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

[PATCH] ASoC: codecs: wsa881x: request gpio direction before setting
by Srinivas Kandagatla 16 Mar '20

16 Mar '20

Make sure that power down gpio direction is set to ouput before even setting it. Fixes: a0aab9e1404a ("ASoC: codecs: add wsa881x amplifier support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wsa881x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index 35b44b297f9e..1810e0775efe 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -1154,7 +1154,7 @@ static int wsa881x_probe(struct sdw_slave *pdev, wsa881x->sconfig.type = SDW_STREAM_PDM; pdev->prop.sink_ports = GENMASK(WSA881X_MAX_SWR_PORTS, 0); pdev->prop.sink_dpn_prop = wsa_sink_dpn_prop; - gpiod_set_value(wsa881x->sd_n, 1); + gpiod_direction_output(wsa881x->sd_n, 1); wsa881x->regmap = devm_regmap_init_sdw(pdev, &wsa881x_regmap_config); if (IS_ERR(wsa881x->regmap)) { -- 2.21.0

2 1

[alsa-devel] [PATCH v3 0/9] x86: Easy way of detecting MS Surface 3
by Andy Shevchenko 16 Mar '20

16 Mar '20

While working on RTC regression, I noticed that we are using the same DMI check over and over in the drivers for MS Surface 3 platform. This series dedicated for making it easier in the same way how it's done for Apple machines. Changelog v3: - fixed typo in patch 5 (Jonathan) - returned back to if {} else {} condition in ASoC driver (Mark) - added Mark's Ack tag Changelog v2: - removed RTC patches for now (the fix will be independent to this series) - added couple more clean ups to arch/x86/kernel/quirks.c - redone DMI quirk to use driver_data instead of callback - simplified check in soc-acpi-intel-cht-match.c to be oneliner - added a new patch to cover rt5645 codec driver Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Andy Shevchenko (9): x86/platform: Rename x86/apple.h -> x86/machine.h x86/quirks: Add missed include to satisfy static checker x86/quirks: Introduce hpet_dev_print_force_hpet_address() helper x86/quirks: Join string literals back x86/quirks: Convert DMI matching to use a table x86/quirks: Add a DMI quirk for Microsoft Surface 3 platform/x86: surface3_wmi: Switch DMI table match to a test of variable ASoC: rt5645: Switch DMI table match to a test of variable ASoC: Intel: Switch DMI table match to a test of variable arch/x86/kernel/quirks.c | 91 +++++++++++++------ drivers/platform/x86/surface3-wmi.c | 16 +--- include/linux/platform_data/x86/apple.h | 14 +-- include/linux/platform_data/x86/machine.h | 20 ++++ sound/soc/codecs/rt5645.c | 14 ++- .../intel/common/soc-acpi-intel-cht-match.c | 28 +----- 6 files changed, 93 insertions(+), 90 deletions(-) create mode 100644 include/linux/platform_data/x86/machine.h -- 2.24.1

2 6

[PATCH 0/3] ALSA USB MIDI: Predictable low latency for USB MIDI output
by Andreas Steinmetz 16 Mar '20

16 Mar '20

The following patch series fixes USB MIDI output starvation and allows to configure a predictable worst case output latency with a minimum of 15ms for 16x16 devices and 3.8ms for 4x4 devices (excluding USB transfer time and latencies of the interface device itself). The reasoning behind this patch series is first to fix the possible indefinite latency and then to allow to configure a predictable low latency to be able to use a cheap embedded Linux system (e.g. Raspberry Pi) in combination with a multiport USB MIDI interface as a drop in replacement for commercial hardware based MIDI routing and filtering products. This includes "real time" (live) usage. For this usage the guaranteed latency within the kernel must be as low as possible, depending on the requirements of the USB MIDI interface in use. Though the current state of the kernel code is acceptable in a fully controlled by a single user environment, it is not usable for uncontrolled standalone use as the driver behaviour then depends on unpredictable MIDI data streams and thus is unpredictable in behaviour itself. After applying this patch series the USB MIDI driver can be used as usual or it can be tuned to low latency when this is desired. >From a developer perspective the actual output rate of USB MIDI interfaces is typically slower than the expected 320us and even varies depending on how many ports are busy (e.g. a M-Audio Midisport 4x4 AE interface exhibits a varying output rate of 357us to 515us per byte depending on device load). Thus the driver needs a predictable low latency to allow userspace to use the rawmidi buffer fill state to estimate the expected output latency and schedule data accordingly. The following table shows relative processing time for varying driver configurations using the M-Audio Midisport 4x4 AE as the test device. For the parameters "outqueue" and "events" do see the descriptions of the following patches. | unpatched | outqueue=7 | outqueue=1 | outqueue=1 | | kernel | events=0 | events=0 | events=4 | --------+---------------+---------------+---------------+---------------+ 1 Port | 100% | 100% | 100% | 100.05% | --------+---------------+---------------+---------------+---------------+ 4 Ports | n/a | 100% | 99.97% | 100.36% | --------+---------------+---------------+---------------+---------------+ As can be seen from the above table the changes to output latency do affect throughput only marginally.

2 3

[PATCH 0/2] ALSA: seq: Fix running status after receiving sysex
by Takashi Iwai 16 Mar '20

16 Mar '20

Hi, here are two small fixes for correcting the behavior of sequencer core wrt the event decoding after receiving a sysex message. Takashi === Takashi Iwai (2): ALSA: seq: virmidi: Fix running status after receiving sysex ALSA: seq: oss: Fix running status after receiving sysex sound/core/seq/oss/seq_oss_midi.c | 1 + sound/core/seq/seq_virmidi.c | 1 + 2 files changed, 2 insertions(+) -- 2.16.4

1 2

BUG: ALSA snd-virmidi: snd-virmidi is not MIDI 1.0 specification compliant
by Andreas Steinmetz 16 Mar '20

16 Mar '20

snd-virmidi, as of kernel 5.4.23, does not adhere to the MIDI specification: >From MIDI 1.0 Detailed Specification, Document Version 4.2, Appendix A- 1 (RUNNING STATUS), Page A-2: "3. Buffer is cleared when a System Exclusive or Common status message is received." snd-virmidi does not clear the running status in case of a SysEx message. How to test: Load snd-virmidi module and get the card number of the module. In the following 5 is assumed as the card number and device 1 is used. You need two shells, in shell 1 do: aconnect 'Virtual Raw MIDI 5-1':0 'Virtual Raw MIDI 5-1':0 amidi -p hw:VirMIDI,1 -d Now, in shell 2 do: amidi -p hw:VirMIDI,1 -S '80 01 02 F0 01 F7 80 02 03' This sends a note off, followed by a SysEx, followed by another note off. The result in shell 1 is: 80 01 02 F0 01 F7 02 03 So snd-virmidi did not clear the running status. The correct result would be: 80 01 02 F0 01 F7 80 02 03

2 1

[PATCH] ASoC: Intel: boards: add stereo playback by woofer speaker
by mac.chiang＠intel.com 16 Mar '20

16 Mar '20

From: Mac Chiang <mac.chiang(a)intel.com> Separate woofer stereo speakers settings. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/cml_rt1011_rt5682.c | 221 +++++++++++++++++++++++------ 1 file changed, 180 insertions(+), 41 deletions(-) diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index dd80d01..fc1bd94 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -30,6 +30,36 @@ #define CML_RT5682_CODEC_DAI "rt5682-aif1" #define NAME_SIZE 32 +#define SOF_RT1011_SPEAKER_WL BIT(0) +#define SOF_RT1011_SPEAKER_WR BIT(1) +#define SOF_RT1011_SPEAKER_TL BIT(2) +#define SOF_RT1011_SPEAKER_TR BIT(3) +#define SPK_CH 4 + +/* Default: Woofer+Tweeter speakers */ +static unsigned long sof_rt1011_quirk = SOF_RT1011_SPEAKER_WL | SOF_RT1011_SPEAKER_WR | + SOF_RT1011_SPEAKER_TL | SOF_RT1011_SPEAKER_TR; + +static int sof_rt1011_quirk_cb(const struct dmi_system_id *id) +{ + sof_rt1011_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id sof_rt1011_quirk_table[] = { + { + .callback = sof_rt1011_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + DMI_MATCH(DMI_PRODUCT_NAME, "Palkia"), + }, + .driver_data = (void *)(SOF_RT1011_SPEAKER_WL | + SOF_RT1011_SPEAKER_WR), + }, + { + } +}; + static struct snd_soc_jack hdmi_jack[3]; struct hdmi_pcm { @@ -48,15 +78,16 @@ struct card_private { static const struct snd_kcontrol_new cml_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("TL Ext Spk"), - SOC_DAPM_PIN_SWITCH("TR Ext Spk"), SOC_DAPM_PIN_SWITCH("WL Ext Spk"), SOC_DAPM_PIN_SWITCH("WR Ext Spk"), }; +static const struct snd_kcontrol_new cml_rt1011_tt_controls[] = { + SOC_DAPM_PIN_SWITCH("TL Ext Spk"), + SOC_DAPM_PIN_SWITCH("TR Ext Spk"), +}; + static const struct snd_soc_dapm_widget cml_rt1011_rt5682_widgets[] = { - SND_SOC_DAPM_SPK("TL Ext Spk", NULL), - SND_SOC_DAPM_SPK("TR Ext Spk", NULL), SND_SOC_DAPM_SPK("WL Ext Spk", NULL), SND_SOC_DAPM_SPK("WR Ext Spk", NULL), SND_SOC_DAPM_HP("Headphone Jack", NULL), @@ -64,6 +95,11 @@ static const struct snd_soc_dapm_widget cml_rt1011_rt5682_widgets[] = { SND_SOC_DAPM_MIC("SoC DMIC", NULL), }; +static const struct snd_soc_dapm_widget cml_rt1011_tt_widgets[] = { + SND_SOC_DAPM_SPK("TL Ext Spk", NULL), + SND_SOC_DAPM_SPK("TR Ext Spk", NULL), +}; + static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = { /*speaker*/ {"TL Ext Spk", NULL, "TL SPO"}, @@ -82,6 +118,12 @@ static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = { {"DMic", NULL, "SoC DMIC"}, }; +static const struct snd_soc_dapm_route cml_rt1011_tt_map[] = { + /*TL/TR speaker*/ + {"TL Ext Spk", NULL, "TL SPO" }, + {"TR Ext Spk", NULL, "TR SPO" }, +}; + static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -121,6 +163,35 @@ static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; }; +static int cml_rt1011_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret = 0; + struct snd_soc_card *card = rtd->card; + + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + + ret = snd_soc_add_card_controls(card, cml_rt1011_tt_controls, + ARRAY_SIZE(cml_rt1011_tt_controls)); + if (ret) + return ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, + cml_rt1011_tt_widgets, + ARRAY_SIZE(cml_rt1011_tt_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, cml_rt1011_tt_map, + ARRAY_SIZE(cml_rt1011_tt_map)); + + if (ret) + return ret; + } + + return ret; +} + static int cml_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -192,31 +263,52 @@ static int cml_rt1011_hw_params(struct snd_pcm_substream *substream, * The feedback is captured for each codec individually. * Hence all 4 codecs use 1 Tx slot each for feedback. */ - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x4, 0x1, 4, 24); - if (ret < 0) - break; - } - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:02")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x1, 0x1, 4, 24); - if (ret < 0) - break; - } - /* TDM Rx slot 2 is used for Right Woofer & Tweeters pair */ - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x8, 0x2, 4, 24); - if (ret < 0) - break; - } - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:03")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x2, 0x2, 4, 24); - if (ret < 0) - break; + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x4, 0x1, 4, 24); + if (ret < 0) + break; + } + + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:02")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x1, 0x1, 4, 24); + if (ret < 0) + break; + } + + /* TDM Rx slot 2 is used for Right Woofer & Tweeters pair */ + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x8, 0x2, 4, 24); + if (ret < 0) + break; + } + + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:03")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x2, 0x2, 4, 24); + if (ret < 0) + break; + } + } else { + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x4, 0x1, 4, 24); + if (ret < 0) + break; + } + + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x8, 0x2, 4, 24); + if (ret < 0) + break; + } } + } if (ret < 0) dev_err(rtd->dev, @@ -302,10 +394,8 @@ SND_SOC_DAILINK_DEF(ssp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); SND_SOC_DAILINK_DEF(ssp1_codec, DAILINK_COMP_ARRAY( - /* WL */ COMP_CODEC("i2c-10EC1011:00", CML_RT1011_CODEC_DAI), - /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI), - /* TL */ COMP_CODEC("i2c-10EC1011:02", CML_RT1011_CODEC_DAI), - /* TR */ COMP_CODEC("i2c-10EC1011:03", CML_RT1011_CODEC_DAI))); + /* WL */ COMP_CODEC("i2c-10EC1011:00", CML_RT1011_CODEC_DAI), + /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI))); SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); @@ -399,6 +489,7 @@ static struct snd_soc_dai_link cml_rt1011_rt5682_dailink[] = { .dpcm_playback = 1, .dpcm_capture = 1, /* Capture stream provides Feedback */ .no_pcm = 1, + .init = cml_rt1011_spk_init, .ops = &cml_rt1011_ops, SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), }, @@ -413,14 +504,6 @@ static struct snd_soc_codec_conf rt1011_conf[] = { .dlc = COMP_CODEC_CONF("i2c-10EC1011:01"), .name_prefix = "WR", }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:02"), - .name_prefix = "TL", - }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:03"), - .name_prefix = "TR", - }, }; /* Cometlake audio machine driver for RT1011 and RT5682 */ @@ -442,10 +525,12 @@ static struct snd_soc_card snd_soc_card_cml = { static int snd_cml_rt1011_probe(struct platform_device *pdev) { + struct snd_soc_dai_link_component *rt1011_dais_components; + struct snd_soc_codec_conf *rt1011_dais_confs; struct card_private *ctx; struct snd_soc_acpi_mach *mach; const char *platform_name; - int ret; + int ret, i; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); if (!ctx) @@ -456,6 +541,59 @@ static int snd_cml_rt1011_probe(struct platform_device *pdev) snd_soc_card_cml.dev = &pdev->dev; platform_name = mach->mach_params.platform; + dmi_check_system(sof_rt1011_quirk_table); + + dev_info(&pdev->dev, "sof_rt1011_quirk = %lx\n", sof_rt1011_quirk); + + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + rt1011_dais_confs = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_codec_conf) * + SPK_CH, GFP_KERNEL); + + rt1011_dais_components = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_dai_link_component) * + SPK_CH, GFP_KERNEL); + + for (i = 0; i < SPK_CH; i++) { + rt1011_dais_confs[i].dlc.name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1011:0%d", + i); + switch (i) { + case 0: + rt1011_dais_confs[i].name_prefix = "WL"; + break; + + case 1: + rt1011_dais_confs[i].name_prefix = "WR"; + break; + case 2: + rt1011_dais_confs[i].name_prefix = "TL"; + break; + case 3: + rt1011_dais_confs[i].name_prefix = "TR"; + break; + } + rt1011_dais_components[i].name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1011:0%d", + i); + rt1011_dais_components[i].dai_name = CML_RT1011_CODEC_DAI; + } + + snd_soc_card_cml.codec_conf = rt1011_dais_confs; + snd_soc_card_cml.num_configs = SPK_CH; + + for (i = 0; i < ARRAY_SIZE(cml_rt1011_rt5682_dailink); i++) { + if (!strcmp(cml_rt1011_rt5682_dailink[i].codecs->dai_name, + CML_RT1011_CODEC_DAI)) { + cml_rt1011_rt5682_dailink[i].codecs = rt1011_dais_components; + cml_rt1011_rt5682_dailink[i].num_codecs = SPK_CH; + } + } + } + /* set platform name for each dailink */ ret = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cml, platform_name); @@ -483,5 +621,6 @@ MODULE_DESCRIPTION("Cometlake Audio Machine driver - RT1011 and RT5682 in I2S mo MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:cml_rt1011_rt5682"); -- 2.7.4

3 2

[PATCH AUTOSEL 5.4 25/35] ASoC: stm32: sai: manage rebind issue
by Sasha Levin 16 Mar '20

16 Mar '20

From: Olivier Moysan <olivier.moysan(a)st.com> [ Upstream commit 0d6defc7e0e437a9fd53622f7fd85740f38d5693 ] 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 with STM32 SAI driver. 1) DMA channel is not requested: With the sound card rebind the simplified call sequence is: stm32_sai_sub_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 SAI driver, to change the call sequence as follows: stm32_sai_sub_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 SAI driver, to have the right sequence. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> Message-Id: <20200304102406.8093-1-olivier.moysan(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 30bcd5d3a32a8..10eb4b8e8e7ee 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1543,20 +1543,20 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(&pdev->dev, &stm32_component, - &sai->cpu_dai_drv, 1); + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register pcm dma\n"); + return ret; + } + + ret = snd_soc_register_component(&pdev->dev, &stm32_component, + &sai->cpu_dai_drv, 1); if (ret) return ret; if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) conf = &stm32_sai_pcm_config_spdif; - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, conf, 0); - if (ret) { - dev_err(&pdev->dev, "Could not register pcm dma\n"); - return ret; - } - return 0; } @@ -1565,6 +1565,8 @@ static int stm32_sai_sub_remove(struct platform_device *pdev) struct stm32_sai_sub_data *sai = dev_get_drvdata(&pdev->dev); clk_unprepare(sai->pdata->pclk); + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } -- 2.20.1

1 0

[PATCH AUTOSEL 5.5 29/41] ASoC: stm32: sai: manage rebind issue
by Sasha Levin 16 Mar '20

16 Mar '20

From: Olivier Moysan <olivier.moysan(a)st.com> [ Upstream commit 0d6defc7e0e437a9fd53622f7fd85740f38d5693 ] 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 with STM32 SAI driver. 1) DMA channel is not requested: With the sound card rebind the simplified call sequence is: stm32_sai_sub_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 SAI driver, to change the call sequence as follows: stm32_sai_sub_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 SAI driver, to have the right sequence. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> Message-Id: <20200304102406.8093-1-olivier.moysan(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 30bcd5d3a32a8..10eb4b8e8e7ee 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1543,20 +1543,20 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(&pdev->dev, &stm32_component, - &sai->cpu_dai_drv, 1); + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register pcm dma\n"); + return ret; + } + + ret = snd_soc_register_component(&pdev->dev, &stm32_component, + &sai->cpu_dai_drv, 1); if (ret) return ret; if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) conf = &stm32_sai_pcm_config_spdif; - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, conf, 0); - if (ret) { - dev_err(&pdev->dev, "Could not register pcm dma\n"); - return ret; - } - return 0; } @@ -1565,6 +1565,8 @@ static int stm32_sai_sub_remove(struct platform_device *pdev) struct stm32_sai_sub_data *sai = dev_get_drvdata(&pdev->dev); clk_unprepare(sai->pdata->pclk); + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } -- 2.20.1

1 0

[PATCH AUTOSEL 5.5 09/41] ASoC: meson: g12a: add tohdmitx reset
by Sasha Levin 16 Mar '20

16 Mar '20

From: Jerome Brunet <jbrunet(a)baylibre.com> [ Upstream commit 22946f37557e27697aabc8e4f62642bfe4a17fd8 ] Reset the g12a hdmi codec glue on probe. This ensure a sane startup state. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200221121146.1498427-1-jbrunet@baylibre.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/meson/g12a-tohdmitx.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c index 9cfbd343a00c8..8a0db28a6a406 100644 --- a/sound/soc/meson/g12a-tohdmitx.c +++ b/sound/soc/meson/g12a-tohdmitx.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <sound/pcm_params.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <sound/soc.h> #include <sound/soc-dai.h> @@ -378,6 +379,11 @@ static int g12a_tohdmitx_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; void __iomem *regs; struct regmap *map; + int ret; + + ret = device_reset(dev); + if (ret) + return ret; regs = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(regs)) -- 2.20.1

1 0

[PATCH v2 00/10] ASoC: Intel: machine driver updates for 5.7
by Pierre-Louis Bossart 15 Mar '20

15 Mar '20

Fix HDaudio/HDMI Kconfig dependencies, cleanup code, and add support for 2 new configs (DA7219+Maxim 98360A and RT5692+RT1015) Note that the last patch is the first case where we added the Reviewed-by tags automagically from the list of GitHub approvers. Feedback/comments welcome on the format, and thanks to Ranjani for the nice script! Changes since v1: Change base to broonie/for-5.7 Rebase and fix for_each_codec_dais() macro in Patch 9. Add new patch to remove GFP_ATOMIC Guennadi Liakhovetski (3): ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations ASoC: Intel: skylake: (cosmetic) remove redundant variable initialisations ASoC: Intel: (cosmetic) simplify structure member access Kai Vehmanen (4): ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC Pierre-Louis Bossart (1): ASoC: Intel: don't use GFP_ATOMIC for machine driver contexts Yong Zhi (2): ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp ASoC: Intel: sof_rt5682: Add rt1015 speaker amp support sound/soc/codecs/hdac_hdmi.c | 6 +- sound/soc/intel/boards/Kconfig | 20 ++-- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 2 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 4 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- .../soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 78 +++++++++++-- sound/soc/intel/boards/sof_pcm512x.c | 38 ++++-- sound/soc/intel/boards/sof_rt5682.c | 110 +++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 34 +++++- sound/soc/intel/skylake/skl.c | 4 +- sound/soc/sof/intel/Kconfig | 11 -- sound/soc/sof/intel/hda.c | 3 +- 26 files changed, 269 insertions(+), 71 deletions(-) base-commit: 101247a3b86e1cc0e382b7e887a56176290fc957 -- 2.20.1

3 14

[PATCH Fix for Kingston HyperX Amp (0951:16d8) 0/2 (Try #2)]
by crwulff＠gmail.com 14 Mar '20

14 Mar '20

From: Chris Wulff <crwulff(a)gmail.com> This patch set is modified to use an inline function for the mixer control interface lookup. The attempt at preventing modifications to a PCM that has previously been registered has been changed to defer registration instead, and has been split into a separate patch. This approach now works with pulseaudio.

2 3

Re: Re: [alsa-devel] [PATCH 4/7] ALSA: intel8x0: constify snd_pcm_ops structures
by sandra luna 14 Mar '20

14 Mar '20

1 0

[PATCH 00/16] SoundWire: cadence: add clock stop and fix programming sequences
by Pierre-Louis Bossart 14 Mar '20

14 Mar '20

To make progress with SoundWire support, this patchset provides the missing support for clock stop modes, and revisits all Cadence Master register settings. The current code is for some reason not aligned with internal documentation and hardware recommended flows, specifically for multi-link operation. Pierre-Louis Bossart (12): soundwire: cadence: s/update_config/config_update soundwire: cadence: handle error cases with CONFIG_UPDATE soundwire: cadence: mask Slave interrupt before stopping clock soundwire: cadence: merge routines to clear/set bits soundwire: cadence: move clock/SSP related inits to dedicated function soundwire: cadence: make SSP interval programmable soundwire: cadence: reorder MCP_CONFIG settings soundwire: cadence: enable NORMAL operation in cdns_init() soundwire: cadence: remove PREQ_DELAY assignment soundwire: cadence: remove automatic command retries soundwire: cadence: commit changes in the exit_reset() sequence soundwire: cadence: multi-link support Rander Wang (4): soundwire: cadence: simplifiy cdns_init() soundwire: cadence: add interface to check clock status soundwire: cadence: add clock_stop/restart routines soundwire: cadence: fix a io timeout issue in S3 test drivers/soundwire/cadence_master.c | 297 ++++++++++++++++++++++++----- drivers/soundwire/cadence_master.h | 9 +- drivers/soundwire/intel.c | 2 +- 3 files changed, 261 insertions(+), 47 deletions(-) base-commit: 5de79ba865d7770c3bdde7c266ed425832764aac -- 2.20.1

2 26

[alsa-devel] [PATCH 0/6] ASoC: soc-acpi: add support for SoundWire-based machines
by Pierre-Louis Bossart 14 Mar '20

14 Mar '20

SoundWire support on Intel platforms relies on ACPI _ADR fields exposed in DSDT tables, with a 64-bit value defining which link a Slave device is connected to and its partId/manufacturerID/uniqueID. The existing definitions based on a globally-visible _HID used for I2C/I2S are no longer sufficient and need to be extended to detect during the probe steps which machine driver should be selected. This patchset adds the structure changes, and a set of static tables already used by Intel developers and customers. These tables are provided ahead of time, but they will only be used once all SoundWire-related parts are merged upstream and SoundWire is enabled in the builds. For reference, the complete set of 100+ patches required for SoundWire on Intel platforms is available here: https://github.com/thesofproject/linux/pull/1692 Bard Liao (1): ASoC: Intel: common: add match tables for ICL w/ SoundWire Pierre-Louis Bossart (4): ASoC: soc-acpi: add _ADR-based link descriptors ASoC: Intel: common: soc-acpi: declare new tables for SoundWire ASoC: Intel: common: add match tables for TGL w/ SoundWire ASoC: SOF: Intel: reference SoundWire machine lists Rander Wang (1): ASoC: Intel: common: add match tables for CNL/CFL/CML w/ SoundWire include/sound/soc-acpi-intel-match.h | 6 ++ include/sound/soc-acpi.h | 21 ++++ .../intel/common/soc-acpi-intel-cfl-match.c | 5 + .../intel/common/soc-acpi-intel-cml-match.c | 89 +++++++++++++++++ .../intel/common/soc-acpi-intel-cnl-match.c | 5 + .../intel/common/soc-acpi-intel-icl-match.c | 98 +++++++++++++++++++ .../intel/common/soc-acpi-intel-tgl-match.c | 51 +++++++++- sound/soc/sof/sof-pci-dev.c | 5 + 8 files changed, 278 insertions(+), 2 deletions(-) base-commit: b2e2a13796889e10ba0390ab338949ba63d89462 -- 2.20.1

4 20

[PATCH 3/3] ALSA USB MIDI: Make amount of MIDI events per output URB selectable by user
by Andreas Steinmetz 14 Mar '20

14 Mar '20

This patch introduces a new module parameter "events" which allows userspace to limit the amount of MIDI events per output URB. An "events" value of 0 which is the default means "use wMaxPacketSize". Out of range values for "events" are clamped to the nearest valid value. It is assumed that the previous patches are applied and "outqueue=1" is used. Looking then at a four port USB MIDI interface with a wMaxPacketSize of 64 and assuming an optimal output rate of 320us/byte the following is true: 16 MIDI events PER URB Up to 3 bytes per MIDI event (3 bytes are assumed below) A driver queue of 1 URB which is completely filled. Resulting latency is 16*3*1*320us=15.36ms which still is in the audible range (a latency of less than 5ms is required, think playing a keyboard to MIDI in and using MIDI out to control a piano module and using headphones, while using another port for sysex transfers). After applying this patch and selecting the valid minimum event value of 4 (there must be at least space for one event per port to assert proper output balancing) tests show that the throughput isn't really affected, the worst case latency however drops to 3.84ms which is acceptable. In case of the largest class compliant devices which do have 16 ports the worst case latency is then 15.36ms which is tolerable. If this is still to long it could be fixed by introducing additional complexity to the balancing of snd_usbmidi_standard_output() at some later stage. As the device default is used by default no implications are expected for existing users. By making the parameter writable via sysfs it is possible to do selective output URB event count management from userspace via a script that manages USB authorize and the module option. Signed-off-by: Andreas Steinmetz <ast(a)domdv.de> --- a/sound/usb/midi.c 2020-03-13 20:18:33.443265194 +0100 +++ b/sound/usb/midi.c 2020-03-13 20:42:49.445546326 +0100 @@ -79,9 +79,12 @@ MODULE_LICENSE("Dual BSD/GPL"); static int outqueue = OUTPUT_URBS; +static int events; module_param(outqueue, int, 0644); -MODULE_PARM_DESC(outqueue, "Outgoing queue size, 1-7 (default: 7)."); +MODULE_PARM_DESC(outqueue, "Outgoing MIDI queue size, 1-7 (default: 7)."); +module_param(events, int, 0644); +MODULE_PARM_DESC(events, "MIDI events per queue element, 0=device default (default: 0)."); struct usb_ms_header_descriptor { @@ -1426,6 +1429,17 @@ switch (umidi->usb_id) { default: ep->max_transfer = usb_maxpacket(umidi->dev, pipe, 1); + if (events) { + int ev = 0; + + for (i = 0; i < 0x10; ++i) + if (ep_info->out_cables & (1 << i)) + ev++; + if (events > ev) + ev = events; + if (ev < ep->max_transfer >> 2) + ep->max_transfer = ev << 2; + } break; /* * Various chips declare a packet size larger than 4 bytes, but

1 0

[PATCH 2/3] ALSA USB MIDI: Make amount of output URBs selectable by user
by Andreas Steinmetz 14 Mar '20

14 Mar '20

This patch introduces a new module parameter "outqueue" which allows userspace to set the amount of output URBs is a range from 1 to 7 with 7 being the default (illegal values are clamped to the next valid value). It is assumed that the previous patch has been applied. Looking then at a multiport USB MIDI interface with a wMaxPacketSize of 64 and assuming an optimal output rate of 320us/byte the following is true: 16 MIDI events PER URB Up to 3 bytes per MIDI event (3 bytes are assumed below) A driver queue of 7 URBs which are completely filled. Resulting latency is 16*3*7*320us=107.52ms So if all URBs are filled by e.g. a sysex transfer on one port, a realtime message delivered by userspace for another port will be delayed by 107.52ms which is far too long. After applying this patch and selecting a queue of only 1 URB tests show that the throughput isn't really affected, the latency however drops to 16*3*1*320us=15.36ms which is considerably better. As the default is not touched no implications are expected for existing users. By making the parameter writable via sysfs it is possible to do selective output URB queue size management from userspace via a script that manages USB authorize and the module option. Signed-off-by: Andreas Steinmetz <ast(a)domdv.de> --- a/sound/usb/midi.c 2020-03-13 19:19:28.614798593 +0100 +++ b/sound/usb/midi.c 2020-03-13 19:37:08.097411436 +0100 @@ -45,6 +45,7 @@ #include <linux/slab.h> #include <linux/timer.h> #include <linux/usb.h> +#include <linux/moduleparam.h> #include <linux/wait.h> #include <linux/usb/audio.h> #include <linux/module.h> @@ -77,6 +78,11 @@ MODULE_DESCRIPTION("USB Audio/MIDI helper module"); MODULE_LICENSE("Dual BSD/GPL"); +static int outqueue = OUTPUT_URBS; + +module_param(outqueue, int, 0644); +MODULE_PARM_DESC(outqueue, "Outgoing queue size, 1-7 (default: 7)."); + struct usb_ms_header_descriptor { __u8 bLength; @@ -141,6 +147,7 @@ } urbs[OUTPUT_URBS]; unsigned int active_urbs; unsigned int drain_urbs; + unsigned int out_urbs; int max_transfer; /* size of urb buffer */ struct tasklet_struct tasklet; unsigned int next_urb; @@ -335,7 +342,7 @@ break; ep->active_urbs |= 1 << urb_index; } - if (++urb_index >= OUTPUT_URBS) + if (++urb_index >= ep->out_urbs) urb_index = 0; if (urb_index == ep->next_urb) break; @@ -1364,7 +1371,7 @@ { unsigned int i; - for (i = 0; i < OUTPUT_URBS; ++i) + for (i = 0; i < ep->out_urbs; ++i) if (ep->urbs[i].urb) { free_urb_and_buffer(ep->umidi, ep->urbs[i].urb, ep->max_transfer); @@ -1397,7 +1404,14 @@ return -ENOMEM; ep->umidi = umidi; - for (i = 0; i < OUTPUT_URBS; ++i) { + if (outqueue < 1) + ep->out_urbs = 1; + else if (outqueue > OUTPUT_URBS) + ep->out_urbs = OUTPUT_URBS; + else + ep->out_urbs = outqueue; + + for (i = 0; i < ep->out_urbs; ++i) { ep->urbs[i].urb = usb_alloc_urb(0, GFP_KERNEL); if (!ep->urbs[i].urb) { err = -ENOMEM; @@ -1434,7 +1448,7 @@ ep->max_transfer = 9; break; } - for (i = 0; i < OUTPUT_URBS; ++i) { + for (i = 0; i < ep->out_urbs; ++i) { buffer = usb_alloc_coherent(umidi->dev, ep->max_transfer, GFP_KERNEL, &ep->urbs[i].urb->transfer_dma); @@ -1525,7 +1539,7 @@ if (ep->out) tasklet_kill(&ep->out->tasklet); if (ep->out) { - for (j = 0; j < OUTPUT_URBS; ++j) + for (j = 0; j < ep->out->out_urbs; ++j) usb_kill_urb(ep->out->urbs[j].urb); if (umidi->usb_protocol_ops->finish_out_endpoint) umidi->usb_protocol_ops->finish_out_endpoint(ep->out);

1 0

BUG: unable to handle kernel NULL pointer dereference in corrupted (5)
by syzbot 14 Mar '20

14 Mar '20

Hello, syzbot found the following crash on: HEAD commit: 30bb5572 Merge tag 'ktest-v5.6' of git://git.kernel.org/pu.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=15d300b1e00000 kernel config: https://syzkaller.appspot.com/x/.config?x=a5295e161cd85b82 dashboard link: https://syzkaller.appspot.com/bug?extid=8b0e78e390d1715b0f4e compiler: clang version 10.0.0 (https://github.com/llvm/llvm-project/ c2443155a0fb245c8f17f2c1c72b6ea391e86e81) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=11a57709e00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=13de9e91e00000 The bug was bisected to: commit 271213ef4d0d3a3b80d4cf95c5f2bebb5643e666 Author: Takashi Iwai <tiwai(a)suse.de> Date: Tue Dec 10 06:34:50 2019 +0000 ALSA: pcxhr: Support PCM sync_stop bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=1566b08de00000 final crash: https://syzkaller.appspot.com/x/report.txt?x=1766b08de00000 console output: https://syzkaller.appspot.com/x/log.txt?x=1366b08de00000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+8b0e78e390d1715b0f4e(a)syzkaller.appspotmail.com Fixes: 271213ef4d0d ("ALSA: pcxhr: Support PCM sync_stop") BUG: kernel NULL pointer dereference, address: 0000000000000086 #PF: supervisor instruction fetch in kernel mode #PF: error_code(0x0010) - not-present page PGD a27fa067 P4D a27fa067 PUD a2185067 PMD 0 Oops: 0010 [#1] PREEMPT SMP KASAN CPU: 1 PID: 9168 Comm: syz-executor418 Not tainted 5.6.0-rc5-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:0x86 Code: Bad RIP value. RSP: 0018:ffffc90002c9f998 EFLAGS: 00010086 RAX: ffffc90002c9f9c8 RBX: fffffe0000000000 RCX: ffff8880a31b6280 RDX: 0000000000000000 RSI: 0000000000000001 RDI: 0000000000000000 RBP: 0000000000000ec0 R08: ffffffff839888d3 R09: ffffffff811c7d9a R10: ffff8880a31b6280 R11: 0000000000000002 R12: dffffc0000000000 R13: fffffe0000000ec8 R14: ffffffff880016f0 R15: fffffe0000000ecb FS: 0000000000981880(0000) GS:ffff8880ae900000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 000000000000005c CR3: 00000000986d2000 CR4: 00000000001426e0 Call Trace: Modules linked in: CR2: 0000000000000086 ---[ end trace 7c78bc94cfc0a37d ]--- RIP: 0010:0x86 Code: Bad RIP value. RSP: 0018:ffffc90002c9f998 EFLAGS: 00010086 RAX: ffffc90002c9f9c8 RBX: fffffe0000000000 RCX: ffff8880a31b6280 RDX: 0000000000000000 RSI: 0000000000000001 RDI: 0000000000000000 RBP: 0000000000000ec0 R08: ffffffff839888d3 R09: ffffffff811c7d9a R10: ffff8880a31b6280 R11: 0000000000000002 R12: dffffc0000000000 R13: fffffe0000000ec8 R14: ffffffff880016f0 R15: fffffe0000000ecb FS: 0000000000981880(0000) GS:ffff8880ae900000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 000000000000005c CR3: 00000000986d2000 CR4: 00000000001426e0 --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. For information about bisection process see: https://goo.gl/tpsmEJ#bisection syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

3 2

[PATCH v2] ASoC: dt-bindings: google, cros-ec-codec: Fix dtc warnings in example
by Rob Herring 13 Mar '20

13 Mar '20

Extra dtc warnings (roughly what W=1 enables) are now enabled by default when building the binding examples. These were fixed treewide in 5.6-rc5, but the newly added google,cros-ec-codec schema adds some new warnings: Documentation/devicetree/bindings/sound/google,cros-ec-codec.example.dts:17.28-21.11: Warning (unit_address_vs_reg): /example-0/reserved_mem: node has a reg or ranges property, but no unit name Documentation/devicetree/bindings/sound/google,cros-ec-codec.example.dts:22.19-32.11: Warning (unit_address_vs_reg): /example-0/cros-ec@0: node has a unit name, but no reg property Documentation/devicetree/bindings/sound/google,cros-ec-codec.example.dts:26.37-31.15: Warning (unit_address_vs_reg): /example-0/cros-ec@0/ec-codec: node has a reg or ranges property, but no unit name Fixing the above, then results in: Documentation/devicetree/bindings/sound/google,cros-ec-codec.example.dts:26.13-23: Warning (reg_format): /example-0/cros-ec@0:reg: property has invalid length (4 bytes) (#address-cells == 1, #size-cells == 1) Documentation/devicetree/bindings/sound/google,cros-ec-codec.example.dts:27.37-32.15: Warning (unit_address_vs_reg): /example-0/cros-ec@0/ec-codec: node has a reg or ranges property, but no unit name Fix these warnings and adjust the node name to 'audio-codec' while we're touching the node name. Fixes: eadd54c75f1e ("dt-bindings: Convert the binding file google, cros-ec-codec.txt to yaml format.") Cc: Cheng-Yi Chiang <cychiang(a)chromium.org> Cc: Guenter Roeck <groeck(a)chromium.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Benson Leung <bleung(a)chromium.org> Cc: alsa-devel(a)alsa-project.org Reviewed-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> Signed-off-by: Rob Herring <robh(a)kernel.org> --- v2: - s/ec-codec/audio-codec/ --- .../bindings/sound/google,cros-ec-codec.yaml | 27 +++++++++++-------- 1 file changed, 16 insertions(+), 11 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml index 94a85d0cbf43..d62284288f23 100644 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -44,19 +44,24 @@ additionalProperties: false examples: - | - reserved_mem: reserved_mem { + reserved_mem: reserved-mem@52800000 { compatible = "shared-dma-pool"; - reg = <0 0x52800000 0 0x100000>; + reg = <0x52800000 0x100000>; no-map; }; - cros-ec@0 { - compatible = "google,cros-ec-spi"; - #address-cells = <2>; - #size-cells = <1>; - cros_ec_codec: ec-codec { - compatible = "google,cros-ec-codec"; - #sound-dai-cells = <1>; - reg = <0x0 0x10500000 0x80000>; - memory-region = <&reserved_mem>; + spi { + #address-cells = <1>; + #size-cells = <0>; + cros-ec@0 { + compatible = "google,cros-ec-spi"; + #address-cells = <2>; + #size-cells = <1>; + reg = <0>; + cros_ec_codec: audio-codec@10500000 { + compatible = "google,cros-ec-codec"; + #sound-dai-cells = <1>; + reg = <0x0 0x10500000 0x80000>; + memory-region = <&reserved_mem>; + }; }; }; -- 2.20.1

2 2

[PATCH] ASoC: max98357a: Add ACPI HID MAX98360A
by Yong Zhi 13 Mar '20

13 Mar '20

Maxim MAX98360A audio amplifier is functionally identical to MAX98357A, add ACPI ID "MAX98360A" for driver reuse. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/codecs/max98357a.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 74f20114297c..a8bd793a7867 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -133,6 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id); #ifdef CONFIG_ACPI static const struct acpi_device_id max98357a_acpi_match[] = { { "MX98357A", 0 }, + { "MX98360A", 0 }, {}, }; MODULE_DEVICE_TABLE(acpi, max98357a_acpi_match); -- 2.7.4

2 1

alsatplg (libasound.a) segmentation fault using AFL
by GitHub issues - opened 13 Mar '20

13 Mar '20

alsa-project/alsa-lib issue #37 was opened from tysonite: I was playing around with [AFL](https://fuzzing-project.org/tutorial3.html) tonight on one of my pet projects. And after it found few crashes, I've decided to fuzz one of open-source projects. The `alsatplg` tool just looked simple enough to exercise it with fuzzing tool. I made a simple Dockerfile that runs AFL on `alsatplg`: ``` FROM ubuntu:18.04 ENV LANG C.UTF-8 RUN apt-get update && \ apt-get install -y apt-utils && \ apt-get install -y afl git build-essential m4 autoconf automake libtool RUN cd / RUN git clone https://github.com/alsa-project/alsa-lib.git RUN cd alsa-lib && \ libtoolize --force --copy --automake && \ aclocal && \ autoheader && \ automake --foreign --copy --add-missing && \ autoconf && \ export CFLAGS="-O2 -Wall -W -Wunused-const-variable=0 -pipe -g" && \ export CC=afl-gcc && \ ./configure --disable-aload && \ make && \ make install \ && cd / RUN apt-get install -y gettext ncurses-base libncurses5 libncurses5-dev pkg-config RUN git clone https://github.com/alsa-project/alsa-utils.git RUN cd alsa-utils && \ export CC=afl-gcc && \ ./gitcompile && \ make install && \ cd / RUN mkdir in #RUN cp alsa-utils/speaker-test/samples/Noise.wav in RUN echo "Hello" > in/input.txt CMD ["afl-fuzz", "-i", "in", "-o", "out", "alsatplg", "-c", "@@", "-o", "/output"] ``` After around 10-15 minutes running on my core i7 laptop, it generated a sequence of bytes that leads to crash. If you want to try it by yourself just run `docker build -t alsa/dev .` followed by `docker run alsa/dev`, and wait a bit. When crash happened, the input data can be copied from the container by running `docker cp <container_id>:/out .`. An example of input data that lead to SIGSEGV: [id:000000,sig:11,src:000325,op:arith8,pos:48,val:-26.txt](https://github.co… And stack trace based on it: ``` Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `alsatplg -c out/crashes/id:000000,sig:11,src:000325,op:arith8,pos:48,val:-26 -o'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007fcb65e05ca8 in snd_config_delete () from /usr/lib/x86_64-linux-gnu/libasound.so.2 (gdb) bt #0 0x00007fcb65e05ca8 in snd_config_delete () from /usr/lib/x86_64-linux-gnu/libasound.so.2 #1 0x00007fcb65e06479 in ?? () from /usr/lib/x86_64-linux-gnu/libasound.so.2 #2 0x00007fcb65e064ba in ?? () from /usr/lib/x86_64-linux-gnu/libasound.so.2 #3 0x00007fcb65e0661c in ?? () from /usr/lib/x86_64-linux-gnu/libasound.so.2 #4 0x00007fcb65e818c4 in snd_tplg_build_file () from /usr/lib/x86_64-linux-gnu/libasound.so.2 #5 0x00005587bce0ab6a in ?? () #6 0x00007fcb65a07b97 in __libc_start_main (main=0x5587bce0aa10, argc=5, argv=0x7ffcfa707628, init=<optimized out>, fini=<optimized out>, rtld_fini=<optimized out>, stack_end=0x7ffcfa707618) at ../csu/libc-start.c:310 #7 0x00005587bce0ac4a in ?? () (gdb) bt full #0 0x00007fcb65e05ca8 in snd_config_delete () from /usr/lib/x86_64-linux-gnu/libasound.so.2 No symbol table info available. #1 0x00007fcb65e06479 in ?? () from /usr/lib/x86_64-linux-gnu/libasound.so.2 No symbol table info available. #2 0x00007fcb65e064ba in ?? () from /usr/lib/x86_64-linux-gnu/libasound.so.2 No symbol table info available. #3 0x00007fcb65e0661c in ?? () from /usr/lib/x86_64-linux-gnu/libasound.so.2 No symbol table info available. #4 0x00007fcb65e818c4 in snd_tplg_build_file () from /usr/lib/x86_64-linux-gnu/libasound.so.2 No symbol table info available. #5 0x00005587bce0ab6a in ?? () No symbol table info available. #6 0x00007fcb65a07b97 in __libc_start_main (main=0x5587bce0aa10, argc=5, argv=0x7ffcfa707628, init=<optimized out>, fini=<optimized out>, rtld_fini=<optimized out>, stack_end=0x7ffcfa707618) at ../csu/libc-start.c:310 self = <optimized out> __self = <optimized out> result = <optimized out> unwind_buf = {cancel_jmp_buf = {{jmp_buf = {0, -5452963434713232627, 94041477786656, 140724510160416, 0, 0, -2259219850243519731, -2248813385476519155}, mask_was_saved = 0}}, priv = {pad = {0x0, 0x0, 0x7fcb660ee733 <_dl_init+259>, 0x7fcb660d6370}, data = {prev = 0x0, cleanup = 0x0, canceltype = 1712252723}}} not_first_call = <optimized out> #7 0x00005587bce0ac4a in ?? () No symbol table info available. ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/37 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/6] ASoC: SOF: multi-cpu dais, IPC and Intel improvements for 5.7
by Pierre-Louis Bossart 13 Mar '20

13 Mar '20

Corrections for compatibility with the multi-cpu dai check for min-channels, IPC simplifications, and removal of unnecessary boot-related code for Intel platforms. Amery Song (2): ASoC: SOF: Intel: hda: remove unnecessary ROM IPC filter function ASoC: SOF: Intel: remove unnecessary waitq before loading firmware Bard Liao (2): ASoC: SOF: Intel: hda-dai: add stream capability ASoC: SOF: topology: connect dai widget to all cpu-dais Karol Trzcinski (2): ASoC: SOF: Make sof_ipc_ext_data enum more rigid ASoC: SOF: Remove SOF_IPC_EXT_DMA_BUFFER include/sound/sof/info.h | 22 +------- sound/soc/sof/intel/cnl.c | 5 -- sound/soc/sof/intel/hda-dai.c | 96 ++++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda-ipc.c | 20 +------ sound/soc/sof/intel/hda-loader.c | 3 - sound/soc/sof/intel/hda.c | 3 - sound/soc/sof/intel/hda.h | 1 - sound/soc/sof/loader.c | 6 -- sound/soc/sof/sof-priv.h | 5 -- sound/soc/sof/topology.c | 8 ++- 10 files changed, 108 insertions(+), 61 deletions(-) base-commit: 101247a3b86e1cc0e382b7e887a56176290fc957 -- 2.20.1

2 13

[PATCH 00/10] ASoC: SOF: Intel: add SoundWire support
by Pierre-Louis Bossart 13 Mar '20

13 Mar '20

This patchset provides the support for SoundWire support on Intel CometLake, IcelLake and TigerLake RVP platforms and form-factor devices to be released 'soon'. The bulk of the code is about detecting a valid SoundWire configuration from ACPI, and implementing the interfaces suggested in '[PATCH 0/8] soundwire: remove platform devices, add SOF interfaces' for interrupts, PCI wakes and clock-stop configurations. Since that SoundWire series is stuck with no resolution, the build support for SOF w/ SoundWire is not provided for now, and fall-back functions will be used. This code is tested on a daily basis in the SOF tree and is not expected to change. If audio maintainers will only accept functional code, which isn't unreasonable, I would kindly ask that they reach out to Vinod Koul. Bard Liao (1): ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers Pierre-Louis Bossart (7): ASoC: soc-acpi: expand description of _ADR-based devices ASoC: SOF: Intel: add SoundWire configuration interface ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks Rander Wang (2): ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread include/sound/soc-acpi.h | 39 +- include/sound/sof/dai-intel.h | 18 +- .../intel/common/soc-acpi-intel-cml-match.c | 87 +++- .../intel/common/soc-acpi-intel-icl-match.c | 97 ++++- .../intel/common/soc-acpi-intel-tgl-match.c | 49 ++- sound/soc/sof/intel/hda-dsp.c | 2 + sound/soc/sof/intel/hda-loader.c | 31 ++ sound/soc/sof/intel/hda.c | 399 ++++++++++++++++++ sound/soc/sof/intel/hda.h | 66 +++ 9 files changed, 728 insertions(+), 60 deletions(-) base-commit: 101247a3b86e1cc0e382b7e887a56176290fc957 -- 2.20.1

2 12

Applied "ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI" to the asoc tree
by Mark Brown 13 Mar '20

13 Mar '20

The patch ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI 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 5c82813ce43e3a2017c43b32a57dc7a8802d9ad4 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Date: Thu, 12 Mar 2020 14:48:50 -0500 Subject: [PATCH] ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI Having a reverse dependency to a config that has its own additional dependencies, is generally not recommended. And this applies to select statements for SND_HDA_CODEC_HDMI, e.g. the case where SND_HDA and SND_SOC_SOF_HDA are built as modules, but the machine driver is built-in, leading to compile errors (reported as i386-randconfig-e003-20200206). Give up on trying to define different dependencies based on SOF/SST selection, and simply add a "depends on" for SND_HDA_CODEC_HDMI. This fixes the issue with randconfigs. Only downside is that SND_HDA_CODEC_HDMI may be built unnecessarily in some cases, but this seems like the lesser evil. Fixes: aa2b4a5 ('ASoC: Intel: boards: fix incorrect HDMI Kconfig dependency') Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200312194859.4051-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/Kconfig | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 755e1de19df9..67d85a7be559 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -289,7 +289,6 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON @@ -302,6 +301,7 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH tristate "Broxton with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON help This adds support for ASoC machine driver for Broxton-P platforms @@ -402,6 +402,7 @@ config SND_SOC_INTEL_GLK_DA7219_MAX98357A_MACH tristate "GLK with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON help This adds support for ASoC machine driver for Geminilake platforms @@ -413,10 +414,10 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH tristate "GLK with RT5682 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_RT5682 select SND_SOC_MAX98357A select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI help This adds support for ASoC machine driver for Geminilake platforms @@ -430,7 +431,7 @@ if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH tristate "SKL/KBL/BXT/APL with HDA Codecs" - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC + depends on SND_HDA_CODEC_HDMI select SND_SOC_HDAC_HDMI select SND_SOC_DMIC # SND_SOC_HDAC_HDA is already selected @@ -448,9 +449,9 @@ config SND_SOC_INTEL_SOF_RT5682_MACH depends on I2C && ACPI depends on (SND_SOC_SOF_HDA_LINK && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) + depends on SND_HDA_CODEC_HDMI select SND_SOC_RT5682 select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI help This adds support for ASoC machine driver for SOF platforms @@ -490,11 +491,11 @@ config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH tristate "CML with RT1011 and RT5682 in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_RT1011 select SND_SOC_RT5682 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC help This adds support for ASoC machine driver for SOF platform with RT1011 + RT5682 I2S codec. @@ -509,10 +510,10 @@ config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH tristate "SOF with DA7219 and MAX98373 in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_DA7219 select SND_SOC_MAX98373 select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC help This adds support for ASoC machine driver for SOF platforms with DA7219 + MAX98373 I2S audio codec. -- 2.20.1

1 0

Applied "ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI" to the asoc tree
by Mark Brown 13 Mar '20

13 Mar '20

The patch ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI 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 4399afd21a0169a125a16c2dbba68f7d9a2cffdb Mon Sep 17 00:00:00 2001 From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Date: Thu, 12 Mar 2020 14:48:51 -0500 Subject: [PATCH] ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI Having a reverse dependency to a config that has its own additional dependencies, is generally not recommended. And this applies to select statements for SND_HDA_CODEC_HDMI, e.g. the case where SND_HDA and SND_SOC_SOF_HDA are built as modules, but the machine driver is built-in, leading to compile errors (reported as i386-randconfig-e003-20200206). Give up on trying to define different dependencies based on SOF/SST selection, and simply add a "depends on" for SND_HDA_CODEC_HDMI. This fixes the issue with randconfigs. Only downside is that SND_HDA_CODEC_HDMI may be built unnecessarily in some cases, but this seems like the lesser evil. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200312194859.4051-3-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 67d85a7be559..6833ef548710 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -464,8 +464,8 @@ config SND_SOC_INTEL_SOF_PCM512x_MACH depends on I2C && ACPI depends on (SND_SOC_SOF_HDA_AUDIO_CODEC && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) + depends on SND_HDA_CODEC_HDMI select SND_SOC_PCM512x_I2C - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC help This adds support for ASoC machine driver for SOF platforms with TI PCM512x I2S audio codec. -- 2.20.1

1 0

Applied "ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms" to the asoc tree
by Mark Brown 13 Mar '20

13 Mar '20

The patch ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms 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 15a5a89597e57e67d4dde1d57fa85105c5930bb3 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Date: Thu, 12 Mar 2020 14:48:52 -0500 Subject: [PATCH] ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms Make HDMI optional for APL and later platforms. If no HDMI codec is found on the HDA bus, the graphics side driver is missing or correct codec driver is not part of kernel build, codec_mask reflects this and HDMI is disabled. The DSP topology will still have the links for HDMI, so connect these to dummy codec to avoid failures in topology loading. This change also fixes a kernel oops that was triggered if sof_pcm512x was used with SOF configured to use hdac-hdmi (can be done via "use_common_hdmi=0" or by selecting CONFIG_SND_SOC_SOF_HDA_COMMON_HDMI_CODEC=n). This is not a supported configuration. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200312194859.4051-4-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/sof_pcm512x.c | 38 +++++++++++++++++++++------- 1 file changed, 29 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c index 626153bd71e7..4ce707b6eb79 100644 --- a/sound/soc/intel/boards/sof_pcm512x.c +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -27,6 +27,8 @@ #define SOF_PCM512X_SSP_CODEC(quirk) ((quirk) & GENMASK(3, 0)) #define SOF_PCM512X_SSP_CODEC_MASK (GENMASK(3, 0)) +#define IDISP_CODEC_MASK 0x4 + /* Default: SSP5 */ static unsigned long sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(5); @@ -40,6 +42,7 @@ struct sof_hdmi_pcm { struct sof_card_private { struct list_head hdmi_pcm_list; + bool idisp_codec; }; static int sof_pcm512x_quirk_cb(const struct dmi_system_id *id) @@ -136,6 +139,9 @@ static int sof_card_late_probe(struct snd_soc_card *card) if (list_empty(&ctx->hdmi_pcm_list)) return -EINVAL; + if (!ctx->idisp_codec) + return 0; + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head); return hda_dsp_hdmi_build_controls(card, pcm->codec_dai->component); @@ -214,7 +220,8 @@ SND_SOC_DAILINK_DEF(dmic_component, static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int dmic_be_num, - int hdmi_num) + int hdmi_num, + bool idisp_codec) { struct snd_soc_dai_link_component *idisp_components; struct snd_soc_dai_link_component *cpus; @@ -316,11 +323,19 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, if (!links[id].cpus->dai_name) goto devm_err; - idisp_components[i - 1].name = "ehdaudio0D2"; - idisp_components[i - 1].dai_name = devm_kasprintf(dev, - GFP_KERNEL, - "intel-hdmi-hifi%d", - i); + /* + * topology cannot be loaded if codec is missing, so + * use the dummy codec if needed + */ + if (idisp_codec) { + idisp_components[i - 1].name = "ehdaudio0D2"; + idisp_components[i - 1].dai_name = + devm_kasprintf(dev, GFP_KERNEL, + "intel-hdmi-hifi%d", i); + } else { + idisp_components[i - 1].name = "snd-soc-dummy"; + idisp_components[i - 1].dai_name = "snd-soc-dummy-dai"; + } if (!idisp_components[i - 1].dai_name) goto devm_err; @@ -341,8 +356,8 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, static int sof_audio_probe(struct platform_device *pdev) { + struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct snd_soc_dai_link *dai_links; - struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; int ret, ssp_codec; @@ -360,6 +375,11 @@ static int sof_audio_probe(struct platform_device *pdev) } else { dmic_be_num = 2; #if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) + if (mach->mach_params.common_hdmi_codec_drv && + (mach->mach_params.codec_mask & IDISP_CODEC_MASK)) + ctx->idisp_codec = true; + + /* links are always present in topology */ hdmi_num = 3; #endif } @@ -374,7 +394,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_audio_card_pcm512x.num_links = 1 + dmic_be_num + hdmi_num; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, - dmic_be_num, hdmi_num); + dmic_be_num, hdmi_num, + ctx->idisp_codec); if (!dai_links) return -ENOMEM; @@ -383,7 +404,6 @@ static int sof_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); sof_audio_card_pcm512x.dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; /* set platform name for each dailink */ ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_pcm512x, -- 2.20.1

1 0