0x10251305 +Revision Id: 0x100002 +No Modem Function Group found +Default PCM: + rates [0x560]: 44100 48000 96000 192000 + bits [0xe]: 16 20 24 + formats [0x1]: PCM +Default Amp-In caps: N/A +Default Amp-Out caps: N/A +State of AFG node 0x01: + Power states: D0 D1 D2 D3 D3cold CLKSTOP EPSS + Power: setting=D0, actual=D0 +GPIO: io=3, o=0, i=0, unsolicited=1, wake=0 + IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 + IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 + IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 +Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out + Control: name="Speaker Playback Volume", index=0, device=0 + ControlAmp: chs=3, dir=Out, idx=0, ofs=0 + Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 + Amp-Out vals: [0x3c 0x3c] + Converter: stream=0, channel=0 + PCM: + rates [0x60]: 44100 48000 + bits [0xe]: 16 20 24 + formats [0x1]: PCM + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 +Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out + Control: name="Headphone Playback Volume", index=0, device=0 + ControlAmp: chs=3, dir=Out, idx=0, ofs=0 + Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 + Amp-Out vals: [0x3c 0x3c] + Converter: stream=0, channel=0 + PCM: + rates [0x60]: 44100 48000 + bits [0xe]: 16 20 24 + formats [0x1]: PCM + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 +Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono +Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono +Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital + Converter: stream=0, channel=0 + Digital: + Digital category: 0x0 + IEC Coding Type: 0x0 + PCM: + rates [0x5e0]: 44100 48000 88200 96000 192000 + bits [0xe]: 16 20 24 + formats [0x1]: PCM + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 +Node 0x07 [Audio Input] wcaps 0x10051b: Stereo Amp-In + Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 + Amp-In vals: [0x97 0x97] + Converter: stream=0, channel=0 + SDI-Select: 0 + PCM: + rates [0x560]: 44100 48000 96000 192000 + bits [0xe]: 16 20 24 + formats [0x1]: PCM + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 + Connection: 1 + 0x24 +Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In + Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 + Amp-In vals: [0x97 0x97] + Converter: stream=0, channel=0 + SDI-Select: 0 + PCM: + rates [0x560]: 44100 48000 96000 192000 + bits [0xe]: 16 20 24 + formats [0x1]: PCM + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 + Connection: 1 + 0x23 +Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In + Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 + Amp-In vals: [0x97 0x97] + Converter: stream=0, channel=0 + SDI-Select: 0 + PCM: + rates [0x560]: 44100 48000 96000 192000 + bits [0xe]: 16 20 24 + formats [0x1]: PCM + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 + Connection: 1 + 0x22 +Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono +Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono +Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono +Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono +Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono +Node 0x0f [Vendor Defined Widget] wcaps 0xf00000: Mono +Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono +Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono +Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In + Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 + Amp-In vals: [0x00 0x00] + Pincap 0x00000020: IN + Pin Default 0x40000000: [N/A] Line Out at Ext N/A + Conn = Unknown, Color = Unknown + DefAssociation = 0x0, Sequence = 0x0 + Pin-ctls: 0x00: + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 +Node 0x13 [Pin Complex] wcaps 0x40040b: Stereo Amp-In + Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 + Amp-In vals: [0x00 0x00] + Pincap 0x00000020: IN + Pin Default 0x411111f0: [N/A] Speaker at Ext Rear + Conn = 1/8, Color = Black + DefAssociation = 0xf, Sequence = 0x0 + Misc = NO_PRESENCE + Pin-ctls: 0x00: + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 +Node 0x14 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out + Control: name="Speaker Playback Switch", index=0, device=0 + ControlAmp: chs=3, dir=Out, idx=0, ofs=0 + Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 + Amp-Out vals: [0x00 0x00] + Pincap 0x00010014: OUT EAPD Detect + EAPD 0x2: EAPD + Pin Default 0x90172120: [Fixed] Speaker at Int N/A + Conn = Analog, Color = Grey + DefAssociation = 0x2, Sequence = 0x0 + Misc = NO_PRESENCE + Pin-ctls: 0x00: + Unsolicited: tag=00, enabled=0 + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 + Connection: 1 + 0x02 +Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono +Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono +Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono +Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In + Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 + Amp-In vals: [0x00 0x00] + Pincap 0x00003724: IN Detect + Vref caps: HIZ 50 GRD 80 100 + Pin Default 0x411111f0: [N/A] Speaker at Ext Rear + Conn = 1/8, Color = Black + DefAssociation = 0xf, Sequence = 0x0 + Misc = NO_PRESENCE + Pin-ctls: 0x20: IN VREF_HIZ + Unsolicited: tag=00, enabled=0 + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 +Node 0x19 [Pin Complex] wcaps 0x40048b: Stereo Amp-In + Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 + Amp-In vals: [0x00 0x00] + Pincap 0x00003724: IN Detect + Vref caps: HIZ 50 GRD 80 100 + Pin Default 0x411111f0: [N/A] Speaker at Ext Rear + Conn = 1/8, Color = Black + DefAssociation = 0xf, Sequence = 0x0 + Misc = NO_PRESENCE + Pin-ctls: 0x20: IN VREF_HIZ + Unsolicited: tag=00, enabled=0 + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 +Node 0x1a [Pin Complex] wcaps 0x40048b: Stereo Amp-In + Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 + Amp-In vals: [0x00 0x00] + Pincap 0x00003724: IN Detect + Vref caps: HIZ 50 GRD 80 100 + Pin Default 0x411111f0: [N/A] Speaker at Ext Rear + Conn = 1/8, Color = Black + DefAssociation = 0xf, Sequence = 0x0 + Misc = NO_PRESENCE + Pin-ctls: 0x00: VREF_HIZ + Unsolicited: tag=00, enabled=0 + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 +Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out + Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 + Amp-In vals: [0x00 0x00] + Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 + Amp-Out vals: [0x80 0x80] + Pincap 0x00013734: IN OUT EAPD Detect + Vref caps: HIZ 50 GRD 80 100 + EAPD 0x2: EAPD + Pin Default 0x411111f0: [N/A] Speaker at Ext Rear + Conn = 1/8, Color = Black + DefAssociation = 0xf, Sequence = 0x0 + Misc = NO_PRESENCE + Pin-ctls: 0x20: IN VREF_HIZ + Unsolicited: tag=00, enabled=0 + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 + Connection: 2 + 0x02* 0x03 +Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono +Node 0x1d [Pin Complex] wcaps 0x400400: Mono + Pincap 0x00000020: IN + Pin Default 0x4058ba2d: [N/A] Digital Out at Ext N/A + Conn = DIN, Color = UNKNOWN + DefAssociation = 0x2, Sequence = 0xd + Pin-ctls: 0x20: IN + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 +Node 0x1e [Pin Complex] wcaps 0x400781: Stereo Digital + Pincap 0x00000014: OUT Detect + Pin Default 0x411111f0: [N/A] Speaker at Ext Rear + Conn = 1/8, Color = Black + DefAssociation = 0xf, Sequence = 0x0 + Misc = NO_PRESENCE + Pin-ctls: 0x40: OUT + Unsolicited: tag=00, enabled=0 + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 + Connection: 1 + 0x06 +Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono +Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono + Processing caps: benign=0, ncoeff=91 +Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out + Control: name="Headphone Playback Switch", index=0, device=0 + ControlAmp: chs=3, dir=Out, idx=0, ofs=0 + Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 + Amp-Out vals: [0x00 0x00] + Pincap 0x0001001c: OUT HP EAPD Detect + EAPD 0x2: EAPD + Pin Default 0x02211010: [Jack] HP Out at Ext Front + Conn = 1/8, Color = Black + DefAssociation = 0x1, Sequence = 0x0 + Pin-ctls: 0xc0: OUT HP + Unsolicited: tag=01, enabled=1 + Power states: D0 D1 D2 D3 EPSS + Power: setting=D0, actual=D0 + Connection: 2 + 0x02 0x03* +Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In + Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 + Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] + Connection: 5 + 0x18 0x19 0x1a 0x1b 0x1d +Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In + Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 + Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] + Connection: 6 + 0x18 0x19 0x1a 0x1b 0x1d 0x12 +Node 0x24 [Audio Selector] wcaps 0x300101: Stereo + Connection: 2 + 0x12* 0x13 +--endcollapse-- + + +!!ALSA Device nodes +!!----------------- + +crw-rw----+ 1 root audio 116, 12 Mar 17 16:31 /dev/snd/controlC0 +crw-rw----+ 1 root audio 116, 8 Mar 17 16:31 /dev/snd/pcmC0D10p +crw-rw----+ 1 root audio 116, 9 Mar 17 16:31 /dev/snd/pcmC0D11p +crw-rw----+ 1 root audio 116, 10 Mar 17 16:31 /dev/snd/pcmC0D12p +crw-rw----+ 1 root audio 116, 11 Mar 17 16:31 /dev/snd/pcmC0D13c +crw-rw----+ 1 root audio 116, 3 Mar 17 16:31 /dev/snd/pcmC0D7c +crw-rw----+ 1 root audio 116, 2 Mar 17 16:31 /dev/snd/pcmC0D7p +crw-rw----+ 1 root audio 116, 5 Mar 17 16:31 /dev/snd/pcmC0D8c +crw-rw----+ 1 root audio 116, 4 Mar 17 16:31 /dev/snd/pcmC0D8p +crw-rw----+ 1 root audio 116, 7 Mar 17 16:31 /dev/snd/pcmC0D9c +crw-rw----+ 1 root audio 116, 6 Mar 17 16:31 /dev/snd/pcmC0D9p +crw-rw----+ 1 root audio 116, 1 Mar 17 16:31 /dev/snd/seq +crw-rw----+ 1 root audio 116, 33 Mar 17 16:31 /dev/snd/timer + +/dev/snd/by-path: +total 0 +drwxr-xr-x 2 root root 60 Mar 17 16:31 . +drwxr-xr-x 3 root root 320 Mar 17 16:31 .. +lrwxrwxrwx 1 root root 12 Mar 17 16:31 platform-skl_hda_dsp_generic -> ../controlC0 + + +!!Aplay/Arecord output +!!-------------------- + +APLAY + +Home directory not accessible: Permission denied +**** List of PLAYBACK Hardware Devices **** +card 0: hdadsp [hda-dsp], device 7: Analog HDA DSP (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 +card 0: hdadsp [hda-dsp], device 8: Digital HDA DSP (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 +card 0: hdadsp [hda-dsp], device 9: Alt Analog HDA DSP (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 +card 0: hdadsp [hda-dsp], device 10: HDMI1 (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 +card 0: hdadsp [hda-dsp], device 11: HDMI2 (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 +card 0: hdadsp [hda-dsp], device 12: HDMI3 (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 + +ARECORD + +Home directory not accessible: Permission denied +**** List of CAPTURE Hardware Devices **** +card 0: hdadsp [hda-dsp], device 7: Analog HDA DSP (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 +card 0: hdadsp [hda-dsp], device 8: Digital HDA DSP (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 +card 0: hdadsp [hda-dsp], device 9: Alt Analog HDA DSP (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 +card 0: hdadsp [hda-dsp], device 13: DMIC1 (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 + +!!Amixer output +!!------------- + +!!-------Mixer controls for card hdadsp + +Card hw:0 'hdadsp'/'WL-SwiftSF515_51T-V1.02-Guinness_WL' + Mixer name : 'Realtek ALC256' + Components : 'HDA:10ec0256,10251305,00100002' + Controls : 37 + Simple ctrls : 29 +Simple mixer control 'Master',0 + Capabilities: pvolume pvolume-joined pswitch pswitch-joined + Playback channels: Mono + Limits: Playback 0 - 87 + Mono: Playback 60 [69%] [-20.25dB] [on] +Simple mixer control 'Headphone',0 + Capabilities: pvolume pswitch + Playback channels: Front Left - Front Right + Limits: Playback 0 - 87 + Mono: + Front Left: Playback 87 [100%] [0.00dB] [on] + Front Right: Playback 87 [100%] [0.00dB] [on] +Simple mixer control 'Speaker',0 + Capabilities: pvolume pswitch + Playback channels: Front Left - Front Right + Limits: Playback 0 - 87 + Mono: + Front Left: Playback 87 [100%] [0.00dB] [on] + Front Right: Playback 87 [100%] [0.00dB] [on] +Simple mixer control 'Auto-Mute Mode',0 + Capabilities: enum + Items: 'Disabled' 'Enabled' + Item0: 'Enabled' +Simple mixer control 'Pin5-Port0 Mux',0 + Capabilities: enum + Items: 'NONE' 'cvt 2' 'cvt 3' 'cvt 4' + Item0: 'NONE' +Simple mixer control 'Pin5-Port1 Mux',0 + Capabilities: enum + Items: 'NONE' 'cvt 2' 'cvt 3' 'cvt 4' + Item0: 'NONE' +Simple mixer control 'Pin5-Port2 Mux',0 + Capabilities: enum + Items: 'NONE' 'cvt 2' 'cvt 3' 'cvt 4' + Item0: 'NONE' +Simple mixer control 'Pin6-Port0 Mux',0 + Capabilities: enum + Items: 'NONE' 'cvt 2' 'cvt 3' 'cvt 4' + Item0: 'NONE' +Simple mixer control 'Pin6-Port1 Mux',0 + Capabilities: enum + Items: 'NONE' 'cvt 2' 'cvt 3' 'cvt 4' + Item0: 'NONE' +Simple mixer control 'Pin6-Port2 Mux',0 + Capabilities: enum + Items: 'NONE' 'cvt 2' 'cvt 3' 'cvt 4' + Item0: 'NONE' +Simple mixer control 'Pin7-Port0 Mux',0 + Capabilities: enum + Items: 'NONE' 'cvt 2' 'cvt 3' 'cvt 4' + Item0: 'NONE' +Simple mixer control 'Pin7-Port1 Mux',0 + Capabilities: enum + Items: 'NONE' 'cvt 2' 'cvt 3' 'cvt 4' + Item0: 'NONE' +Simple mixer control 'Pin7-Port2 Mux',0 + Capabilities: enum + Items: 'NONE' 'cvt 2' 'cvt 3' 'cvt 4' + Item0: 'NONE' +Simple mixer control 'codec0_out mo media0_in mi',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [on] +Simple mixer control 'codec1_out mo media1_in mi',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [off] +Simple mixer control 'codec2_out mo media2_in mi',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [off] +Simple mixer control 'hif5-0 Jack',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [off] +Simple mixer control 'hif5-1 Jack',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [off] +Simple mixer control 'hif5-2 Jack',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [off] +Simple mixer control 'hif6-0 Jack',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [off] +Simple mixer control 'hif6-1 Jack',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [off] +Simple mixer control 'hif6-2 Jack',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [off] +Simple mixer control 'hif7-0 Jack',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [off] +Simple mixer control 'hif7-1 Jack',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [off] +Simple mixer control 'hif7-2 Jack',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [off] +Simple mixer control 'mch_cap_out mo mch_cap_in mi',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [on] +Simple mixer control 'media0_out mo codec0_in mi',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [on] +Simple mixer control 'media1_out mo codec1_in mi',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [off] +Simple mixer control 'media2_out mo codec2_in mi',0 + Capabilities: pswitch pswitch-joined + Playback channels: Mono + Mono: Playback [off] + + +!!Alsactl output +!!-------------- + +--startcollapse-- +state.hdadsp { + control.1 { + iface MIXER + name 'codec0_out mo media0_in mi Switch' + value true + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.2 { + iface MIXER + name 'codec1_out mo media1_in mi Switch' + value false + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.3 { + iface MIXER + name 'codec2_out mo media2_in mi Switch' + value false + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.4 { + iface MIXER + name 'media0_out mo codec0_in mi Switch' + value true + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.5 { + iface MIXER + name 'media1_out mo codec1_in mi Switch' + value false + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.6 { + iface MIXER + name 'media2_out mo codec2_in mi Switch' + value false + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.7 { + iface MIXER + name 'mch_cap_out mo mch_cap_in mi Switch' + value true + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.8 { + iface MIXER + name 'Pin5-Port0 Mux' + value NONE + comment { + access 'read write' + type ENUMERATED + count 1 + item.0 NONE + item.1 'cvt 2' + item.2 'cvt 3' + item.3 'cvt 4' + } + } + control.9 { + iface MIXER + name 'Pin5-Port1 Mux' + value NONE + comment { + access 'read write' + type ENUMERATED + count 1 + item.0 NONE + item.1 'cvt 2' + item.2 'cvt 3' + item.3 'cvt 4' + } + } + control.10 { + iface MIXER + name 'Pin5-Port2 Mux' + value NONE + comment { + access 'read write' + type ENUMERATED + count 1 + item.0 NONE + item.1 'cvt 2' + item.2 'cvt 3' + item.3 'cvt 4' + } + } + control.11 { + iface MIXER + name 'Pin6-Port0 Mux' + value NONE + comment { + access 'read write' + type ENUMERATED + count 1 + item.0 NONE + item.1 'cvt 2' + item.2 'cvt 3' + item.3 'cvt 4' + } + } + control.12 { + iface MIXER + name 'Pin6-Port1 Mux' + value NONE + comment { + access 'read write' + type ENUMERATED + count 1 + item.0 NONE + item.1 'cvt 2' + item.2 'cvt 3' + item.3 'cvt 4' + } + } + control.13 { + iface MIXER + name 'Pin6-Port2 Mux' + value NONE + comment { + access 'read write' + type ENUMERATED + count 1 + item.0 NONE + item.1 'cvt 2' + item.2 'cvt 3' + item.3 'cvt 4' + } + } + control.14 { + iface MIXER + name 'Pin7-Port0 Mux' + value NONE + comment { + access 'read write' + type ENUMERATED + count 1 + item.0 NONE + item.1 'cvt 2' + item.2 'cvt 3' + item.3 'cvt 4' + } + } + control.15 { + iface MIXER + name 'Pin7-Port1 Mux' + value NONE + comment { + access 'read write' + type ENUMERATED + count 1 + item.0 NONE + item.1 'cvt 2' + item.2 'cvt 3' + item.3 'cvt 4' + } + } + control.16 { + iface MIXER + name 'Pin7-Port2 Mux' + value NONE + comment { + access 'read write' + type ENUMERATED + count 1 + item.0 NONE + item.1 'cvt 2' + item.2 'cvt 3' + item.3 'cvt 4' + } + } + control.17 { + iface MIXER + name 'Headphone Playback Volume' + value.0 87 + value.1 87 + comment { + access 'read write' + type INTEGER + count 2 + range '0 - 87' + dbmin -6525 + dbmax 0 + dbvalue.0 0 + dbvalue.1 0 + } + } + control.18 { + iface MIXER + name 'Headphone Playback Switch' + value.0 true + value.1 true + comment { + access 'read write' + type BOOLEAN + count 2 + } + } + control.19 { + iface MIXER + name 'Speaker Playback Volume' + value.0 87 + value.1 87 + comment { + access 'read write' + type INTEGER + count 2 + range '0 - 87' + dbmin -6525 + dbmax 0 + dbvalue.0 0 + dbvalue.1 0 + } + } + control.20 { + iface MIXER + name 'Speaker Playback Switch' + value.0 true + value.1 true + comment { + access 'read write' + type BOOLEAN + count 2 + } + } + control.21 { + iface MIXER + name 'Auto-Mute Mode' + value Enabled + comment { + access 'read write' + type ENUMERATED + count 1 + item.0 Disabled + item.1 Enabled + } + } + control.22 { + iface MIXER + name 'Master Playback Volume' + value 60 + comment { + access 'read write' + type INTEGER + count 1 + range '0 - 87' + dbmin -6525 + dbmax 0 + dbvalue.0 -2025 + } + } + control.23 { + iface MIXER + name 'Master Playback Switch' + value true + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.24 { + iface CARD + name 'Front Headphone Jack' + value true + comment { + access read + type BOOLEAN + count 1 + } + } + control.25 { + iface CARD + name 'Speaker Phantom Jack' + value true + comment { + access read + type BOOLEAN + count 1 + } + } + control.26 { + iface CARD + name 'HDMI/DP, pcm=17 Jack' + value false + comment { + access read + type BOOLEAN + count 1 + } + } + control.27 { + iface CARD + name 'HDMI/DP, pcm=18 Jack' + value false + comment { + access read + type BOOLEAN + count 1 + } + } + control.28 { + iface CARD + name 'HDMI/DP, pcm=19 Jack' + value false + comment { + access read + type BOOLEAN + count 1 + } + } + control.29 { + iface MIXER + name 'hif5-0 Jack Switch' + value false + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.30 { + iface MIXER + name 'hif5-1 Jack Switch' + value false + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.31 { + iface MIXER + name 'hif5-2 Jack Switch' + value false + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.32 { + iface MIXER + name 'hif6-0 Jack Switch' + value false + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.33 { + iface MIXER + name 'hif6-1 Jack Switch' + value false + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.34 { + iface MIXER + name 'hif6-2 Jack Switch' + value false + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.35 { + iface MIXER + name 'hif7-0 Jack Switch' + value false + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.36 { + iface MIXER + name 'hif7-1 Jack Switch' + value false + comment { + access 'read write' + type BOOLEAN + count 1 + } + } + control.37 { + iface MIXER + name 'hif7-2 Jack Switch' + value false + comment { + access 'read write' + type BOOLEAN + count 1 + } + } +} +--endcollapse-- + + +!!All Loaded Modules +!!------------------ + +acpi_thermal_rel +asix +drm +drm_kms_helper +efivarfs +fb_sys_fops +i2c_algo_bit +i915 +int3400_thermal +int3403_thermal +int340x_thermal_zone +intel_lpss +intel_lpss_pci +intel_pch_thermal +intel_pmc_core +intel_powerclamp +intel_soc_dts_iosf +iwlmvm +iwlwifi +mei +mei_me +mfd_core +processor_thermal_device +snd +snd_compress +snd_hda_codec +snd_hda_codec_generic +snd_hda_codec_realtek +snd_hda_core +snd_hda_ext_core +snd_intel_dspcfg +snd_pcm +snd_rawmidi +snd_seq +snd_seq_device +snd_seq_midi +snd_seq_midi_event +snd_soc_acpi +snd_soc_acpi_intel_match +snd_soc_core +snd_soc_dmic +snd_soc_hdac_hda +snd_soc_hdac_hdmi +snd_soc_skl +snd_soc_skl_hda_dsp +snd_soc_sst_dsp +snd_soc_sst_ipc +snd_timer +soundcore +syscopyarea +sysfillrect +sysimgblt +usbnet +x86_pkg_temp_thermal +xhci_hcd +xhci_pci + + +!!ALSA/HDA dmesg +!!-------------- + +[ 1.425954] ACPI: Added _OSI(Linux-Dell-Video) +[ 1.425954] ACPI: Added _OSI(Linux-Lenovo-NV-HDMI-Audio) +[ 1.425954] ACPI: Added _OSI(Linux-HPI-Hybrid-Graphics) +-- +[ 13.086396] asix 1-4:1.0 enx00606e431d6e: renamed from eth0 +[ 13.088643] snd_soc_skl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 +[ 13.088655] snd_soc_skl 0000:00:1f.3: enabling device (0000 -> 0002) +[ 13.204656] ACPI: Video Device [GFX0] (multi-head: yes rom: no post: no) +[ 13.228149] input: Video Bus as /devices/LNXSYSTM:00/LNXSYBUS:00/PNP0A08:00/LNXVIDEO:00/input/input5 +[ 13.232518] snd_soc_skl 0000:00:1f.3: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) +[ 13.247709] fbcon: i915drmfb (fb0) is primary device +-- +[ 13.299004] i915 0000:00:02.0: fb0: i915drmfb frame buffer device +[ 13.380753] HDMI HDA Codec ehdaudio0D2: Max dais supported: 3 +[ 13.387487] snd_soc_skl 0000:00:1f.3: Direct firmware load for 9dc8-ACRSYS-ACRPRDCT-2-tplg.bin failed with error -2 +[ 13.387515] snd_soc_skl 0000:00:1f.3: tplg fw 9dc8-ACRSYS-ACRPRDCT-2-tplg.bin load failed with -2, falling back to dfw_sst.bin +[ 13.398166] snd_soc_skl 0000:00:1f.3: ASoC: Parent card not yet available, widget card binding deferred +[ 13.416863] intel-lpss 0000:00:15.1: enabling device (0000 -> 0002) +[ 13.517378] debugfs: File 'DMIC01 Rx' in directory 'dapm' already present! +[ 13.520614] HDMI HDA Codec ehdaudio0D2: hdac_hdmi_present_sense: disconnect for pin:port 5:0 +[ 13.521088] HDMI HDA Codec ehdaudio0D2: hdac_hdmi_present_sense: disconnect for pin:port 6:0 +[ 13.524154] HDMI HDA Codec ehdaudio0D2: hdac_hdmi_present_sense: disconnect for pin:port 7:0 +[ 13.533604] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC256: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:speaker +[ 13.533611] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) +[ 13.533616] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) +[ 13.533620] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 +[ 13.533624] snd_hda_codec_realtek ehdaudio0D0: inputs: +[ 13.776042] skl_hda_dsp_generic skl_hda_dsp_generic: intel-hdmi-hifi1 <-> iDisp1 Pin mapping ok +-- +[ 13.776790] skl_hda_dsp_generic skl_hda_dsp_generic: dmic-hifi <-> DMIC16k Pin mapping ok +[ 13.779057] skl_hda_dsp_generic skl_hda_dsp_generic: snd-soc-dummy-dai <-> hda-dsp-analog-dai mapping ok +[ 13.780710] skl_hda_dsp_generic skl_hda_dsp_generic: snd-soc-dummy-dai <-> hda-dsp-digital-dai mapping ok +[ 13.782562] skl_hda_dsp_generic skl_hda_dsp_generic: snd-soc-dummy-dai <-> hda-dsp-alt-analog-dai mapping ok +[ 13.783676] skl_hda_dsp_generic skl_hda_dsp_generic: snd-soc-dummy-dai <-> hdmi1 mapping ok +[ 13.784759] skl_hda_dsp_generic skl_hda_dsp_generic: snd-soc-dummy-dai <-> hdmi2 mapping ok +[ 13.785833] skl_hda_dsp_generic skl_hda_dsp_generic: snd-soc-dummy-dai <-> hdmi3 mapping ok +[ 13.787028] skl_hda_dsp_generic skl_hda_dsp_generic: snd-soc-dummy-dai <-> hda-dsp-dmic-dai mapping ok +[ 13.787942] skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi3 overwritten +-- +[ 13.788343] skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi1 overwritten +[ 13.805991] input: hda-dsp Front Headphone as /devices/platform/skl_hda_dsp_generic/sound/card0/input6 +[ 13.808237] input: hda-dsp HDMI/DP, pcm=17 Jack as /devices/platform/skl_hda_dsp_generic/sound/card0/input7 +[ 13.811958] input: hda-dsp HDMI/DP, pcm=18 Jack as /devices/platform/skl_hda_dsp_generic/sound/card0/input8 +[ 13.813135] input: hda-dsp HDMI/DP, pcm=19 Jack as /devices/platform/skl_hda_dsp_generic/sound/card0/input9 +[ 14.056305] iwlwifi 0000:00:14.3: Applying debug destination EXTERNAL_DRAM + + -- 2.17.1

1 0

[PATCH 0/2] ALSA: hda/realtek: Enable headset for Acer desktops with ALC662
by Jian-Hong Pan 17 Mar '20

17 Mar '20

This series enables the headset on Acer X2660G and N50-600 desktops with ALC662. Jian-Hong Pan (2): ALSA: hda/realtek - Enable headset mic of Acer X2660G with ALC662 ALSA: hda/realtek - Enable the headset of Acer N50-600 with ALC662 sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) -- 2.25.1

2 4

[PATCH v3] ucm2: hdadsp: add basic ucm config
by Mateusz Gorski 17 Mar '20

17 Mar '20

Basic UCM configuration for Intel Skylake SST with HDA DSP generic machine driver enabling codec playback and capture on both HDA codec and DMIC ports. Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> --- Changes in v3: - fixed error found by UCM Validator - added volume controls for Speaker and Headphones devices - removed ValueDefaults section since stereo is the default option anyway ucm2/hdadsp/Hdmi1.conf | 24 +++++++++++ ucm2/hdadsp/Hdmi2.conf | 24 +++++++++++ ucm2/hdadsp/HiFi.conf | 96 +++++++++++++++++++++++++++++++++++++++++ ucm2/hdadsp/hdadsp.conf | 16 +++++++ 4 files changed, 160 insertions(+) create mode 100644 ucm2/hdadsp/Hdmi1.conf create mode 100644 ucm2/hdadsp/Hdmi2.conf create mode 100644 ucm2/hdadsp/HiFi.conf create mode 100644 ucm2/hdadsp/hdadsp.conf diff --git a/ucm2/hdadsp/Hdmi1.conf b/ucm2/hdadsp/Hdmi1.conf new file mode 100644 index 0000000..d402dab --- /dev/null +++ b/ucm2/hdadsp/Hdmi1.conf @@ -0,0 +1,24 @@ +# Usecase for device HDMI1/Display Port stereo playback on Intel cAVS platforms +# For Audio in HDA mode + +SectionDevice."HDMI1" { + Comment "HDMI/Display Port 1 Stereo" + + Value { + PlaybackPriority 300 + PlaybackPCM "hw:${CardId},10" + If.1 { + Condition { + Type ControlExists + Control "iface=CARD,name='HDMI/DP,pcm=17 Jack'" + } + True { + JackControl "HDMI/DP,pcm=17 Jack" + } + False { + JackControl "HDMI/DP, pcm=17 Jack" + } + } + } +} + diff --git a/ucm2/hdadsp/Hdmi2.conf b/ucm2/hdadsp/Hdmi2.conf new file mode 100644 index 0000000..153bbe1 --- /dev/null +++ b/ucm2/hdadsp/Hdmi2.conf @@ -0,0 +1,24 @@ +# Usecase for device HDMI2/Display Port stereo playback on Intel cAVS platforms +# For Audio in HDA mode + +SectionDevice."HDMI2" { + Comment "HDMI/Display Port 2 Stereo" + + Value { + PlaybackPriority 400 + PlaybackPCM "hw:${CardId},11" + If.1 { + Condition { + Type ControlExists + Control "iface=CARD,name='HDMI/DP,pcm=18 Jack'" + } + True { + JackControl "HDMI/DP,pcm=18 Jack" + } + False { + JackControl "HDMI/DP, pcm=18 Jack" + } + } + } +} + diff --git a/ucm2/hdadsp/HiFi.conf b/ucm2/hdadsp/HiFi.conf new file mode 100644 index 0000000..e90ba32 --- /dev/null +++ b/ucm2/hdadsp/HiFi.conf @@ -0,0 +1,96 @@ +# Usecase for stereo playback Speaker and Headset, Recording on DMIC and Headset MIC. +# For Audio in HDA mode on Intel cAVS platforms + +SectionVerb { + + EnableSequence [ + cset "name='mch_cap_out mo mch_cap_in mi Switch' 1" + ] + + Value { + TQ "HiFi" + } +} + +SectionDevice."Headphones" { + Comment "Headphones" + + ConflictingDevice [ + "Speaker" + ] + + EnableSequence [ + cset "name='Headphone Playback Switch' 1,1" + cset "name='Speaker Playback Switch' 0,0" + ] + + Value { + PlaybackPriority 200 + PlaybackPCM "hw:${CardId},7" + PlaybackMixerElem "Headphone" + PlaybackMasterElem "Master" + PlaybackVolume "Headphone Playback Volume" + PlaybackSwitch "Headphone Playback Switch" + JackControl "Front Headphone Jack" + JackHWMute "Speaker" + } +} + +SectionDevice."Speaker" { + Comment "Speaker" + + ConflictingDevice [ + "Headphones" + ] + + EnableSequence [ + cset "name='Headphone Playback Switch' 0,0" + cset "name='Speaker Playback Switch' 1,1" + ] + + Value { + PlaybackPriority 100 + PlaybackPCM "hw:${CardId},7" + PlaybackMixerElem "Speaker" + PlaybackMasterElem "Master" + PlaybackVolume "Speaker Playback Volume" + PlaybackSwitch "Speaker Playback Switch" + } +} + +SectionDevice."Headset" { + Comment "Headset Microphone" + + ConflictingDevice [ + "Mic" + ] + + EnableSequence [ + cset "name='media0_out mo codec0_in mi Switch' 1" + ] + + Value { + CapturePriority 200 + CapturePCM "hw:${CardId},7" + } +} + +SectionDevice."Mic" { + Comment "Digital Microphone" + + ConflictingDevice [ + "Headset" + ] + + EnableSequence [ + cset "name='mch_cap_out mo mch_cap_in mi Switch' 1" + ] + + Value { + CapturePriority 100 + CapturePCM "hw:${CardId},13" + } +} + +<hdadsp/Hdmi1.conf> +<hdadsp/Hdmi2.conf> diff --git a/ucm2/hdadsp/hdadsp.conf b/ucm2/hdadsp/hdadsp.conf new file mode 100644 index 0000000..48c07c8 --- /dev/null +++ b/ucm2/hdadsp/hdadsp.conf @@ -0,0 +1,16 @@ +# UCM for Intel CAVS platforms +# For Audio in HDA and DMIC mode + +Syntax 2 + +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Play HiFi quality Music" +} + +SectionDefaults [ + cset "name='Master Playback Switch' 1" + cset "name='codec0_out mo media0_in mi Switch' 1" + cset "name='media0_out mo codec0_in mi Switch' 1" + cset "name='mch_cap_out mo mch_cap_in mi Switch' 1" +] -- 2.17.1

1 0

[PATCH v2] ucm2: hdadsp: add basic ucm config
by Mateusz Gorski 17 Mar '20

17 Mar '20

Basic UCM configuration for Intel Skylake SST HDA DSP generic machine driver enabling codec playback and capture on both HDA codec and DMIC ports. Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> --- Changes in v2: - addressed Jaroslav hints about v1 patch - added HiFi and Hdmi use cases - adjusted to ucm2 syntax ucm2/hdadsp/Hdmi1.conf | 24 +++++++++++ ucm2/hdadsp/Hdmi2.conf | 24 +++++++++++ ucm2/hdadsp/HiFi.conf | 88 +++++++++++++++++++++++++++++++++++++++++ ucm2/hdadsp/hdadsp.conf | 21 ++++++++++ 4 files changed, 157 insertions(+) create mode 100644 ucm2/hdadsp/Hdmi1.conf create mode 100644 ucm2/hdadsp/Hdmi2.conf create mode 100644 ucm2/hdadsp/HiFi.conf create mode 100644 ucm2/hdadsp/hdadsp.conf diff --git a/ucm2/hdadsp/Hdmi1.conf b/ucm2/hdadsp/Hdmi1.conf new file mode 100644 index 0000000..d402dab --- /dev/null +++ b/ucm2/hdadsp/Hdmi1.conf @@ -0,0 +1,24 @@ +# Usecase for device HDMI1/Display Port stereo playback on Intel cAVS platforms +# For Audio in HDA mode + +SectionDevice."HDMI1" { + Comment "HDMI/Display Port 1 Stereo" + + Value { + PlaybackPriority 300 + PlaybackPCM "hw:${CardId},10" + If.1 { + Condition { + Type ControlExists + Control "iface=CARD,name='HDMI/DP,pcm=17 Jack'" + } + True { + JackControl "HDMI/DP,pcm=17 Jack" + } + False { + JackControl "HDMI/DP, pcm=17 Jack" + } + } + } +} + diff --git a/ucm2/hdadsp/Hdmi2.conf b/ucm2/hdadsp/Hdmi2.conf new file mode 100644 index 0000000..153bbe1 --- /dev/null +++ b/ucm2/hdadsp/Hdmi2.conf @@ -0,0 +1,24 @@ +# Usecase for device HDMI2/Display Port stereo playback on Intel cAVS platforms +# For Audio in HDA mode + +SectionDevice."HDMI2" { + Comment "HDMI/Display Port 2 Stereo" + + Value { + PlaybackPriority 400 + PlaybackPCM "hw:${CardId},11" + If.1 { + Condition { + Type ControlExists + Control "iface=CARD,name='HDMI/DP,pcm=18 Jack'" + } + True { + JackControl "HDMI/DP,pcm=18 Jack" + } + False { + JackControl "HDMI/DP, pcm=18 Jack" + } + } + } +} + diff --git a/ucm2/hdadsp/HiFi.conf b/ucm2/hdadsp/HiFi.conf new file mode 100644 index 0000000..fe025de --- /dev/null +++ b/ucm2/hdadsp/HiFi.conf @@ -0,0 +1,88 @@ +# Usecase for stereo playback Speaker and Headset, Recording on DMIC and Headset MIC. +# For Audio in HDA mode on Intel cAVS platforms + +SectionVerb { + + EnableSequence [ + cset "name='mch_cap_out mo mch_cap_in mi Switch' 1" + ] + + Value { + TQ "HiFi" + } +} + +SectionDevice."Headphones" { + Comment "Headphones" + + ConflictingDevice [ + "Speaker" + ] + + EnableSequence [ + cset "name='Headphone Playback Switch' 1,1" + cset "name='Speaker Playback Switch' 0,0" + ] + + Value [ + PlaybackPriority 200 + PlaybackPCM "hw:${CardId},7" + JackControl "Front Headphone Jack" + JackHWMute "Speaker" + ] +} + +SectionDevice."Speaker" { + Comment "Speaker" + + ConflictingDevice [ + "Headphones" + ] + + EnableSequence [ + cset "name='Headphone Playback Switch' 0,0" + cset "name='Speaker Playback Switch' 1,1" + ] + + Value { + PlaybackPriority 100 + PlaybackPCM "hw:${CardId},7" + } +} + +SectionDevice."Headset" { + Comment "Headset Microphone" + + ConflictingDevice [ + "Mic" + ] + + EnableSequence [ + cset "name='media0_out mo codec0_in mi Switch' 1" + ] + + Value { + CapturePriority 200 + CapturePCM "hw:${CardId},7" + } +} + +SectionDevice."Mic" { + Comment "Digital Microphone" + + ConflictingDevice [ + "Headset" + ] + + EnableSequence [ + cset "name='mch_cap_out mo mch_cap_in mi Switch' 1" + ] + + Value { + CapturePriority 100 + CapturePCM "hw:${CardId},13" + } +} + +<hdadsp/Hdmi1.conf> +<hdadsp/Hdmi2.conf> diff --git a/ucm2/hdadsp/hdadsp.conf b/ucm2/hdadsp/hdadsp.conf new file mode 100644 index 0000000..a90299f --- /dev/null +++ b/ucm2/hdadsp/hdadsp.conf @@ -0,0 +1,21 @@ +# UCM for Intel CAVS platforms +# For Audio in HDA and DMIC mode + +Syntax 2 + +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Play HiFi quality Music" +} + +ValueDefaults { + PlaybackChannels "2" + CaptureChannels "2" +} + +SectionDefaults [ + cset "name='Master Playback Switch' 1" + cset "name='codec0_out mo media0_in mi Switch' 1" + cset "name='media0_out mo codec0_in mi Switch' 1" + cset "name='mch_cap_out mo mch_cap_in mi Switch' 1" +] -- 2.17.1

3 2

[PATCH 1/3] ASoC: rt5682: Add a property for DMIC clock rate setting
by Oder Chiou 17 Mar '20

17 Mar '20

The patch adds a property for DMIC clock rate setting and changes the default clock rate to the common optimize DMIC clock rate. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5682.h | 1 + sound/soc/codecs/rt5682.c | 9 +++++++-- 2 files changed, 8 insertions(+), 2 deletions(-) diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h index 6bf0e3581056..96b268ac96bd 100644 --- a/include/sound/rt5682.h +++ b/include/sound/rt5682.h @@ -38,6 +38,7 @@ struct rt5682_platform_data { enum rt5682_dmic1_clk_pin dmic1_clk_pin; enum rt5682_jd_src jd_src; unsigned int btndet_delay; + unsigned int dmic_clk_rate; const char *dai_clk_names[RT5682_DAI_NUM_CLKS]; }; diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 7ca02a5e52e9..809dc91a89b4 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1231,10 +1231,13 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - int idx = -EINVAL; + int idx = -EINVAL, dmic_clk_rate = 3072000; static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128}; - idx = rt5682_div_sel(rt5682, 1500000, div, ARRAY_SIZE(div)); + if (rt5682->pdata.dmic_clk_rate) + dmic_clk_rate = rt5682->pdata.dmic_clk_rate; + + idx = rt5682_div_sel(rt5682, dmic_clk_rate, div, ARRAY_SIZE(div)); snd_soc_component_update_bits(component, RT5682_DMIC_CTRL_1, RT5682_DMIC_CLK_MASK, idx << RT5682_DMIC_CLK_SFT); @@ -3231,6 +3234,8 @@ static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) &rt5682->pdata.jd_src); device_property_read_u32(dev, "realtek,btndet-delay", &rt5682->pdata.btndet_delay); + device_property_read_u32(dev, "realtek,dmic-clk-rate", + &rt5682->pdata.dmic_clk_rate); rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node, "realtek,ldo1-en-gpios", 0); -- 2.25.1

2 3

sof-soundwire: initial UCM2 version
by GitHub pull_request - opened 17 Mar '20

17 Mar '20

alsa-project/alsa-ucm-conf pull request #17 was opened from plbossart: Add conditional inclusion of sections This goes with the kernel PR https://github.com/thesofproject/linux/pull/1889 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/17 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/17.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Applied "ALSA: compress: add wma codec profiles" to the asoc tree
by Mark Brown 16 Mar '20

16 Mar '20

The patch ALSA: compress: add wma codec profiles has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3bd7ac41d85527126c31f336a75cb2275a6a8898 Mon Sep 17 00:00:00 2001 From: Vinod Koul <vkoul(a)kernel.org> Date: Mon, 16 Mar 2020 11:22:13 +0530 Subject: [PATCH] ALSA: compress: add wma codec profiles Some codec profiles were missing for WMA, like WMA9/10 lossless and wma10 pro, so add these profiles Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200316055221.1944464-2-vkoul@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/uapi/sound/compress_params.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index 9c96fb0e4d90..a47d9df0fd7b 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -142,6 +142,9 @@ #define SND_AUDIOPROFILE_WMA8 ((__u32) 0x00000002) #define SND_AUDIOPROFILE_WMA9 ((__u32) 0x00000004) #define SND_AUDIOPROFILE_WMA10 ((__u32) 0x00000008) +#define SND_AUDIOPROFILE_WMA9_PRO ((__u32) 0x00000010) +#define SND_AUDIOPROFILE_WMA9_LOSSLESS ((__u32) 0x00000020) +#define SND_AUDIOPROFILE_WMA10_LOSSLESS ((__u32) 0x00000040) #define SND_AUDIOMODE_WMA_LEVEL1 ((__u32) 0x00000001) #define SND_AUDIOMODE_WMA_LEVEL2 ((__u32) 0x00000002) -- 2.20.1

1 0

Applied "ALSA: compress: Add wma decoder params" to the asoc tree
by Mark Brown 16 Mar '20

16 Mar '20

The patch ALSA: compress: Add wma decoder params has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 20ff1456d26807aa5441be02f0f5459014664aad Mon Sep 17 00:00:00 2001 From: Vinod Koul <vkoul(a)kernel.org> Date: Mon, 16 Mar 2020 11:22:14 +0530 Subject: [PATCH] ALSA: compress: Add wma decoder params Some WMA decoders like WMAv10 etc need some additional encoder option parameters, so add these as WMA decoder params. Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200316055221.1944464-3-vkoul@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/uapi/sound/compress_params.h | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index a47d9df0fd7b..bf6f7155e775 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -329,6 +329,13 @@ struct snd_dec_flac { __u16 reserved; } __attribute__((packed, aligned(4))); +struct snd_dec_wma { + __u32 encoder_option; + __u32 adv_encoder_option; + __u32 adv_encoder_option2; + __u32 reserved; +} __attribute__((packed, aligned(4))); + union snd_codec_options { struct snd_enc_wma wma; struct snd_enc_vorbis vorbis; @@ -336,6 +343,7 @@ union snd_codec_options { struct snd_enc_flac flac; struct snd_enc_generic generic; struct snd_dec_flac flac_d; + struct snd_dec_wma wma_d; } __attribute__((packed, aligned(4))); /** struct snd_codec_desc - description of codec capabilities -- 2.20.1

1 0

Applied "ASoC: qcom: q6asm: pass codec profile to q6asm_open_write" to the asoc tree
by Mark Brown 16 Mar '20

16 Mar '20

The patch ASoC: qcom: q6asm: pass codec profile to q6asm_open_write has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8504a72f7ce231989cf2a82f0cdbba5b72f65015 Mon Sep 17 00:00:00 2001 From: Vinod Koul <vkoul(a)kernel.org> Date: Mon, 16 Mar 2020 11:22:15 +0530 Subject: [PATCH] ASoC: qcom: q6asm: pass codec profile to q6asm_open_write Codec profile is required to be passed for WMA codecs so that we know the codec profile present and tell DSP accordingly, so update this API to pass the codec profile as argument Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200316055221.1944464-4-vkoul@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++-- sound/soc/qcom/qdsp6/q6asm.c | 2 +- sound/soc/qcom/qdsp6/q6asm.h | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 8b48815ff918..bc0e3f7cfd8e 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -252,7 +252,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM, - prtd->bits_per_sample); + 0, prtd->bits_per_sample); } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM, prtd->bits_per_sample); @@ -654,7 +654,7 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, prtd->bits_per_sample = 16; if (dir == SND_COMPRESS_PLAYBACK) { ret = q6asm_open_write(prtd->audio_client, params->codec.id, - prtd->bits_per_sample); + params->codec.profile, prtd->bits_per_sample); if (ret < 0) { dev_err(dev, "q6asm_open_write failed\n"); diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 36e0eab13a98..64eb7b6ba305 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -858,7 +858,7 @@ static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt) * Return: Will be an negative value on error or zero on success */ int q6asm_open_write(struct audio_client *ac, uint32_t format, - uint16_t bits_per_sample) + u32 codec_profile, uint16_t bits_per_sample) { struct asm_stream_cmd_open_write_v3 *open; struct apr_pkt *pkt; diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h index 6764f55f7078..1cff7f68b95d 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -55,7 +55,7 @@ void q6asm_audio_client_free(struct audio_client *ac); int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts, uint32_t lsw_ts, uint32_t flags); int q6asm_open_write(struct audio_client *ac, uint32_t format, - uint16_t bits_per_sample); + u32 codec_profile, uint16_t bits_per_sample); int q6asm_open_read(struct audio_client *ac, uint32_t format, uint16_t bits_per_sample); -- 2.20.1

1 0

Applied "ASoC: qcom: q6asm: add support to wma config" to the asoc tree
by Mark Brown 16 Mar '20

16 Mar '20

The patch ASoC: qcom: q6asm: add support to wma config has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 97163eadf18bff306b1cf8a8fa81938c5509899c Mon Sep 17 00:00:00 2001 From: Vinod Koul <vkoul(a)kernel.org> Date: Mon, 16 Mar 2020 11:22:16 +0530 Subject: [PATCH] ASoC: qcom: q6asm: add support to wma config Qualcomm DSPs expect wma v9 and wma v10 configs to be set for wma decoders, so add the API to program the respective wma config to the DSP Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200316055221.1944464-5-vkoul@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm.c | 123 +++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 17 +++++ 2 files changed, 140 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 64eb7b6ba305..4cec95c657ba 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -39,6 +39,8 @@ #define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5 #define ASM_MEDIA_FMT_MP3 0x00010BE9 #define ASM_MEDIA_FMT_FLAC 0x00010C16 +#define ASM_MEDIA_FMT_WMA_V9 0x00010DA8 +#define ASM_MEDIA_FMT_WMA_V10 0x00010DA7 #define ASM_DATA_CMD_WRITE_V2 0x00010DAB #define ASM_DATA_CMD_READ_V2 0x00010DAC #define ASM_SESSION_CMD_SUSPEND 0x00010DEC @@ -104,6 +106,33 @@ struct asm_flac_fmt_blk_v2 { u16 reserved; } __packed; +struct asm_wmastdv9_fmt_blk_v2 { + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + u16 fmtag; + u16 num_channels; + u32 sample_rate; + u32 bytes_per_sec; + u16 blk_align; + u16 bits_per_sample; + u32 channel_mask; + u16 enc_options; + u16 reserved; +} __packed; + +struct asm_wmaprov10_fmt_blk_v2 { + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + u16 fmtag; + u16 num_channels; + u32 sample_rate; + u32 bytes_per_sec; + u16 blk_align; + u16 bits_per_sample; + u32 channel_mask; + u16 enc_options; + u16 advanced_enc_options1; + u32 advanced_enc_options2; +} __packed; + struct asm_stream_cmd_set_encdec_param { u32 param_id; u32 param_size; @@ -894,6 +923,24 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format, case SND_AUDIOCODEC_FLAC: open->dec_fmt_id = ASM_MEDIA_FMT_FLAC; break; + case SND_AUDIOCODEC_WMA: + switch (codec_profile) { + case SND_AUDIOPROFILE_WMA9: + open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V9; + break; + case SND_AUDIOPROFILE_WMA10: + case SND_AUDIOPROFILE_WMA9_PRO: + case SND_AUDIOPROFILE_WMA9_LOSSLESS: + case SND_AUDIOPROFILE_WMA10_LOSSLESS: + open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V10; + break; + default: + dev_err(ac->dev, "Invalid codec profile 0x%x\n", + codec_profile); + rc = -EINVAL; + goto err; + } + break; default: dev_err(ac->dev, "Invalid format 0x%x\n", format); rc = -EINVAL; @@ -1075,6 +1122,82 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac, return rc; } EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac); + +int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, + struct q6asm_wma_cfg *cfg) +{ + struct asm_wmastdv9_fmt_blk_v2 *fmt; + struct apr_pkt *pkt; + void *p; + int rc, pkt_size; + + pkt_size = APR_HDR_SIZE + sizeof(*fmt); + p = kzalloc(pkt_size, GFP_KERNEL); + if (!p) + return -ENOMEM; + + pkt = p; + fmt = p + APR_HDR_SIZE; + + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + + pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; + fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); + fmt->fmtag = cfg->fmtag; + fmt->num_channels = cfg->num_channels; + fmt->sample_rate = cfg->sample_rate; + fmt->bytes_per_sec = cfg->bytes_per_sec; + fmt->blk_align = cfg->block_align; + fmt->bits_per_sample = cfg->bits_per_sample; + fmt->channel_mask = cfg->channel_mask; + fmt->enc_options = cfg->enc_options; + fmt->reserved = 0; + + rc = q6asm_ac_send_cmd_sync(ac, pkt); + kfree(pkt); + + return rc; +} +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9); + +int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, + struct q6asm_wma_cfg *cfg) +{ + struct asm_wmaprov10_fmt_blk_v2 *fmt; + struct apr_pkt *pkt; + void *p; + int rc, pkt_size; + + pkt_size = APR_HDR_SIZE + sizeof(*fmt); + p = kzalloc(pkt_size, GFP_KERNEL); + if (!p) + return -ENOMEM; + + pkt = p; + fmt = p + APR_HDR_SIZE; + + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + + pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; + fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); + fmt->fmtag = cfg->fmtag; + fmt->num_channels = cfg->num_channels; + fmt->sample_rate = cfg->sample_rate; + fmt->bytes_per_sec = cfg->bytes_per_sec; + fmt->blk_align = cfg->block_align; + fmt->bits_per_sample = cfg->bits_per_sample; + fmt->channel_mask = cfg->channel_mask; + fmt->enc_options = cfg->enc_options; + fmt->advanced_enc_options1 = cfg->adv_enc_options; + fmt->advanced_enc_options2 = cfg->adv_enc_options2; + + rc = q6asm_ac_send_cmd_sync(ac, pkt); + kfree(pkt); + + return rc; +} +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10); + /** * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture * diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h index 1cff7f68b95d..5d9fbc75688c 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -45,6 +45,19 @@ struct q6asm_flac_cfg { u16 md5_sum; }; +struct q6asm_wma_cfg { + u32 fmtag; + u32 num_channels; + u32 sample_rate; + u32 bytes_per_sec; + u32 block_align; + u32 bits_per_sample; + u32 channel_mask; + u32 enc_options; + u32 adv_enc_options; + u32 adv_enc_options2; +}; + typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token, void *payload, void *priv); struct audio_client; @@ -69,6 +82,10 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac, uint16_t bits_per_sample); int q6asm_stream_media_format_block_flac(struct audio_client *ac, struct q6asm_flac_cfg *cfg); +int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, + struct q6asm_wma_cfg *cfg); +int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, + struct q6asm_wma_cfg *cfg); int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts); int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, -- 2.20.1

1 0

Applied "ASoC: qcom: q6asm-dai: add support to wma decoder" to the asoc tree
by Mark Brown 16 Mar '20

16 Mar '20

The patch ASoC: qcom: q6asm-dai: add support to wma decoder has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 40519a1c02303ebdf09e07fe57a52c2d1d188b01 Mon Sep 17 00:00:00 2001 From: Vinod Koul <vkoul(a)kernel.org> Date: Mon, 16 Mar 2020 11:22:17 +0530 Subject: [PATCH] ASoC: qcom: q6asm-dai: add support to wma decoder Qualcomm DSPs also supports the wma decoder, so add support for wma decoder and convert the snd_codec_params to qdsp format. Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200316055221.1944464-6-vkoul@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 67 +++++++++++++++++++++++++++++++- 1 file changed, 66 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index bc0e3f7cfd8e..fa685fe4a027 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -629,10 +629,13 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, int dir = stream->direction; struct q6asm_dai_data *pdata; struct q6asm_flac_cfg flac_cfg; + struct q6asm_wma_cfg wma_cfg; + unsigned int wma_v9 = 0; struct device *dev = c->dev; int ret; union snd_codec_options *codec_options; struct snd_dec_flac *flac; + struct snd_dec_wma *wma; codec_options = &(prtd->codec_param.codec.options); @@ -694,6 +697,67 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, return -EIO; } break; + + case SND_AUDIOCODEC_WMA: + wma = &codec_options->wma_d; + + memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg)); + + wma_cfg.sample_rate = params->codec.sample_rate; + wma_cfg.num_channels = params->codec.ch_in; + wma_cfg.bytes_per_sec = params->codec.bit_rate / 8; + wma_cfg.block_align = params->codec.align; + wma_cfg.bits_per_sample = prtd->bits_per_sample; + wma_cfg.enc_options = wma->encoder_option; + wma_cfg.adv_enc_options = wma->adv_encoder_option; + wma_cfg.adv_enc_options2 = wma->adv_encoder_option2; + + if (wma_cfg.num_channels == 1) + wma_cfg.channel_mask = 4; /* Mono Center */ + else if (wma_cfg.num_channels == 2) + wma_cfg.channel_mask = 3; /* Stereo FL/FR */ + else + return -EINVAL; + + /* check the codec profile */ + switch (params->codec.profile) { + case SND_AUDIOPROFILE_WMA9: + wma_cfg.fmtag = 0x161; + wma_v9 = 1; + break; + + case SND_AUDIOPROFILE_WMA10: + wma_cfg.fmtag = 0x166; + break; + + case SND_AUDIOPROFILE_WMA9_PRO: + wma_cfg.fmtag = 0x162; + break; + + case SND_AUDIOPROFILE_WMA9_LOSSLESS: + wma_cfg.fmtag = 0x163; + break; + + case SND_AUDIOPROFILE_WMA10_LOSSLESS: + wma_cfg.fmtag = 0x167; + break; + + default: + dev_err(dev, "Unknown WMA profile:%x\n", + params->codec.profile); + return -EIO; + } + + if (wma_v9) + ret = q6asm_stream_media_format_block_wma_v9( + prtd->audio_client, &wma_cfg); + else + ret = q6asm_stream_media_format_block_wma_v10( + prtd->audio_client, &wma_cfg); + if (ret < 0) { + dev_err(dev, "WMA9 CMD failed:%d\n", ret); + return -EIO; + } default: break; } @@ -793,9 +857,10 @@ static int q6asm_dai_compr_get_caps(struct snd_compr_stream *stream, caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE; caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS; caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS; - caps->num_codecs = 2; + caps->num_codecs = 3; caps->codecs[0] = SND_AUDIOCODEC_MP3; caps->codecs[1] = SND_AUDIOCODEC_FLAC; + caps->codecs[2] = SND_AUDIOCODEC_WMA; return 0; } -- 2.20.1

1 0

Applied "ALSA: compress: add alac & ape decoder params" to the asoc tree
by Mark Brown 16 Mar '20

16 Mar '20

The patch ALSA: compress: add alac & ape decoder params has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0f546d6f0292fb624b4cf0cc70096ddb9ea070e6 Mon Sep 17 00:00:00 2001 From: Vinod Koul <vkoul(a)kernel.org> Date: Mon, 16 Mar 2020 11:22:18 +0530 Subject: [PATCH] ALSA: compress: add alac & ape decoder params Add ALAC (Apple Lossless Audio Codec) and APE (Monkey's Lossless Audio Codec) defines and parameters required to configure these. Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200316055221.1944464-7-vkoul@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/uapi/sound/compress_params.h | 26 +++++++++++++++++++++++++- 1 file changed, 25 insertions(+), 1 deletion(-) diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index bf6f7155e775..79b14389ae41 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -75,7 +75,9 @@ #define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C) #define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D) #define SND_AUDIOCODEC_BESPOKE ((__u32) 0x0000000E) -#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_BESPOKE +#define SND_AUDIOCODEC_ALAC ((__u32) 0x0000000F) +#define SND_AUDIOCODEC_APE ((__u32) 0x00000010) +#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_APE /* * Profile and modes are listed with bit masks. This allows for a @@ -336,6 +338,26 @@ struct snd_dec_wma { __u32 reserved; } __attribute__((packed, aligned(4))); +struct snd_dec_alac { + __u32 frame_length; + __u8 compatible_version; + __u8 pb; + __u8 mb; + __u8 kb; + __u32 max_run; + __u32 max_frame_bytes; +} __attribute__((packed, aligned(4))); + +struct snd_dec_ape { + __u16 compatible_version; + __u16 compression_level; + __u32 format_flags; + __u32 blocks_per_frame; + __u32 final_frame_blocks; + __u32 total_frames; + __u32 seek_table_present; +} __attribute__((packed, aligned(4))); + union snd_codec_options { struct snd_enc_wma wma; struct snd_enc_vorbis vorbis; @@ -344,6 +366,8 @@ union snd_codec_options { struct snd_enc_generic generic; struct snd_dec_flac flac_d; struct snd_dec_wma wma_d; + struct snd_dec_alac alac_d; + struct snd_dec_ape ape_d; } __attribute__((packed, aligned(4))); /** struct snd_codec_desc - description of codec capabilities -- 2.20.1

1 0

Applied "ASoC: qcom: q6asm: add support for alac and ape configs" to the asoc tree
by Mark Brown 16 Mar '20

16 Mar '20

The patch ASoC: qcom: q6asm: add support for alac and ape configs has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7076bf4da0b1b53dd5d96dc930c9899a8ad6f217 Mon Sep 17 00:00:00 2001 From: Vinod Koul <vkoul(a)kernel.org> Date: Mon, 16 Mar 2020 11:22:19 +0530 Subject: [PATCH] ASoC: qcom: q6asm: add support for alac and ape configs Qualcomm DSPs expect ALAC and APE configs to be send for decoders, so add the API to program the respective config to the DSP. Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200316055221.1944464-8-vkoul@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm.c | 118 +++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 32 ++++++++++ 2 files changed, 150 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 4cec95c657ba..0e0e8f7a460a 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -48,6 +48,8 @@ #define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4 #define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A #define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D +#define ASM_MEDIA_FMT_ALAC 0x00012f31 +#define ASM_MEDIA_FMT_APE 0x00012f32 #define ASM_LEGACY_STREAM_SESSION 0 @@ -133,6 +135,36 @@ struct asm_wmaprov10_fmt_blk_v2 { u32 advanced_enc_options2; } __packed; +struct asm_alac_fmt_blk_v2 { + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + u32 frame_length; + u8 compatible_version; + u8 bit_depth; + u8 pb; + u8 mb; + u8 kb; + u8 num_channels; + u16 max_run; + u32 max_frame_bytes; + u32 avg_bit_rate; + u32 sample_rate; + u32 channel_layout_tag; +} __packed; + +struct asm_ape_fmt_blk_v2 { + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + u16 compatible_version; + u16 compression_level; + u32 format_flags; + u32 blocks_per_frame; + u32 final_frame_blocks; + u32 total_frames; + u16 bits_per_sample; + u16 num_channels; + u32 sample_rate; + u32 seek_table_present; +} __packed; + struct asm_stream_cmd_set_encdec_param { u32 param_id; u32 param_size; @@ -941,6 +973,12 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format, goto err; } break; + case SND_AUDIOCODEC_ALAC: + open->dec_fmt_id = ASM_MEDIA_FMT_ALAC; + break; + case SND_AUDIOCODEC_APE: + open->dec_fmt_id = ASM_MEDIA_FMT_APE; + break; default: dev_err(ac->dev, "Invalid format 0x%x\n", format); rc = -EINVAL; @@ -1198,6 +1236,86 @@ int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, } EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10); +int q6asm_stream_media_format_block_alac(struct audio_client *ac, + struct q6asm_alac_cfg *cfg) +{ + struct asm_alac_fmt_blk_v2 *fmt; + struct apr_pkt *pkt; + void *p; + int rc, pkt_size; + + pkt_size = APR_HDR_SIZE + sizeof(*fmt); + p = kzalloc(pkt_size, GFP_KERNEL); + if (!p) + return -ENOMEM; + + pkt = p; + fmt = p + APR_HDR_SIZE; + + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + + pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; + fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); + + fmt->frame_length = cfg->frame_length; + fmt->compatible_version = cfg->compatible_version; + fmt->bit_depth = cfg->bit_depth; + fmt->num_channels = cfg->num_channels; + fmt->max_run = cfg->max_run; + fmt->max_frame_bytes = cfg->max_frame_bytes; + fmt->avg_bit_rate = cfg->avg_bit_rate; + fmt->sample_rate = cfg->sample_rate; + fmt->channel_layout_tag = cfg->channel_layout_tag; + fmt->pb = cfg->pb; + fmt->mb = cfg->mb; + fmt->kb = cfg->kb; + + rc = q6asm_ac_send_cmd_sync(ac, pkt); + kfree(pkt); + + return rc; +} +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac); + +int q6asm_stream_media_format_block_ape(struct audio_client *ac, + struct q6asm_ape_cfg *cfg) +{ + struct asm_ape_fmt_blk_v2 *fmt; + struct apr_pkt *pkt; + void *p; + int rc, pkt_size; + + pkt_size = APR_HDR_SIZE + sizeof(*fmt); + p = kzalloc(pkt_size, GFP_KERNEL); + if (!p) + return -ENOMEM; + + pkt = p; + fmt = p + APR_HDR_SIZE; + + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + + pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; + fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); + + fmt->compatible_version = cfg->compatible_version; + fmt->compression_level = cfg->compression_level; + fmt->format_flags = cfg->format_flags; + fmt->blocks_per_frame = cfg->blocks_per_frame; + fmt->final_frame_blocks = cfg->final_frame_blocks; + fmt->total_frames = cfg->total_frames; + fmt->bits_per_sample = cfg->bits_per_sample; + fmt->num_channels = cfg->num_channels; + fmt->sample_rate = cfg->sample_rate; + fmt->seek_table_present = cfg->seek_table_present; + + rc = q6asm_ac_send_cmd_sync(ac, pkt); + kfree(pkt); + + return rc; +} +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape); + /** * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture * diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h index 5d9fbc75688c..38a207d6cd95 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -58,6 +58,34 @@ struct q6asm_wma_cfg { u32 adv_enc_options2; }; +struct q6asm_alac_cfg { + u32 frame_length; + u8 compatible_version; + u8 bit_depth; + u8 pb; + u8 mb; + u8 kb; + u8 num_channels; + u16 max_run; + u32 max_frame_bytes; + u32 avg_bit_rate; + u32 sample_rate; + u32 channel_layout_tag; +}; + +struct q6asm_ape_cfg { + u16 compatible_version; + u16 compression_level; + u32 format_flags; + u32 blocks_per_frame; + u32 final_frame_blocks; + u32 total_frames; + u16 bits_per_sample; + u16 num_channels; + u32 sample_rate; + u32 seek_table_present; +}; + typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token, void *payload, void *priv); struct audio_client; @@ -86,6 +114,10 @@ int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, struct q6asm_wma_cfg *cfg); int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, struct q6asm_wma_cfg *cfg); +int q6asm_stream_media_format_block_alac(struct audio_client *ac, + struct q6asm_alac_cfg *cfg); +int q6asm_stream_media_format_block_ape(struct audio_client *ac, + struct q6asm_ape_cfg *cfg); int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts); int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, -- 2.20.1

1 0

[PATCH v3 0/9] ALSA: compress: Add wma, alac and ape support
by Vinod Koul 16 Mar '20

16 Mar '20

This series adds more WMA profiles and WMA decoder parameters to UAPI and then support for these in qcom driver. It also adds FLAC and APE IDs and decoder parameters to UAPI and then support in qcom driver This was tested on Dragon board RB3. Last, bump up the compressed version so that userspace can check for the support. Since the series touches compress uapi and asoc, it would make sense to go thru asoc tree with acks. Changes in v3: - add r-b from Srini - use macros for FLAC channel layout tags Changes in v2: - use bitflags for wma profiles Vinod Koul (9): ALSA: compress: add wma codec profiles ALSA: compress: Add wma decoder params ASoC: qcom: q6asm: pass codec profile to q6asm_open_write ASoC: qcom: q6asm: add support to wma config ASoC: qcom: q6asm-dai: add support to wma decoder ALSA: compress: add alac & ape decoder params ASoC: qcom: q6asm: add support for alac and ape configs ASoC: qcom: q6asm-dai: add support for ALAC and APE decoders ALSA: compress: bump the version include/uapi/sound/compress_offload.h | 2 +- include/uapi/sound/compress_params.h | 37 +++- sound/soc/qcom/qdsp6/q6asm-dai.c | 139 ++++++++++++++- sound/soc/qcom/qdsp6/q6asm.c | 243 +++++++++++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.h | 51 +++++- 5 files changed, 465 insertions(+), 7 deletions(-) -- 2.24.1

3 11

Applied "ALSA: compress: bump the version" to the asoc tree
by Mark Brown 16 Mar '20

16 Mar '20

The patch ALSA: compress: bump the version has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 54ce83a3080c3d06b14faa7533a9adb4e158dcea Mon Sep 17 00:00:00 2001 From: Vinod Koul <vkoul(a)kernel.org> Date: Mon, 16 Mar 2020 11:22:21 +0530 Subject: [PATCH] ALSA: compress: bump the version We have added support for bunch of new decoders and parameters for decoders. To help users find the support bump the version up to 0,2,0. Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200316055221.1944464-10-vkoul@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/uapi/sound/compress_offload.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 56d95673ce0f..7184265c0b0d 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -31,7 +31,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 2) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 2, 0) /** * struct snd_compressed_buffer - compressed buffer * @fragment_size: size of buffer fragment in bytes -- 2.20.1

1 0

[PATCH] ASoC: soc-dai: return proper error for get_sdw_stream()
by Srinivas Kandagatla 16 Mar '20

16 Mar '20

snd_soc_dai_get_sdw_stream() returns null if dai does not support this callback, this is no very useful for the caller to differentiate if this is an error or unsupported call for the dai. return -ENOTSUPP in cases where this callback is not supported. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- include/sound/soc-dai.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 7f70db149b81..78bac995db15 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -433,7 +433,7 @@ static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai, * This routine only retrieves that was previously configured * with snd_soc_dai_get_sdw_stream() * - * Returns pointer to stream or NULL; + * Returns pointer to stream or -ENOTSUPP if callback is not supported; */ static inline void *snd_soc_dai_get_sdw_stream(struct snd_soc_dai *dai, int direction) @@ -441,7 +441,7 @@ static inline void *snd_soc_dai_get_sdw_stream(struct snd_soc_dai *dai, if (dai->driver->ops->get_sdw_stream) return dai->driver->ops->get_sdw_stream(dai, direction); else - return NULL; + return ERR_PTR(-ENOTSUPP); } #endif -- 2.21.0

2 1