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

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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

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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

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[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

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[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

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[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

Applied "ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC" to the asoc tree
by Mark Brown 13 Mar '20

13 Mar '20

The patch ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC 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 42c67753cae12f68b936901792ecd367e8dfca9b Mon Sep 17 00:00:00 2001 From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Date: Thu, 12 Mar 2020 14:48:53 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC To help user-space with HDMI codec driver transition, both a kernel module parameter and a kernel option were initially provided to configure default behaviour of SOF on Intel hardware with commit 139c7febad1a ("ASoC: SOF: Intel: add support for snd-hda-codec-hdmi"). As hdac-hdmi is already now lagging in features compared to snd-hda-codec-hdmi, move ahead with the transition and remove the build option to select between the two, and instead default to snd-hda-codec-hdmi if it is enabled in kernel build. The old behaviour of using hdac-hdmi driver can still be forced via the kernel module parameter. 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/20200312194859.4051-5-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/Kconfig | 11 ----------- sound/soc/sof/intel/hda.c | 3 +-- 2 files changed, 1 insertion(+), 13 deletions(-) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 3bc64dee7c39..c9a2bee4b55c 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -324,17 +324,6 @@ config SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1 Say Y if you want to enable DMI Link L1 If unsure, select "N". -config SND_SOC_SOF_HDA_COMMON_HDMI_CODEC - bool "SOF common HDA HDMI codec driver" - depends on SND_SOC_SOF_HDA_LINK - depends on SND_HDA_CODEC_HDMI - default SND_HDA_CODEC_HDMI - help - This adds support for HDMI audio by using the common HDA - HDMI/DisplayPort codec driver. - Say Y if you want to use the common codec driver with SOF. - If unsure select "Y". - endif ## SND_SOC_SOF_HDA_COMMON config SND_SOC_SOF_HDA_LINK_BASELINE diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 7ca887041a34..1de750a1dd19 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -54,8 +54,7 @@ static int hda_dmic_num = -1; module_param_named(dmic_num, hda_dmic_num, int, 0444); MODULE_PARM_DESC(dmic_num, "SOF HDA DMIC number"); -static bool hda_codec_use_common_hdmi = - IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_COMMON_HDMI_CODEC); +static bool hda_codec_use_common_hdmi = IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI); module_param_named(use_common_hdmi, hda_codec_use_common_hdmi, bool, 0444); MODULE_PARM_DESC(use_common_hdmi, "SOF HDA use common HDMI codec driver"); #endif -- 2.20.1

1 0

Applied "ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations" to the asoc tree
by Mark Brown 13 Mar '20

13 Mar '20

The patch ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations 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 c4aafb337d311556d4448f8de857ca2683f5ca5b Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Date: Thu, 12 Mar 2020 14:48:54 -0500 Subject: [PATCH] ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations Remove several redundant variable initialisations. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200312194859.4051-6-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/hdac_hdmi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index e6558475e006..fba9b749839d 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1998,11 +1998,11 @@ static struct hdac_hdmi_drv_data intel_drv_data = { static int hdac_hdmi_dev_probe(struct hdac_device *hdev) { - struct hdac_hdmi_priv *hdmi_priv = NULL; + struct hdac_hdmi_priv *hdmi_priv; struct snd_soc_dai_driver *hdmi_dais = NULL; - struct hdac_ext_link *hlink = NULL; + struct hdac_ext_link *hlink; int num_dais = 0; - int ret = 0; + int ret; struct hdac_driver *hdrv = drv_to_hdac_driver(hdev->dev.driver); const struct hda_device_id *hdac_id = hdac_get_device_id(hdev, hdrv); -- 2.20.1

1 0

Applied "ASoC: Intel: skylake: (cosmetic) remove redundant variable initialisations" to the asoc tree
by Mark Brown 13 Mar '20

13 Mar '20

The patch ASoC: Intel: skylake: (cosmetic) remove redundant variable initialisations 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 ca841843a3a8038494e48968c2fd1c7ec5473ce3 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Date: Thu, 12 Mar 2020 14:48:55 -0500 Subject: [PATCH] ASoC: Intel: skylake: (cosmetic) remove redundant variable initialisations Variables, used as loop iterators, don't need to be initialised. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200312194859.4051-7-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index f755ca2484cf..1aa8114a4f77 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -359,7 +359,7 @@ static int skl_resume(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct hdac_bus *bus = pci_get_drvdata(pci); struct skl_dev *skl = bus_to_skl(bus); - struct hdac_ext_link *hlink = NULL; + struct hdac_ext_link *hlink; int ret; /* @@ -794,7 +794,7 @@ static void skl_probe_work(struct work_struct *work) { struct skl_dev *skl = container_of(work, struct skl_dev, probe_work); struct hdac_bus *bus = skl_to_bus(skl); - struct hdac_ext_link *hlink = NULL; + struct hdac_ext_link *hlink; int err; if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { -- 2.20.1

1 0

Applied "ASoC: Intel: (cosmetic) simplify structure member access" to the asoc tree
by Mark Brown 13 Mar '20

13 Mar '20

The patch ASoC: Intel: (cosmetic) simplify structure member access 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 42432196cfb01500ec058e8acc8dcfcf27eb76c9 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Date: Thu, 12 Mar 2020 14:48:56 -0500 Subject: [PATCH] ASoC: Intel: (cosmetic) simplify structure member access Fix a clumsy structure member dereference in all machine drivers. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200312194859.4051-8-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- 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 | 2 +- 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 +- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/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 | 2 +- sound/soc/intel/boards/sof_rt5682.c | 2 +- 19 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c index 1a302436d450..058abf3eec50 100644 --- a/sound/soc/intel/boards/bdw-rt5650.c +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -298,7 +298,7 @@ static int bdw_rt5650_probe(struct platform_device *pdev) return -ENOMEM; /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5650_card, mach->mach_params.platform); diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index bb643c99069d..a94f498388e1 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -412,7 +412,7 @@ static int bdw_rt5677_probe(struct platform_device *pdev) } /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5677_card, mach->mach_params.platform); if (ret) diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index b9c12e24c70b..25178000c6a5 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -283,7 +283,7 @@ static int broadwell_audio_probe(struct platform_device *pdev) broadwell_rt286.dev = &pdev->dev; /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(&broadwell_rt286, mach->mach_params.platform); if (ret) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 9177401c37a5..061462248bce 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -721,7 +721,7 @@ static int broxton_audio_probe(struct platform_device *pdev) } /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; platform_name = mach->mach_params.platform; ret = snd_soc_fixup_dai_links_platform_name(&broxton_audio_card, diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 4b67f261377c..4b5e7f6dbdf1 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -627,7 +627,7 @@ static int broxton_audio_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, ctx); /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; platform_name = mach->mach_params.platform; ret = snd_soc_fixup_dai_links_platform_name(card, diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index eda7a500cad6..d6b912c013fc 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -231,7 +231,7 @@ static int bytcht_da7213_probe(struct platform_device *pdev) int ret_val = 0; int i; - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; card = &bytcht_da7213_card; card->dev = &pdev->dev; diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 70bb86f3342f..ea119d523926 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -553,7 +553,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* override plaform name, if required */ snd_soc_card_cht.dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; platform_name = mach->mach_params.platform; ret_val = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cht, diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c index 501bad3976fb..34d4e17e3295 100644 --- a/sound/soc/intel/boards/cht_bsw_nau8824.c +++ b/sound/soc/intel/boards/cht_bsw_nau8824.c @@ -259,7 +259,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* override plaform name, if required */ snd_soc_card_cht.dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; platform_name = mach->mach_params.platform; ret_val = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cht, diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index b5b016d493f1..452691db12cc 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -539,7 +539,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) if (!drv) return -ENOMEM; - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; for (i = 0; i < ARRAY_SIZE(snd_soc_cards); i++) { if (acpi_dev_found(snd_soc_cards[i].codec_id) && diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index 2a6e5b124099..30de502b4fbb 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -451,7 +451,7 @@ static int snd_cml_rt1011_probe(struct platform_device *pdev) return -ENOMEM; INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; snd_soc_card_cml.dev = &pdev->dev; platform_name = mach->mach_params.platform; diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 8e947bad143c..ea1de8b3f3cd 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -604,7 +604,7 @@ static int geminilake_audio_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, ctx); /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; platform_name = mach->mach_params.platform; ret = snd_soc_fixup_dai_links_platform_name(card, platform_name); diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 3dadf9bff796..6589fa56873f 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -193,7 +193,7 @@ static int haswell_audio_probe(struct platform_device *pdev) haswell_rt5640.dev = &pdev->dev; /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(&haswell_rt5640, mach->mach_params.platform); if (ret) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index f65feee1c166..20d566e9dd9d 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -962,7 +962,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) kabylake_audio_card->dev = &pdev->dev; snd_soc_card_set_drvdata(kabylake_audio_card, ctx); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; if (mach) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 341bb47311a6..6493ede89300 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -772,7 +772,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) kabylake_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&kabylake_audio_card, ctx); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; if (mach) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index fe2d3a23a4ef..3be764299ab0 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -182,7 +182,7 @@ static int skl_hda_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; if (!mach) return -EINVAL; diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index e6de3b28d840..8216c15fc8da 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -660,7 +660,7 @@ static int skylake_audio_probe(struct platform_device *pdev) skylake_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&skylake_audio_card, ctx); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; if (mach) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index c99c8b23e509..6f68712ffce9 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -703,7 +703,7 @@ static int skylake_audio_probe(struct platform_device *pdev) skylake_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&skylake_audio_card, ctx); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; if (mach) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index 8f44f13d2848..8c657da5fcf0 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -335,7 +335,7 @@ static int audio_probe(struct platform_device *pdev) card = (struct snd_soc_card *)pdev->id_entry->driver_data; card->dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(card, mach->mach_params.platform); if (ret) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 5d878873a8e0..99b5a5e01e38 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -604,7 +604,7 @@ static int sof_audio_probe(struct platform_device *pdev) dmi_check_system(sof_rt5682_quirk_table); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; /* A speaker amp might not be present when the quirk claims one is. * Detect this via whether the machine driver match includes quirk_data. -- 2.20.1

1 0

Applied "ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp" to the asoc tree
by Mark Brown 13 Mar '20

13 Mar '20

The patch ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp 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 3f32e596b03ef50fd2c49f2e13d42e1931525f35 Mon Sep 17 00:00:00 2001 From: Yong Zhi <yong.zhi(a)intel.com> Date: Thu, 12 Mar 2020 14:48:57 -0500 Subject: [PATCH] ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp Add Maxim MAX98360A plug-and-play Class-D amplifier support on SSP1, new card ID is sofda7219max98360a, name sof-da7219max98360a. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200312194859.4051-9-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/Kconfig | 4 +- sound/soc/intel/boards/sof_da7219_max98373.c | 74 +++++++++++++++++-- .../intel/common/soc-acpi-intel-jsl-match.c | 19 ++++- 3 files changed, 87 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 6833ef548710..ab4ce652cc1a 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -507,7 +507,7 @@ endif ## SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK if SND_SOC_SOF_JASPERLAKE config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH - tristate "SOF with DA7219 and MAX98373 in I2S Mode" + tristate "SOF with DA7219 and MAX98373/MAX98360A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_HDA_CODEC_HDMI @@ -516,7 +516,7 @@ config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH select SND_SOC_DMIC help This adds support for ASoC machine driver for SOF platforms - with DA7219 + MAX98373 I2S audio codec. + with DA7219 + MAX98373/MAX98360A I2S audio codec. Say Y if you have such a device. If unsure select "N". diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index 8c657da5fcf0..7847dd44f41b 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -2,7 +2,7 @@ // Copyright(c) 2019 Intel Corporation. /* - * Intel SOF Machine driver for DA7219 + MAX98373 codec + * Intel SOF Machine driver for DA7219 + MAX98373/MAX98360A codec */ #include <linux/input.h> @@ -69,11 +69,14 @@ static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Right Spk"), }; +static const struct snd_kcontrol_new m98360a_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + static const struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU), @@ -83,15 +86,23 @@ static const struct snd_soc_dapm_route audio_map[] = { { "Headphone Jack", NULL, "HPL" }, { "Headphone Jack", NULL, "HPR" }, - { "Left Spk", NULL, "Left BE_OUT" }, - { "Right Spk", NULL, "Right BE_OUT" }, - { "MIC", NULL, "Headset Mic" }, { "Headphone Jack", NULL, "Platform Clock" }, { "Headset Mic", NULL, "Platform Clock" }, }; +/* For MAX98373 amp */ +static const struct snd_soc_dapm_widget max98373_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + +static const struct snd_soc_dapm_route max98373_map[] = { + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, +}; + static struct snd_soc_jack headset; static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) @@ -133,6 +144,21 @@ static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } +static int speaker_amp_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + + /* Add widgets */ + ret = snd_soc_dapm_new_controls(&rtd->card->dapm, max98373_widgets, + ARRAY_SIZE(max98373_widgets)); + if (ret) + return ret; + + /* Add routes */ + return snd_soc_dapm_add_routes(&rtd->card->dapm, max98373_map, + ARRAY_SIZE(max98373_map)); +} + static int ssp1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -223,6 +249,8 @@ SND_SOC_DAILINK_DEF(ssp1_amps, DAILINK_COMP_ARRAY( /* Left */ COMP_CODEC(MAXIM_DEV0_NAME, MAX98373_CODEC_DAI), /* Right */ COMP_CODEC(MAXIM_DEV1_NAME, MAX98373_CODEC_DAI))); +/* For the driver-less spk amp */ +SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); @@ -254,6 +282,7 @@ static struct snd_soc_dai_link dais[] = { .id = 0, .ignore_pmdown_time = 1, .no_pcm = 1, + .init = speaker_amp_init, .dpcm_playback = 1, .dpcm_capture = 1, /* IV feedback */ .ops = &ssp1_ops, @@ -320,6 +349,21 @@ static struct snd_soc_card card_da7219_m98373 = { .late_probe = card_late_probe, }; +static struct snd_soc_card card_da7219_m98360a = { + .name = "da7219max98360a", + .owner = THIS_MODULE, + .dai_link = dais, + .num_links = ARRAY_SIZE(dais), + .controls = m98360a_controls, + .num_controls = ARRAY_SIZE(m98360a_controls), + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), + .fully_routed = true, + .late_probe = card_late_probe, +}; + static int audio_probe(struct platform_device *pdev) { static struct snd_soc_card *card; @@ -331,6 +375,17 @@ static int audio_probe(struct platform_device *pdev) if (!ctx) return -ENOMEM; + /* By default dais[0] is configured for max98373 */ + if (!strcmp(pdev->name, "sof_da7219_max98360a")) { + dais[0] = (struct snd_soc_dai_link) { + .name = "SSP1-Codec", + .id = 0, + .no_pcm = 1, + .dpcm_playback = 1, + .ignore_pmdown_time = 1, + SND_SOC_DAILINK_REG(ssp1_pin, dummy, platform) }; + } + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); card = (struct snd_soc_card *)pdev->id_entry->driver_data; card->dev = &pdev->dev; @@ -351,13 +406,17 @@ static const struct platform_device_id board_ids[] = { .name = "sof_da7219_max98373", .driver_data = (kernel_ulong_t)&card_da7219_m98373, }, + { + .name = "sof_da7219_max98360a", + .driver_data = (kernel_ulong_t)&card_da7219_m98360a, + }, { } }; static struct platform_driver audio = { .probe = audio_probe, .driver = { - .name = "sof_da7219_max98373", + .name = "sof_da7219_max98_360a_373", .pm = &snd_soc_pm_ops, }, .id_table = board_ids, @@ -368,4 +427,5 @@ module_platform_driver(audio) MODULE_DESCRIPTION("ASoC Intel(R) SOF Machine driver"); MODULE_AUTHOR("Yong Zhi <yong.zhi(a)intel.com>"); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:sof_da7219_max98360a"); MODULE_ALIAS("platform:sof_da7219_max98373"); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index ed2b125f6a11..70f01495a166 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -9,13 +9,30 @@ #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> +static struct snd_soc_acpi_codecs jsl_7219_98373_codecs = { + .num_codecs = 1, + .codecs = {"MX98373"} +}; + +/* + * When adding new entry to the snd_soc_acpi_intel_jsl_machines array, + * use .quirk_data member to distinguish different machine driver, + * and keep ACPI .id field unchanged for the common codec. + */ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { { .id = "DLGS7219", .drv_name = "sof_da7219_max98373", - .machine_quirk = snd_soc_acpi_codec_list, .sof_fw_filename = "sof-jsl.ri", .sof_tplg_filename = "sof-jsl-da7219.tplg", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &jsl_7219_98373_codecs, + }, + { + .id = "DLGS7219", + .drv_name = "sof_da7219_max98360a", + .sof_fw_filename = "sof-jsl.ri", + .sof_tplg_filename = "sof-jsl-da7219-mx98360a.tplg", }, {}, }; -- 2.20.1

1 0

[PATCH v2 2/2] ASoC: brcm: DSL/PON SoC device tree bindings of audio driver
by Kevin Li 13 Mar '20

13 Mar '20

Signed-off-by: Kevin Li <kevin-ke.li(a)broadcom.com> --- .../bindings/sound/brcm,bcm63xx-audio.txt | 29 +++++++++++++++++++ 1 file changed, 29 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt diff --git a/Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt b/Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt new file mode 100644 index 000000000000..007f524b4d15 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt @@ -0,0 +1,29 @@ +Broadcom DSL/PON BCM63xx Audio I2S controller + +Required properties: +- compatible: Should be "brcm,bcm63xx-i2s". +- #address-cells: 32bit valued, 1 cell. +- #size-cells: 32bit valued, 0 cell. +- reg: Should contain audio registers location and length +- interrupts: Should contain the interrupt for the controller. +- clocks: Must contain an entry for each entry in clock-names. + Please refer to clock-bindings.txt. +- clock-names: One of each entry matching the clocks phandles list: + - "i2sclk" (generated clock) Required. + - "i2sosc" (fixed 200MHz clock) Required. + +(1) : The generated clock is required only when any of TX and RX + works on Master Mode. +(2) : The fixed 200MHz clock is from internal chip and always on + +Example: + + i2s: bcm63xx-i2s { + #address-cells = <1>; + #size-cells = <0>; + compatible = "brcm,bcm63xx-i2s"; + reg = <0xFF802080 0xFF>; + interrupts = <GIC_SPI 84 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&i2sclk>, <&osc>; + clock-names = "i2sclk","i2sosc"; + }; -- 2.25.1

2 4

[PATCH 1/2] ASoC: rt5682: Fine tune the HP performance in soundwire mode
by Oder Chiou 13 Mar '20

13 Mar '20

The setting is sync with I2C/I2S mode. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5682.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index e1df2d076533..f4b8af128828 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -3462,6 +3462,8 @@ int rt5682_io_init(struct device *dev, struct sdw_slave *slave) RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA); regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1, 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); /* Soundwire */ regmap_write(rt5682->regmap, RT5682_PLL2_INTERNAL, 0xa266); -- 2.25.1

2 3

Re: linux-next: Tree for Mar 13 (sound/soc/codecs/rt5682.o)
by Randy Dunlap 13 Mar '20

13 Mar '20

On 3/13/20 4:03 AM, Stephen Rothwell wrote: > Hi all, > > Changes since 20200312: > on i386: ld: sound/soc/codecs/rt5682.o: in function `rt5682_sdw_hw_free': rt5682.c:(.text+0x8a2): undefined reference to `sdw_stream_remove_slave' ld: sound/soc/codecs/rt5682.o: in function `rt5682_sdw_hw_params': rt5682.c:(.text+0x989): undefined reference to `sdw_stream_add_slave' Full randconfig file is attached. -- ~Randy Reported-by: Randy Dunlap <rdunlap(a)infradead.org>

1 0

[PATCH] ASoC: wm_adsp: Use scnprintf() for the limited buffer output
by Takashi Iwai 13 Mar '20

13 Mar '20

snprintf() is a hard-to-use function, it's especially difficult to use it for concatenating substrings in a buffer with a limited size. Since snprintf() returns the would-be-output size, not the actual size, the subsequent use of snprintf() may point to the incorrect position. Use scnprintf() instead for fixing such potential errors. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/wm_adsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d3d32b501aca..ffb9836e0538 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1436,12 +1436,12 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, subname = NULL; /* don't append subname */ break; case 2: - ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s%c %.12s %x", dsp->name, *region_name, wm_adsp_fw_text[dsp->fw], alg_region->alg); break; default: - ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %.12s %x", dsp->name, wm_adsp_fw_text[dsp->fw], alg_region->alg); break; -- 2.16.4

3 2

[PATCH] ALSA: hda: Use scnprintf() for string truncation
by Takashi Iwai 13 Mar '20

13 Mar '20

snd_hdac_codec_modalias() truncates the string to the given size and returns its size, but it returned a wrong size from snprintf(). snprintf() returns the would-be-output size, not the actual size. Use scnprintf() instead to return the correct size. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 9a526aeef8da..e3119f5cb0d5 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -204,7 +204,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_set_chip_name); */ int snd_hdac_codec_modalias(struct hdac_device *codec, char *buf, size_t size) { - return snprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n", + return scnprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n", codec->vendor_id, codec->revision_id, codec->type); } EXPORT_SYMBOL_GPL(snd_hdac_codec_modalias); -- 2.16.4

1 0

[PATCH] ALSA: pcm: Fix superfluous snprintf() usage
by Takashi Iwai 13 Mar '20

13 Mar '20

show_pcm_class() returns obviously a short string that can't overflow PAGE_SIZE. And even if it were to overflow, using snprintf() there is just wrong, as it doesn't return the correct size. So simplify with sprintf() instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index a141a301369f..b6d2331a82f7 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1019,7 +1019,7 @@ static ssize_t show_pcm_class(struct device *dev, str = "none"; else str = strs[pcm->dev_class]; - return snprintf(buf, PAGE_SIZE, "%s\n", str); + return sprintf(buf, "%s\n", str); } static DEVICE_ATTR(pcm_class, 0444, show_pcm_class, NULL); -- 2.16.4

1 0

[RESEND PATCH v2 0/9] ALSA: compress: Add wma, alac and ape support
by Vinod Koul 13 Mar '20

13 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 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 | 136 +++++++++++++- sound/soc/qcom/qdsp6/q6asm.c | 243 +++++++++++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.h | 51 +++++- 5 files changed, 462 insertions(+), 7 deletions(-) -- 2.24.1

2 15

[PATCH] soundwire: stream: use sdw_write instead of update
by Srinivas Kandagatla 13 Mar '20

13 Mar '20

There is no point in using update for registers with write mask as 0xFF, this adds unecessary traffic on the bus. Just use sdw_write directly. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/stream.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 00348d1fc606..1b43d03c79ea 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -313,9 +313,9 @@ static int sdw_enable_disable_slave_ports(struct sdw_bus *bus, * it is safe to reset this register */ if (en) - ret = sdw_update(s_rt->slave, addr, 0xFF, p_rt->ch_mask); + ret = sdw_write(s_rt->slave, addr, p_rt->ch_mask); else - ret = sdw_update(s_rt->slave, addr, 0xFF, 0x0); + ret = sdw_write(s_rt->slave, addr, 0x0); if (ret < 0) dev_err(&s_rt->slave->dev, @@ -464,10 +464,9 @@ static int sdw_prep_deprep_slave_ports(struct sdw_bus *bus, addr = SDW_DPN_PREPARECTRL(p_rt->num); if (prep) - ret = sdw_update(s_rt->slave, addr, - 0xFF, p_rt->ch_mask); + ret = sdw_write(s_rt->slave, addr, p_rt->ch_mask); else - ret = sdw_update(s_rt->slave, addr, 0xFF, 0x0); + ret = sdw_write(s_rt->slave, addr, 0x0); if (ret < 0) { dev_err(&s_rt->slave->dev, -- 2.21.0

3 2

[PATCH v2 0/9] ALSA: compress: Add wma, alac and ape support
by Vinod Koul 13 Mar '20

13 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 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 | 136 +++++++++++++- sound/soc/qcom/qdsp6/q6asm.c | 243 +++++++++++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.h | 51 +++++- 5 files changed, 462 insertions(+), 7 deletions(-) -- 2.24.1

1 1

[PATCH v2 9/9] ALSA: compress: bump the version
by Vinod Koul 13 Mar '20

13 Mar '20

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> --- 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 -- 2.24.1

1 0

[PATCH v2 8/9] ASoC: qcom: q6asm-dai: add support for ALAC and APE decoders
by Vinod Koul 13 Mar '20

13 Mar '20

Qualcomm DSPs also supports the ALAC and APE decoders, so add support for these and convert the snd_codec_params to qdsp format. Signed-off-by: Vinod Koul <vkoul(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 53c250778eea..948710759824 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -628,12 +628,16 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, struct q6asm_dai_data *pdata; struct q6asm_flac_cfg flac_cfg; struct q6asm_wma_cfg wma_cfg; + struct q6asm_alac_cfg alac_cfg; + struct q6asm_ape_cfg ape_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; + struct snd_dec_alac *alac; + struct snd_dec_ape *ape; codec_options = &(prtd->codec_param.codec.options); @@ -756,6 +760,65 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, dev_err(dev, "WMA9 CMD failed:%d\n", ret); return -EIO; } + break; + + case SND_AUDIOCODEC_ALAC: + memset(&alac_cfg, 0x0, sizeof(alac_cfg)); + alac = &codec_options->alac_d; + + alac_cfg.sample_rate = params->codec.sample_rate; + alac_cfg.avg_bit_rate = params->codec.bit_rate; + alac_cfg.bit_depth = prtd->bits_per_sample; + alac_cfg.num_channels = params->codec.ch_in; + + alac_cfg.frame_length = alac->frame_length; + alac_cfg.pb = alac->pb; + alac_cfg.mb = alac->mb; + alac_cfg.kb = alac->kb; + alac_cfg.max_run = alac->max_run; + alac_cfg.compatible_version = alac->compatible_version; + alac_cfg.max_frame_bytes = alac->max_frame_bytes; + + switch (params->codec.ch_in) { + case 1: + alac_cfg.channel_layout_tag = (100 << 16) | 1; + break; + case 2: + alac_cfg.channel_layout_tag = (101 << 16) | 2; + break; + } + ret = q6asm_stream_media_format_block_alac(prtd->audio_client, + &alac_cfg); + if (ret < 0) { + dev_err(dev, "ALAC CMD Format block failed:%d\n", ret); + return -EIO; + } + break; + + case SND_AUDIOCODEC_APE: + memset(&ape_cfg, 0x0, sizeof(ape_cfg)); + ape = &codec_options->ape_d; + + ape_cfg.sample_rate = params->codec.sample_rate; + ape_cfg.num_channels = params->codec.ch_in; + ape_cfg.bits_per_sample = prtd->bits_per_sample; + + ape_cfg.compatible_version = ape->compatible_version; + ape_cfg.compression_level = ape->compression_level; + ape_cfg.format_flags = ape->format_flags; + ape_cfg.blocks_per_frame = ape->blocks_per_frame; + ape_cfg.final_frame_blocks = ape->final_frame_blocks; + ape_cfg.total_frames = ape->total_frames; + ape_cfg.seek_table_present = ape->seek_table_present; + + ret = q6asm_stream_media_format_block_ape(prtd->audio_client, + &ape_cfg); + if (ret < 0) { + dev_err(dev, "APE CMD Format block failed:%d\n", ret); + return -EIO; + } + break; + default: break; } @@ -855,10 +918,12 @@ 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 = 3; + caps->num_codecs = 5; caps->codecs[0] = SND_AUDIOCODEC_MP3; caps->codecs[1] = SND_AUDIOCODEC_FLAC; caps->codecs[2] = SND_AUDIOCODEC_WMA; + caps->codecs[3] = SND_AUDIOCODEC_ALAC; + caps->codecs[4] = SND_AUDIOCODEC_APE; return 0; } -- 2.24.1

1 0

[PATCH v2 7/9] ASoC: qcom: q6asm: add support for alac and ape configs
by Vinod Koul 13 Mar '20

13 Mar '20

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> --- 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, -- 2.24.1

1 0

[PATCH v2 6/9] ALSA: compress: add alac & ape decoder params
by Vinod Koul 13 Mar '20

13 Mar '20

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> --- 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 -- 2.24.1

1 0

[PATCH v2 5/9] ASoC: qcom: q6asm-dai: add support to wma decoder
by Vinod Koul 13 Mar '20

13 Mar '20

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> --- 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 8f245d03b6f5..53c250778eea 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -627,10 +627,13 @@ 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); @@ -692,6 +695,67 @@ 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; } @@ -791,9 +855,10 @@ 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; } -- 2.24.1

1 0

[PATCH v2 4/9] ASoC: qcom: q6asm: add support to wma config
by Vinod Koul 13 Mar '20

13 Mar '20

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> --- 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, -- 2.24.1

1 0

[PATCH v2 3/9] ASoC: qcom: q6asm: pass codec profile to q6asm_open_write
by Vinod Koul 13 Mar '20

13 Mar '20

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> --- 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 c0d422d0ab94..8f245d03b6f5 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -250,7 +250,7 @@ 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); @@ -652,7 +652,7 @@ 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); -- 2.24.1

1 0

[PATCH v2 2/9] ALSA: compress: Add wma decoder params
by Vinod Koul 13 Mar '20

13 Mar '20

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> --- 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 -- 2.24.1

1 0

[PATCH v2 1/9] ALSA: compress: add wma codec profiles
by Vinod Koul 13 Mar '20

13 Mar '20

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> --- 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) -- 2.24.1

1 0

[PATCH 0/9] ALSA: compress: Add wma, alac and ape support
by Vinod Koul 13 Mar '20

13 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. 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 | 136 +++++++++++++- sound/soc/qcom/qdsp6/q6asm.c | 243 +++++++++++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.h | 51 +++++- 5 files changed, 462 insertions(+), 7 deletions(-) ~Vinod -- 2.24.1

2 11

conf: USB-Audio: Add C-Media USB Headphone Set to the IEC958 blacklist
by GitHub pull_request - opened 13 Mar '20

13 Mar '20

alsa-project/alsa-lib pull request #36 was opened from tanuk: Fixes: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/issues/317 Signed-off-by: Tanu Kaskinen <tanuk(a)iki.fi> Request URL : https://github.com/alsa-project/alsa-lib/pull/36 Patch URL : https://github.com/alsa-project/alsa-lib/pull/36.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Re: mmotm 2020-03-11-21-11 uploaded (sound/soc/codecs/wcd9335.c)
by Randy Dunlap 13 Mar '20

13 Mar '20

On 3/11/20 9:12 PM, Andrew Morton wrote: > The mm-of-the-moment snapshot 2020-03-11-21-11 has been uploaded to > > http://www.ozlabs.org/~akpm/mmotm/ > > mmotm-readme.txt says > > README for mm-of-the-moment: > > http://www.ozlabs.org/~akpm/mmotm/ > > This is a snapshot of my -mm patch queue. Uploaded at random hopefully > more than once a week. > > You will need quilt to apply these patches to the latest Linus release (5.x > or 5.x-rcY). The series file is in broken-out.tar.gz and is duplicated in > http://ozlabs.org/~akpm/mmotm/series > > The file broken-out.tar.gz contains two datestamp files: .DATE and > .DATE-yyyy-mm-dd-hh-mm-ss. Both contain the string yyyy-mm-dd-hh-mm-ss, > followed by the base kernel version against which this patch series is to > be applied. > > This tree is partially included in linux-next. To see which patches are > included in linux-next, consult the `series' file. Only the patches > within the #NEXT_PATCHES_START/#NEXT_PATCHES_END markers are included in > linux-next. > > > A full copy of the full kernel tree with the linux-next and mmotm patches > already applied is available through git within an hour of the mmotm > release. Individual mmotm releases are tagged. The master branch always > points to the latest release, so it's constantly rebasing. > > https://github.com/hnaz/linux-mm > > The directory http://www.ozlabs.org/~akpm/mmots/ (mm-of-the-second) > contains daily snapshots of the -mm tree. It is updated more frequently > than mmotm, and is untested. > > A git copy of this tree is also available at > > https://github.com/hnaz/linux-mm on x86_64: ld: sound/soc/codecs/wcd9335.o: in function `wcd9335_trigger': wcd9335.c:(.text+0x451): undefined reference to `slim_stream_prepare' ld: wcd9335.c:(.text+0x465): undefined reference to `slim_stream_enable' ld: wcd9335.c:(.text+0x48f): undefined reference to `slim_stream_unprepare' ld: wcd9335.c:(.text+0x4a3): undefined reference to `slim_stream_disable' ld: sound/soc/codecs/wcd9335.o: in function `wcd9335_slim_status': wcd9335.c:(.text+0x23df): undefined reference to `of_slim_get_device' ld: wcd9335.c:(.text+0x2414): undefined reference to `slim_get_logical_addr' ld: wcd9335.c:(.text+0x2427): undefined reference to `__regmap_init_slimbus' ld: wcd9335.c:(.text+0x245f): undefined reference to `__regmap_init_slimbus' ld: sound/soc/codecs/wcd9335.o: in function `wcd9335_hw_params': wcd9335.c:(.text+0x3e05): undefined reference to `slim_stream_allocate' ld: sound/soc/codecs/wcd9335.o: in function `wcd9335_slim_driver_init': wcd9335.c:(.init.text+0x15): undefined reference to `__slim_driver_register' ld: sound/soc/codecs/wcd9335.o: in function `wcd9335_slim_driver_exit': wcd9335.c:(.exit.text+0x11): undefined reference to `slim_driver_unregister' Full randconfig file is attached. -- ~Randy Reported-by: Randy Dunlap <rdunlap(a)infradead.org>

3 2

[alsa-devel] [RESEND 0/4] Add a better separation between i.MX8 families
by Daniel Baluta 12 Mar '20

12 Mar '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> So far the implementation was designed to support a generic platform named i.MX8. Anyhow, now working with specific i.MX8 instances we need to account for the differences. i.MX8 naming can be confusing at the first glance, so we need to have a clean separation between different platforms. Here is the split of i.MX8 per platforms. Notice that i.MX8 names the entire family, but also a sub-family. imx8 ├── imx8 │ └── imx8qm (*) ├── imx8m │ ├── imx8mm │ ├── imx8mn │ ├── imx8mp (*) │ └── imx8mq └── imx8x └── imx8qxp (*) Platforms marked with (*) contain a DSP. In the future there might be more platforms. This patchseries does the following: * renames imx8 to imx8x (because the only supported platform now is imx8qxp). * adds support for imx8 (which is imx8qm) Paul Olaru (4): ASoC: SOF: Rename i.MX8 platform to i.MX8X ASoC: SOF: imx8: Add ops for i.MX8QM ASoC: SOF: Add i.MX8QM device descriptor dt-bindings: dsp: fsl: Add fsl,imx8qm-dsp entry .../devicetree/bindings/dsp/fsl,dsp.yaml | 1 + sound/soc/sof/imx/imx8.c | 57 ++++++++++++++++++- sound/soc/sof/sof-of-dev.c | 10 ++++ 3 files changed, 65 insertions(+), 3 deletions(-) -- 2.17.1

5 10

[PATCH] acpi: Add NHLT table signature
by Cezary Rojewski 12 Mar '20

12 Mar '20

NHLT (Non-HDAudio Link Table) provides configuration of audio endpoints for Intel SST (Smart Sound Technology) DSP products. Similarly to other ACPI tables, data provided by BIOS may not describe it correctly, thus overriding is required. ACPI override mechanism checks for unknown signature before proceeding. Update known signatures array to support NHLT. CC: Erik Kaneda <erik.kaneda(a)intel.com> CC: Robert Moore <robert.moore(a)intel.com> CC: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- drivers/acpi/tables.c | 2 +- include/acpi/actbl2.h | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/acpi/tables.c b/drivers/acpi/tables.c index 180ac4329763..0e905c3d1645 100644 --- a/drivers/acpi/tables.c +++ b/drivers/acpi/tables.c @@ -501,7 +501,7 @@ static const char * const table_sigs[] = { ACPI_SIG_WDDT, ACPI_SIG_WDRT, ACPI_SIG_DSDT, ACPI_SIG_FADT, ACPI_SIG_PSDT, ACPI_SIG_RSDT, ACPI_SIG_XSDT, ACPI_SIG_SSDT, ACPI_SIG_IORT, ACPI_SIG_NFIT, ACPI_SIG_HMAT, ACPI_SIG_PPTT, - NULL }; + ACPI_SIG_NHLT, NULL }; #define ACPI_HEADER_SIZE sizeof(struct acpi_table_header) diff --git a/include/acpi/actbl2.h b/include/acpi/actbl2.h index e45ced27f4c3..876ccf50ec36 100644 --- a/include/acpi/actbl2.h +++ b/include/acpi/actbl2.h @@ -43,6 +43,7 @@ #define ACPI_SIG_SBST "SBST" /* Smart Battery Specification Table */ #define ACPI_SIG_SDEI "SDEI" /* Software Delegated Exception Interface Table */ #define ACPI_SIG_SDEV "SDEV" /* Secure Devices table */ +#define ACPI_SIG_NHLT "NHLT" /* Non-HDAudio Link Table */ /* * All tables must be byte-packed to match the ACPI specification, since -- 2.17.1

1 0

[PATCH v1 1/2] dt-bindings: sound: convert rockchip spdif bindings to yaml
by Johan Jonker 12 Mar '20

12 Mar '20

Current dts files with 'spdif' nodes are manually verified. In order to automate this process rockchip-spdif.txt has to be converted to yaml. Also rk3188.dtsi, rk3288.dtsi use an extra fallback string, so change this in the documentation. Changed: "rockchip,rk3188-spdif", "rockchip,rk3066-spdif" "rockchip,rk3288-spdif", "rockchip,rk3066-spdif" Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> --- .../devicetree/bindings/sound/rockchip-spdif.txt | 45 ---------- .../devicetree/bindings/sound/rockchip-spdif.yaml | 96 ++++++++++++++++++++++ 2 files changed, 96 insertions(+), 45 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt deleted file mode 100644 index ec20c1271..000000000 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt +++ /dev/null @@ -1,45 +0,0 @@ -* Rockchip SPDIF transceiver - -The S/PDIF audio block is a stereo transceiver that allows the -processor to receive and transmit digital audio via an coaxial cable or -a fibre cable. - -Required properties: - -- compatible: should be one of the following: - - "rockchip,rk3066-spdif" - - "rockchip,rk3188-spdif" - - "rockchip,rk3228-spdif" - - "rockchip,rk3288-spdif" - - "rockchip,rk3328-spdif" - - "rockchip,rk3366-spdif" - - "rockchip,rk3368-spdif" - - "rockchip,rk3399-spdif" -- reg: physical base address of the controller and length of memory mapped - region. -- interrupts: should contain the SPDIF interrupt. -- dmas: DMA specifiers for tx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should be "tx" -- clocks: a list of phandle + clock-specifier pairs, one for each entry - in clock-names. -- clock-names: should contain following: - - "hclk": clock for SPDIF controller - - "mclk" : clock for SPDIF bus - -Required properties on RK3288: - - rockchip,grf: the phandle of the syscon node for the general register - file (GRF) - -Example for the rk3188 SPDIF controller: - -spdif: spdif@1011e000 { - compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; - reg = <0x1011e000 0x2000>; - interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&dmac1_s 8>; - dma-names = "tx"; - clock-names = "hclk", "mclk"; - clocks = <&cru HCLK_SPDIF>, <&cru SCLK_SPDIF>; - #sound-dai-cells = <0>; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml new file mode 100644 index 000000000..45c6eea30 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -0,0 +1,96 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip SPDIF transceiver + +description: + The S/PDIF audio block is a stereo transceiver that allows the + processor to receive and transmit digital audio via a coaxial or + fibre cable. + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + oneOf: + - const: rockchip,rk3066-spdif + - const: rockchip,rk3228-spdif + - const: rockchip,rk3328-spdif + - const: rockchip,rk3366-spdif + - const: rockchip,rk3368-spdif + - const: rockchip,rk3399-spdif + - items: + - enum: + - rockchip,rk3188-spdif + - rockchip,rk3288-spdif + - const: rockchip,rk3066-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for SPDIF bus + - description: clock for SPDIF controller + + clock-names: + items: + - const: mclk + - const: hclk + + dmas: + items: + - description: TX DMA Channel + + dma-names: + items: + - const: tx + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + Required property on RK3288. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +if: + properties: + compatible: + contains: + const: rockchip,rk3288-spdif + +then: + required: + - rockchip,grf + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3188-cru-common.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + spdif: spdif@1011e000 { + compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; + reg = <0x1011e000 0x2000>; + interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_SPDIF>, <&cru HCLK_SPDIF>; + clock-names = "mclk", "hclk"; + dmas = <&dmac1_s 8>; + dma-names = "tx"; + }; -- 2.11.0

1 1

[PATCH] ASoC: topology: Perform component check upfront
by Amadeusz Sławiński 12 Mar '20

12 Mar '20

Function soc_tplg_dbytes_create(), calls soc_tplg_init_kcontrol() to perform additional driver specific initialization. While soc_tplg_init_kcontrol() ensures that component is valid before invoking ops->control_load, there is no such check at the end of soc_tplg_dbytes_create() where list_add() is used. Also in quite a few places, there is reference of tplg->comp->dapm or tplg->comp->card, without any checks for tplg->comp. In consequence of the above this may lead to referencing NULL pointer. This allows for removal of now unnecessary checks. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-topology.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..66958c57d884 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -251,7 +251,7 @@ static int soc_tplg_vendor_load_(struct soc_tplg *tplg, { int ret = 0; - if (tplg->comp && tplg->ops && tplg->ops->vendor_load) + if (tplg->ops && tplg->ops->vendor_load) ret = tplg->ops->vendor_load(tplg->comp, tplg->index, hdr); else { dev_err(tplg->dev, "ASoC: no vendor load callback for ID %d\n", @@ -283,7 +283,7 @@ static int soc_tplg_vendor_load(struct soc_tplg *tplg, static int soc_tplg_widget_load(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { - if (tplg->comp && tplg->ops && tplg->ops->widget_load) + if (tplg->ops && tplg->ops->widget_load) return tplg->ops->widget_load(tplg->comp, tplg->index, w, tplg_w); @@ -295,7 +295,7 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg, static int soc_tplg_widget_ready(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { - if (tplg->comp && tplg->ops && tplg->ops->widget_ready) + if (tplg->ops && tplg->ops->widget_ready) return tplg->ops->widget_ready(tplg->comp, tplg->index, w, tplg_w); @@ -307,7 +307,7 @@ static int soc_tplg_dai_load(struct soc_tplg *tplg, struct snd_soc_dai_driver *dai_drv, struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { - if (tplg->comp && tplg->ops && tplg->ops->dai_load) + if (tplg->ops && tplg->ops->dai_load) return tplg->ops->dai_load(tplg->comp, tplg->index, dai_drv, pcm, dai); @@ -318,7 +318,7 @@ static int soc_tplg_dai_load(struct soc_tplg *tplg, static int soc_tplg_dai_link_load(struct soc_tplg *tplg, struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg) { - if (tplg->comp && tplg->ops && tplg->ops->link_load) + if (tplg->ops && tplg->ops->link_load) return tplg->ops->link_load(tplg->comp, tplg->index, link, cfg); return 0; @@ -327,7 +327,7 @@ static int soc_tplg_dai_link_load(struct soc_tplg *tplg, /* tell the component driver that all firmware has been loaded in this request */ static void soc_tplg_complete(struct soc_tplg *tplg) { - if (tplg->comp && tplg->ops && tplg->ops->complete) + if (tplg->ops && tplg->ops->complete) tplg->ops->complete(tplg->comp); } @@ -684,7 +684,7 @@ EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_bind_event); static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, struct snd_kcontrol_new *k, struct snd_soc_tplg_ctl_hdr *hdr) { - if (tplg->comp && tplg->ops && tplg->ops->control_load) + if (tplg->ops && tplg->ops->control_load) return tplg->ops->control_load(tplg->comp, tplg->index, k, hdr); @@ -1174,7 +1174,7 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, static int soc_tplg_add_route(struct soc_tplg *tplg, struct snd_soc_dapm_route *route) { - if (tplg->comp && tplg->ops && tplg->ops->dapm_route_load) + if (tplg->ops && tplg->ops->dapm_route_load) return tplg->ops->dapm_route_load(tplg->comp, tplg->index, route); @@ -2564,7 +2564,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, } /* pass control to component driver for optional further init */ - if (tplg->comp && tplg->ops && tplg->ops->manifest) + if (tplg->ops && tplg->ops->manifest) ret = tplg->ops->manifest(tplg->comp, tplg->index, _manifest); if (!abi_match) /* free the duplicated one */ @@ -2736,6 +2736,10 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, struct soc_tplg tplg; int ret; + /* component needs to exist to keep and reference data while parsing */ + if (!comp) + return -EINVAL; + /* setup parsing context */ memset(&tplg, 0, sizeof(tplg)); tplg.fw = fw; -- 2.17.1

3 4

[PATCH] ALSA: pcm: oss: Remove WARNING from snd_pcm_plug_alloc() checks
by Takashi Iwai 12 Mar '20

12 Mar '20

The return value checks in snd_pcm_plug_alloc() are covered with snd_BUG_ON() macro that may trigger a kernel WARNING depending on the kconfig. But since the error condition can be triggered by a weird user space parameter passed to OSS layer, we shouldn't give the kernel stack trace just for that. As it's a normal error condition, let's remove snd_BUG_ON() macro usage there. Reported-by: syzbot+2a59ee7a9831b264f45e(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_plugin.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index c9401832967c..752d078908e9 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -111,7 +111,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->next) { if (plugin->dst_frames) frames = plugin->dst_frames(plugin, frames); - if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) + if ((snd_pcm_sframes_t)frames <= 0) return -ENXIO; plugin = plugin->next; err = snd_pcm_plugin_alloc(plugin, frames); @@ -123,7 +123,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->prev) { if (plugin->src_frames) frames = plugin->src_frames(plugin, frames); - if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) + if ((snd_pcm_sframes_t)frames <= 0) return -ENXIO; plugin = plugin->prev; err = snd_pcm_plugin_alloc(plugin, frames); -- 2.16.4

1 0

linux-next test error: WARNING in snd_pcm_plug_alloc
by syzbot 12 Mar '20

12 Mar '20

Hello, syzbot found the following crash on: HEAD commit: 407b0a62 Add linux-next specific files for 20200312 git tree: linux-next console output: https://syzkaller.appspot.com/x/log.txt?x=14dd6545e00000 kernel config: https://syzkaller.appspot.com/x/.config?x=d644875269fc730b dashboard link: https://syzkaller.appspot.com/bug?extid=2a59ee7a9831b264f45e compiler: gcc (GCC) 9.0.0 20181231 (experimental) Unfortunately, I don't have any reproducer for this crash yet. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+2a59ee7a9831b264f45e(a)syzkaller.appspotmail.com ------------[ cut here ]------------ WARNING: CPU: 0 PID: 8559 at sound/core/oss/pcm_plugin.c:126 snd_pcm_plug_alloc+0x29a/0x330 sound/core/oss/pcm_plugin.c:126 Kernel panic - not syncing: panic_on_warn set ... CPU: 0 PID: 8559 Comm: syz-fuzzer Not tainted 5.6.0-rc5-next-20200312-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x188/0x20d lib/dump_stack.c:118 panic+0x2e3/0x75c kernel/panic.c:221 __warn.cold+0x2f/0x35 kernel/panic.c:582 report_bug+0x27b/0x2f0 lib/bug.c:195 fixup_bug arch/x86/kernel/traps.c:175 [inline] fixup_bug arch/x86/kernel/traps.c:170 [inline] do_error_trap+0x12b/0x220 arch/x86/kernel/traps.c:267 do_invalid_op+0x32/0x40 arch/x86/kernel/traps.c:286 invalid_op+0x23/0x30 arch/x86/entry/entry_64.S:1027 RIP: 0010:snd_pcm_plug_alloc+0x29a/0x330 sound/core/oss/pcm_plugin.c:126 Code: ff ff 45 31 e4 e8 d6 4d 7f fb 44 89 e0 5b 5d 41 5c 41 5d 41 5e c3 e8 c5 4d 7f fb 0f 0b 41 bc fa ff ff ff eb e0 e8 b6 4d 7f fb <0f> 0b 41 bc fa ff ff ff eb d1 e8 a7 4d 7f fb 0f 0b 41 bc fa ff ff RSP: 0018:ffffc90002317b88 EFLAGS: 00010293 RAX: ffff888093a7c200 RBX: ffff8880a4086000 RCX: ffffffff85f37886 RDX: 0000000000000000 RSI: ffffffff85f37a0a RDI: 0000000000000007 RBP: 0000000000000000 R08: ffff888093a7c200 R09: ffffed10132a791f R10: ffff88809953c8f3 R11: ffffed10132a791e R12: ffff8880a4086058 R13: dffffc0000000000 R14: 0000000000000000 R15: ffff88809953c800 snd_pcm_oss_change_params_locked+0x1c05/0x34b0 sound/core/oss/pcm_oss.c:1021 snd_pcm_oss_change_params+0x76/0xd0 sound/core/oss/pcm_oss.c:1084 snd_pcm_oss_make_ready+0xb7/0x170 sound/core/oss/pcm_oss.c:1143 snd_pcm_oss_sync.isra.0+0x1be/0x7d0 sound/core/oss/pcm_oss.c:1708 snd_pcm_oss_release+0x210/0x280 sound/core/oss/pcm_oss.c:2546 __fput+0x2da/0x850 fs/file_table.c:280 task_work_run+0xf4/0x1b0 kernel/task_work.c:123 tracehook_notify_resume include/linux/tracehook.h:188 [inline] exit_to_usermode_loop+0x2fa/0x360 arch/x86/entry/common.c:165 prepare_exit_to_usermode arch/x86/entry/common.c:196 [inline] syscall_return_slowpath arch/x86/entry/common.c:279 [inline] do_syscall_64+0x6b1/0x7d0 arch/x86/entry/common.c:305 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x4afb40 Code: 8b 7c 24 10 48 8b 74 24 18 48 8b 54 24 20 49 c7 c2 00 00 00 00 49 c7 c0 00 00 00 00 49 c7 c1 00 00 00 00 48 8b 44 24 08 0f 05 <48> 3d 01 f0 ff ff 76 20 48 c7 44 24 28 ff ff ff ff 48 c7 44 24 30 RSP: 002b:000000c000213588 EFLAGS: 00000212 ORIG_RAX: 0000000000000003 RAX: 0000000000000000 RBX: 000000c00002c000 RCX: 00000000004afb40 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 0000000000000003 RBP: 000000c0002135c8 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000212 R12: 0000000000000006 R13: 0000000000000005 R14: 0000000000000200 R15: 0000000000000000 Kernel Offset: disabled Rebooting in 86400 seconds.. --- 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.

2 1

[PATCH v2 0/2] ASoC: SOF: preparatory patches
by Guennadi Liakhovetski 12 Mar '20

12 Mar '20

2 preparatory patches for the SOF virtualisation work. No functional changes, only some code refactoring and exporting of helper functions. v2: rebase, no changes otherwise. Thanks Guennadi Guennadi Liakhovetski (2): ASoC: (cosmetic) simplify dpcm_prune_paths() ASoC: export DPCM runtime update functions include/sound/soc-dpcm.h | 4 +-- sound/soc/soc-dapm.c | 8 +++--- sound/soc/soc-pcm.c | 68 +++++++++++++++++++++++++----------------------- 3 files changed, 42 insertions(+), 38 deletions(-) -- 1.9.3

3 6

[PATCH] ASoC: dt-bindings: google, cros-ec-codec: Fix dtc warnings in example
by Rob Herring 12 Mar '20

12 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 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: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> 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 Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../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..c84e656afb0a 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: ec-codec@10500000 { + compatible = "google,cros-ec-codec"; + #sound-dai-cells = <1>; + reg = <0x0 0x10500000 0x80000>; + memory-region = <&reserved_mem>; + }; }; }; -- 2.20.1

3 2

[PATCH] MAINTAINERS: add entry for Sound Open Firmware drivers
by Kai Vehmanen 12 Mar '20

12 Mar '20

Sound Open Firmware (SOF) is an open source audio DSP firwmare instrastructure and SDK. The kernel drivers for SOF are part of the ALSA subsystem. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Acked-by: Daniel Baluta <daniel.baluta(a)nxp.com> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Acked-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- MAINTAINERS | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index d87507d53ca5..0b022b26bfa9 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -15618,6 +15618,17 @@ F: sound/soc/ F: include/dt-bindings/sound/ F: include/sound/soc* +SOUND - SOUND OPEN FIRMWARE (SOF) DRIVERS +M: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> +M: Liam Girdwood <lgirdwood(a)gmail.com> +M: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> +M: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> +M: Daniel Baluta <daniel.baluta(a)nxp.com> +L: sound-open-firmware(a)alsa-project.org (moderated for non-subscribers) +W: https://github.com/thesofproject/linux/ +S: Supported +F: sound/soc/sof/ + SOUNDWIRE SUBSYSTEM M: Vinod Koul <vkoul(a)kernel.org> M: Sanyog Kale <sanyog.r.kale(a)intel.com> -- 2.17.1

2 1

[PATCH] MAINTAINERS: Update Cirrus Logic codec driver maintainers
by James Schulman 12 Mar '20

12 Mar '20

Brian & Paul are no longer active audio codec driver maintainers. Update list to reflect myself and David Rhodes as the active maintainers. Signed-off-by: James Schulman <james.schulman(a)cirrus.com> --- MAINTAINERS | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index a6fbdf354d34..73f1539d430e 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4021,8 +4021,8 @@ F: Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml F: sound/soc/codecs/cros_ec_codec.* CIRRUS LOGIC AUDIO CODEC DRIVERS -M: Brian Austin <brian.austin(a)cirrus.com> -M: Paul Handrigan <Paul.Handrigan(a)cirrus.com> +M: James Schulman <james.schulman(a)cirrus.com> +M: David Rhodes <david.rhodes(a)cirrus.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained F: sound/soc/codecs/cs* -- 2.17.1

2 2

Applied "ASoC: SOF: Intel: Fix stream cleanup on hw free" to the asoc tree
by Mark Brown 12 Mar '20

12 Mar '20

The patch ASoC: SOF: Intel: Fix stream cleanup on hw free 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 16dcefc23eefda57ee705dcffba67f1bd42c01bc Mon Sep 17 00:00:00 2001 From: Cezary Rojewski <cezary.rojewski(a)intel.com> Date: Thu, 12 Mar 2020 13:00:58 +0100 Subject: [PATCH] ASoC: SOF: Intel: Fix stream cleanup on hw free Field "substream" gets assigned during stream setup in hda_dsp_pcm_hw_params() but it is never cleared afterwards during cleanup procedure. Now, any non-pcm operation e.g.: compress can mistakenly make use of that pointer as it's bypassing all "if (s->substream)" checks. Nulling the pointer during hw_free operation ensures no wild pointers are left behind. Fixes: cdae3b9a47aa ("ASoC: SOF: Intel: Add Intel specific HDA PCM operations") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200312120058.15057-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-stream.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index c0ab9bb2a797..d2234f802788 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -547,6 +547,8 @@ int hda_dsp_stream_hw_free(struct snd_sof_dev *sdev, SOF_HDA_REG_PP_PPCTL, mask, 0); spin_unlock_irq(&bus->reg_lock); + stream->substream = NULL; + return 0; } -- 2.20.1

1 0

[PATCH 0/3] [RESEND] ASoC: SOF: preparatory patches
by Guennadi Liakhovetski 12 Mar '20

12 Mar '20

This is the first set of patches for the SOF virtualisation work. We send these patches first because they touch the ASoC core. 2 of them are mostly cosmetic with no functional changes, but patch 2/3 might cause some discussions. Please review and comment. Thanks Guennadi Guennadi Liakhovetski (3): ASoC: (cosmetic) simplify dpcm_prune_paths() ASoC: add function parameters to enable forced path pruning ASoC: export DPCM runtime update functions include/sound/soc-dpcm.h | 30 ++++++++--- sound/soc/soc-compress.c | 2 +- sound/soc/soc-dapm.c | 8 +-- sound/soc/soc-pcm.c | 130 +++++++++++++++++++++++++++++------------------ 4 files changed, 109 insertions(+), 61 deletions(-) -- 1.9.3

3 20

[PATCH v3] ASoC: SOF: Intel: Fix stream cleanup on hw free
by Cezary Rojewski 12 Mar '20

12 Mar '20

Field "substream" gets assigned during stream setup in hda_dsp_pcm_hw_params() but it is never cleared afterwards during cleanup procedure. Now, any non-pcm operation e.g.: compress can mistakenly make use of that pointer as it's bypassing all "if (s->substream)" checks. Nulling the pointer during hw_free operation ensures no wild pointers are left behind. Fixes: cdae3b9a47aa ("ASoC: SOF: Intel: Add Intel specific HDA PCM operations") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- Changes in v3: - simple rewording of commit title & message Changes in v2: - relocated stream->substream assignment from _pcm_close() to _hw_free() sound/soc/sof/intel/hda-stream.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 7daa913dbde0..5d386956906f 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -547,6 +547,8 @@ int hda_dsp_stream_hw_free(struct snd_sof_dev *sdev, SOF_HDA_REG_PP_PPCTL, mask, 0); spin_unlock_irq(&bus->reg_lock); + stream->substream = NULL; + return 0; } -- 2.17.1

1 0

[PATCH v2] ASoC: SOF: Intel: Fix stream cleanup on pcm close
by Cezary Rojewski 12 Mar '20

12 Mar '20

Field "substream" gets assigned during stream setup in hda_dsp_pcm_hw_params() but it is never cleared afterwards during cleanup procedure. Now, any non-pcm operation e.g.: compress can mistakenly make use of that pointer as it's bypassing all "if (s->substream)" checks. Nulling the pointer during close operation ensures no wild pointers are left behind. Fixes: cdae3b9a47aa ("ASoC: SOF: Intel: Add Intel specific HDA PCM operations") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- Changes in v2: - relocated stream->substream assignment from _pcm_close() to _hw_free() sound/soc/sof/intel/hda-stream.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 7daa913dbde0..5d386956906f 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -547,6 +547,8 @@ int hda_dsp_stream_hw_free(struct snd_sof_dev *sdev, SOF_HDA_REG_PP_PPCTL, mask, 0); spin_unlock_irq(&bus->reg_lock); + stream->substream = NULL; + return 0; } -- 2.17.1

2 1

[PATCH] ASoC: SOF: Intel: Fix stream cleanup on pcm close
by Cezary Rojewski 12 Mar '20

12 Mar '20

Field "substream" gets assigned during stream setup in hda_dsp_pcm_hw_params() but it is never cleared afterwards during hda_dsp_pcm_close(). Now, any non-pcm operation e.g.: compress can mistakenly make use of that pointer as it's bypassing all "if (s->substream)" checks. Nulling the pointer during close operation ensures no wild pointers are left behind. Fixes: cdae3b9a47aa ("ASoC: SOF: Intel: Add Intel specific HDA PCM operations") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/sof/intel/hda-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index a46a6baa1c3f..4b3a89cf20e7 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -246,5 +246,6 @@ int hda_dsp_pcm_close(struct snd_sof_dev *sdev, /* unbinding pcm substream to hda stream */ substream->runtime->private_data = NULL; + hstream->substream = NULL; return 0; } -- 2.17.1

2 4

[PATCH -next 000/491] treewide: use fallthrough;
by Joe Perches 12 Mar '20

12 Mar '20

There is a new fallthrough pseudo-keyword macro that can be used to replace the various /* fallthrough */ style comments that are used to indicate a case label code block is intended to fallthrough to the next case label block. See commit 294f69e662d1 ("compiler_attributes.h: Add 'fallthrough' pseudo keyword for switch/case use") These patches are intended to allow clang to detect missing switch/case fallthrough uses. Do a depth-first pass on the MAINTAINERS file and find the various F: pattern files and convert the fallthrough comments to fallthrough; for all files matched by all F: patterns in in each section. Done via the perl script below and the previously posted cvt_fallthrough.pl script. Link: https://lore.kernel.org/lkml/b56602fcf79f849e733e7b521bb0e17895d390fa.15822… These patches are based on next-20200310 and are available in git://repo.or.cz/linux-2.6/trivial-mods.git in branch 20200310_fallthrough_2 $ cat commit_fallthrough.pl #!/usr/bin/env perl use sort 'stable'; # # Reorder a sorted array so file entries are before directory entries # depends on a trailing / for directories # so: # foo/ # foo/bar.c # becomes # foo/bar.c # foo/ # sub file_before_directory { my ($array_ref) = (@_); my $count = scalar(@$array_ref); for (my $i = 1; $i < $count; $i++) { if (substr(@$array_ref[$i - 1], -1) eq '/' && substr(@$array_ref[$i], 0, length(@$array_ref[$i - 1])) eq @$array_ref[$i - 1]) { my $string = @$array_ref[$i - 1]; @$array_ref[$i - 1] = @$array_ref[$i]; @$array_ref[$i] = $string; } } } sub uniq { my (@parms) = @_; my %saw; @parms = grep(!$saw{$_}++, @parms); return @parms; } # Get all the F: file patterns in MAINTAINERS that could be a .[ch] file my $maintainer_patterns = `grep -P '^F:\\s+' MAINTAINERS`; my @patterns = split('\n', $maintainer_patterns); s/^F:\s*// for @patterns; @patterns = grep(!/^(?:Documentation|tools|scripts)\//, @patterns); @patterns = grep(!/\.(?:dtsi?|rst|config)$/, @patterns); @patterns = sort @patterns; @patterns = sort { $b =~ tr/\//\// cmp $a =~ tr/\//\// } @patterns; file_before_directory(\@patterns); my %sections_done; foreach my $pattern (@patterns) { # Find the files the pattern matches my $pattern_files = `git ls-files -- $pattern`; my @new_patterns = split('\n', $pattern_files); $pattern_files = join(' ', @new_patterns); next if ($pattern_files =~ /^\s*$/); # Find the section the first file matches my $pattern_file = @new_patterns[0]; my $section_output = `./scripts/get_maintainer.pl --nogit --nogit-fallback --sections --pattern-depth=1 $pattern_file`; my @section = split('\n', $section_output); my $section_header = @section[0]; print("Section: <$section_header>\n"); # Skip the section if it's already done print("Already done '$section_header'\n") if ($sections_done{$section_header}); next if ($sections_done{$section_header}++); # Find all the .[ch] files in all F: lines in that section my @new_section; foreach my $line (@section) { last if ($line =~ /^\s*$/); push(@new_section, $line); } @section = grep(/^F:/, @new_section); s/^F:\s*// for @section; @section = grep(!/^(?:Documentation|tools|scripts)\//, @section); @section = grep(!/\.(?:dtsi?|rst|config)$/, @section); @section = sort @section; @section = uniq(@section); my $section_files = join(' ', @section); print("section_files: <$section_files>\n"); next if ($section_files =~ /^\s*$/); my $cvt_files = `git ls-files -- $section_files`; my @files = split('\n', $cvt_files); @files = grep(!/^(?:Documentation|tools|scripts)\//, @files); @files = grep(!/\.(?:dtsi?|rst|config)$/, @files); @files = grep(/\.[ch]$/, @files); @files = sort @files; @files = uniq(@files); $cvt_files = join(' ', @files); print("files: <$cvt_files>\n"); next if (scalar(@files) < 1); # Convert fallthroughs for all [.ch] files in the section print("doing cvt_fallthrough.pl -- $cvt_files\n"); `cvt_fallthrough.pl -- $cvt_files`; # If nothing changed, nothing to commit `git diff-index --quiet HEAD --`; next if (!$?); # Commit the changes my $fh; open($fh, "+>", "cvt_fallthrough.commit_msg") or die "$0: can't create temporary file: $!\n"; print $fh <<EOF $section_header: Use fallthrough; Convert the various uses of fallthrough comments to fallthrough; Done via script Link: https://lore.kernel.org/lkml/b56602fcf79f849e733e7b521bb0e17895d390fa.15822… EOF ; close $fh; `git commit -s -a -F cvt_fallthrough.commit_msg`; } Joe Perches (491): MELLANOX ETHERNET INNOVA DRIVERS: Use fallthrough; MARVELL OCTEONTX2 RVU ADMIN FUNCTION DRIVER: Use fallthrough; MELLANOX MLX5 core VPI driver: Use fallthrough; PERFORMANCE EVENTS SUBSYSTEM: Use fallthrough; ARM/UNIPHIER ARCHITECTURE: Use fallthrough; ARM/RISCPC ARCHITECTURE: Use fallthrough; KERNEL VIRTUAL MACHINE FOR ARM/ARM64 (KVM/arm, KVM/arm64): Use fallthrough; ARM/CIRRUS LOGIC EP93XX ARM ARCHITECTURE: Use fallthrough; ARM/Amlogic Meson SoC support: Use fallthrough; ARM/SAMSUNG EXYNOS ARM ARCHITECTURES: Use fallthrough; ARM/QUALCOMM SUPPORT: Use fallthrough; ARM/ZTE ARCHITECTURE: Use fallthrough; INGENIC JZ47xx SoCs: Use fallthrough; DECSTATION PLATFORM SUPPORT: Use fallthrough; KERNEL VIRTUAL MACHINE FOR MIPS (KVM/mips): Use fallthrough; KERNEL VIRTUAL MACHINE FOR POWERPC (KVM/powerpc): Use fallthrough; CELL BROADBAND ENGINE ARCHITECTURE: Use fallthrough; KERNEL VIRTUAL MACHINE for s390 (KVM/s390): Use fallthrough; Hyper-V CORE AND DRIVERS: Use fallthrough; XEN HYPERVISOR INTERFACE: Use fallthrough; KERNEL VIRTUAL MACHINE FOR X86 (KVM/x86): Use fallthrough; X86 MCE INFRASTRUCTURE: Use fallthrough; AMD KFD: Use fallthrough; AMD DISPLAY CORE: Use fallthrough; AMD POWERPLAY: Use fallthrough; INTEL GVT-g DRIVERS (Intel GPU Virtualization): Use fallthrough; AMD XGBE DRIVER: Use fallthrough; BROADCOM BNX2X 10 GIGABIT ETHERNET DRIVER: Use fallthrough; BROADCOM BNXT_EN 50 GIGABIT ETHERNET DRIVER: Use fallthrough; BROADCOM GENET ETHERNET DRIVER: Use fallthrough; BROCADE BNA 10 GIGABIT ETHERNET DRIVER: Use fallthrough; CAVIUM LIQUIDIO NETWORK DRIVER: Use fallthrough; ARM/CAVIUM THUNDER NETWORK DRIVER: Use fallthrough; CXGB3 ETHERNET DRIVER (CXGB3): Use fallthrough; CXGB4 ETHERNET DRIVER (CXGB4): Use fallthrough; CXGB4VF ETHERNET DRIVER (CXGB4VF): Use fallthrough; CISCO VIC ETHERNET NIC DRIVER: Use fallthrough; TULIP NETWORK DRIVERS: Use fallthrough; Emulex 10Gbps NIC BE2, BE3-R, Lancer, Skyhawk-R DRIVER (be2net): Use fallthrough; DPAA2 ETHERNET DRIVER: Use fallthrough; HISILICON NETWORK SUBSYSTEM 3 DRIVER (HNS3): Use fallthrough; EHEA (IBM pSeries eHEA 10Gb ethernet adapter) DRIVER: Use fallthrough; INTEL ETHERNET DRIVERS: Use fallthrough; MELLANOX ETHERNET DRIVER (mlx4_en): Use fallthrough; MELLANOX MLX4 core VPI driver: Use fallthrough; MELLANOX FIRMWARE FLASH LIBRARY (mlxfw): Use fallthrough; MELLANOX ETHERNET SWITCH DRIVERS: Use fallthrough; NETXEN (1/10) GbE SUPPORT: Use fallthrough; QLOGIC QL4xxx ETHERNET DRIVER: Use fallthrough; QLOGIC QLCNIC (1/10)Gb ETHERNET DRIVER: Use fallthrough; SAMSUNG SXGBE DRIVERS: Use fallthrough; STMMAC ETHERNET DRIVER: Use fallthrough; QUALCOMM ATHEROS ATH10K WIRELESS DRIVER: Use fallthrough; QUALCOMM ATHEROS ATH11K WIRELESS DRIVER: Use fallthrough; ATHEROS ATH5K WIRELESS DRIVER: Use fallthrough; ATHEROS ATH6KL WIRELESS DRIVER: Use fallthrough; QUALCOMM ATHEROS ATH9K WIRELESS DRIVER: Use fallthrough; CARL9170 LINUX COMMUNITY WIRELESS DRIVER: Use fallthrough; QUALCOMM WCN36XX WIRELESS DRIVER: Use fallthrough; B43 WIRELESS DRIVER: Use fallthrough; B43LEGACY WIRELESS DRIVER: Use fallthrough; BROADCOM BRCM80211 IEEE802.11n WIRELESS DRIVER: Use fallthrough; INTEL WIRELESS 3945ABG/BG, 4965AGN (iwlegacy): Use fallthrough; INTEL WIRELESS WIFI LINK (iwlwifi): Use fallthrough; ORINOCO DRIVER: Use fallthrough; PRISM54 WIRELESS DRIVER: Use fallthrough; MARVELL MWIFIEX WIRELESS DRIVER: Use fallthrough; MEDIATEK MT76 WIRELESS LAN DRIVER: Use fallthrough; MEDIATEK MT7601U WIRELESS LAN DRIVER: Use fallthrough; RALINK RT2X00 WIRELESS LAN DRIVER: Use fallthrough; RTL8XXXU WIRELESS DRIVER (rtl8xxxu): Use fallthrough; REALTEK WIRELESS DRIVER (rtlwifi family): Use fallthrough; REALTEK WIRELESS DRIVER (rtw88): Use fallthrough; TRACING: Use fallthrough; OPROFILE: Use fallthrough; PTRACE SUPPORT: Use fallthrough; HIBERNATION (aka Software Suspend, aka swsusp): Use fallthrough; ARM PMU PROFILING AND DEBUGGING: Use fallthrough; ARM/ARTPEC MACHINE SUPPORT: Use fallthrough; ARM/Marvell Dove/MV78xx0/Orion SOC support: Use fallthrough; MMP SUPPORT: Use fallthrough; PXA2xx/PXA3xx SUPPORT: Use fallthrough; ARM/RDA MICRO ARCHITECTURE: Use fallthrough; ARM/Rockchip SoC support: Use fallthrough; ARM/NOMADIK/U300/Ux500 ARCHITECTURES: Use fallthrough; PS3 PLATFORM SUPPORT: Use fallthrough; LINUX FOR POWER MACINTOSH: Use fallthrough; SCx200 CPU SUPPORT: Use fallthrough; CAVIUM OCTEON-TX CRYPTO DRIVER: Use fallthrough; ARM MALI-DP DRM DRIVER: Use fallthrough; DRM DRIVER FOR AST SERVER GRAPHICS CHIPS: Use fallthrough; DRM DRIVERS FOR BRIDGE CHIPS: Use fallthrough; DRM DRIVERS FOR EXYNOS: Use fallthrough; DRM DRIVERS FOR FREESCALE DCU: Use fallthrough; INTEL DRM DRIVERS (excluding Poulsbo, Moorestown and derivative chipsets): Use fallthrough; DRM DRIVERS FOR FREESCALE IMX: Use fallthrough; DRM DRIVERS FOR AMLOGIC SOCS: Use fallthrough; DRM DRIVER FOR MSM ADRENO GPU: Use fallthrough; DRM DRIVER FOR NVIDIA GEFORCE/QUADRO GPUS: Use fallthrough; DRM DRIVERS FOR TI OMAP: Use fallthrough; RADEON and AMDGPU DRM DRIVERS: Use fallthrough; DRM DRIVER FOR SAVAGE VIDEO CARDS: Use fallthrough; DRM DRIVERS FOR ALLWINNER A10: Use fallthrough; DRM DRIVERS FOR NVIDIA TEGRA: Use fallthrough; DRM DRIVERS FOR TI LCDC: Use fallthrough; DRM TTM SUBSYSTEM: Use fallthrough; DRM DRIVERS FOR XEN: Use fallthrough; BROADCOM NETXTREME-E ROCE DRIVER: Use fallthrough; CXGB4 IWARP RNIC DRIVER (IW_CXGB4): Use fallthrough; INTEL RDMA RNIC DRIVER: Use fallthrough; MELLANOX MLX4 IB driver: Use fallthrough; MELLANOX MLX5 IB driver: Use fallthrough; EMULEX ONECONNECT ROCE DRIVER: Use fallthrough; QLOGIC QL4xxx RDMA DRIVER: Use fallthrough; QIB DRIVER: Use fallthrough; VMWARE PVRDMA DRIVER: Use fallthrough; SOFT-ROCE DRIVER (rxe): Use fallthrough; SOFT-IWARP DRIVER (siw): Use fallthrough; ISCSI EXTENSIONS FOR RDMA (ISER) INITIATOR: Use fallthrough; SIANO DVB DRIVER: Use fallthrough; VIDEOBUF2 FRAMEWORK: Use fallthrough; SAMSUNG S5C73M3 CAMERA DRIVER: Use fallthrough; SMIA AND SMIA++ IMAGE SENSOR DRIVER: Use fallthrough; BTTV VIDEO4LINUX DRIVER: Use fallthrough; CX88 VIDEO4LINUX DRIVER: Use fallthrough; MEDIA DRIVERS FOR DIGITAL DEVICES PCIE DEVICES: Use fallthrough; MOTION EYE VAIO PICTUREBOOK CAMERA DRIVER: Use fallthrough; SAA7134 VIDEO4LINUX DRIVER: Use fallthrough; SOFTLOGIC 6x10 MPEG CODEC: Use fallthrough; CODA V4L2 MEM2MEM DRIVER: Use fallthrough; SAMSUNG S5P/EXYNOS4 SOC SERIES CAMERA SUBSYSTEM DRIVERS: Use fallthrough; CAFE CMOS INTEGRATED CAMERA CONTROLLER DRIVER: Use fallthrough; OMAP IMAGING SUBSYSTEM (OMAP3 ISP and OMAP4 ISS): Use fallthrough; VICODEC VIRTUAL CODEC DRIVER: Use fallthrough; VIVID VIRTUAL VIDEO DRIVER: Use fallthrough; SI4713 FM RADIO TRANSMITTER I2C DRIVER: Use fallthrough; CEC FRAMEWORK: Use fallthrough; DVB_USB_AF9015 MEDIA DRIVER: Use fallthrough; DVB_USB_GL861 MEDIA DRIVER: Use fallthrough; LME2510 MEDIA DRIVER: Use fallthrough; DVB_USB_MXL111SF MEDIA DRIVER: Use fallthrough; EM28XX VIDEO4LINUX DRIVER: Use fallthrough; GO7007 MPEG CODEC: Use fallthrough; GSPCA SN9C20X SUBDRIVER: Use fallthrough; GSPCA USB WEBCAM DRIVER: Use fallthrough; PULSE8-CEC DRIVER: Use fallthrough; PWC WEBCAM DRIVER: Use fallthrough; TM6000 VIDEO4LINUX DRIVER: Use fallthrough; USB VIDEO CLASS: Use fallthrough; ONENAND FLASH DRIVER: Use fallthrough; BROADCOM B53 ETHERNET SWITCH DRIVER: Use fallthrough; MICROCHIP KSZ SERIES ETHERNET SWITCH DRIVER: Use fallthrough; MARVELL 88E6XXX ETHERNET SWITCH FABRIC DRIVER: Use fallthrough; 8390 NETWORK DRIVERS [WD80x3/SMC-ELITE, SMC-ULTRA, NE2000, 3C503, etc.]: Use fallthrough; ALACRITECH GIGABIT ETHERNET DRIVER: Use fallthrough; ACENIC DRIVER: Use fallthrough; BROADCOM BNX2 GIGABIT ETHERNET DRIVER: Use fallthrough; BROADCOM TG3 GIGABIT ETHERNET DRIVER: Use fallthrough; ATMEL MACB ETHERNET DRIVER: Use fallthrough; FREESCALE QORIQ DPAA ETHERNET DRIVER: Use fallthrough; FREESCALE QORIQ DPAA FMAN DRIVER: Use fallthrough; FREESCALE QUICC ENGINE UCC ETHERNET DRIVER: Use fallthrough; HISILICON NETWORK SUBSYSTEM DRIVER: Use fallthrough; LINUX FOR POWERPC (32-BIT AND 64-BIT): Use fallthrough; MARVELL MVNETA ETHERNET DRIVER: Use fallthrough; MARVELL GIGABIT ETHERNET DRIVERS (skge/sky2): Use fallthrough; MEDIATEK ETHERNET DRIVER: Use fallthrough; MICROCHIP LAN743X ETHERNET DRIVER: Use fallthrough; MICROSEMI ETHERNET SWITCH DRIVER: Use fallthrough; NATSEMI ETHERNET DRIVER (DP8381x): Use fallthrough; NETERION 10GbE DRIVERS (s2io/vxge): Use fallthrough; NETRONOME ETHERNET DRIVERS: Use fallthrough; QLOGIC QLA3XXX NETWORK DRIVER: Use fallthrough; 8169 10/100/1000 GIGABIT ETHERNET DRIVER: Use fallthrough; ROCKER DRIVER: Use fallthrough; SFC NETWORK DRIVER: Use fallthrough; SIS 900/7016 FAST ETHERNET DRIVER: Use fallthrough; SOCIONEXT (SNI) NETSEC NETWORK DRIVER: Use fallthrough; TI ETHERNET SWITCH DRIVER (CPSW): Use fallthrough; TLAN NETWORK DRIVER: Use fallthrough; SPIDERNET NETWORK DRIVER for CELL: Use fallthrough; INTEL WIRELESS WIMAX CONNECTION 2400: Use fallthrough; ATMEL WIRELESS DRIVER: Use fallthrough; TI WILINK WIRELESS DRIVERS: Use fallthrough; USB ZD1201 DRIVER: Use fallthrough; PCI DRIVER FOR IMX6: Use fallthrough; ANALOG DEVICES INC IIO DRIVERS: Use fallthrough; HANTRO VPU CODEC DRIVER: Use fallthrough; MEDIA DRIVERS FOR FREESCALE IMX: Use fallthrough; SOC-CAMERA V4L2 SUBSYSTEM: Use fallthrough; USB VISION DRIVER: Use fallthrough; PARISC ARCHITECTURE: Use fallthrough; MICROCHIP USBA UDC DRIVER: Use fallthrough; FREESCALE USB PERIPHERAL DRIVERS: Use fallthrough; USB TYPEC PORT CONTROLLER DRIVERS: Use fallthrough; RADEON FRAMEBUFFER DISPLAY DRIVER: Use fallthrough; ARC FRAMEBUFFER DRIVER: Use fallthrough; NVIDIA (rivafb and nvidiafb) FRAMEBUFFER DRIVER: Use fallthrough; OMAP FRAMEBUFFER SUPPORT: Use fallthrough; OMAP DISPLAY SUBSYSTEM and FRAMEBUFFER SUPPORT (DSS2): Use fallthrough; S3 SAVAGE FRAMEBUFFER DRIVER: Use fallthrough; SIS FRAMEBUFFER DRIVER: Use fallthrough; VIA UNICHROME(PRO)/CHROME9 FRAMEBUFFER DRIVER: Use fallthrough; CAN NETWORK DRIVERS: Use fallthrough; WOLFSON MICROELECTRONICS DRIVERS: Use fallthrough; DIALOG SEMICONDUCTOR DRIVERS: Use fallthrough; MEDIA INPUT INFRASTRUCTURE (V4L/DVB): Use fallthrough; CAN NETWORK LAYER: Use fallthrough; HSI SUBSYSTEM: Use fallthrough; MULTIMEDIA CARD (MMC), SECURE DIGITAL (SD) AND SDIO SUBSYSTEM: Use fallthrough; KERNEL NFSD, SUNRPC, AND LOCKD SERVERS: Use fallthrough; SOFTWARE RAID (Multiple Disks) SUPPORT: Use fallthrough; SPI SUBSYSTEM: Use fallthrough; NFS, SUNRPC, AND LOCKD CLIENTS: Use fallthrough; TC subsystem: Use fallthrough; FCOE SUBSYSTEM (libfc, libfcoe, fcoe): Use fallthrough; CRYPTO API: Use fallthrough; BPF (Safe dynamic programs and tools): Use fallthrough; OMAP USB SUPPORT: Use fallthrough; OMAP1 SUPPORT: Use fallthrough; OMAP2+ SUPPORT: Use fallthrough; ARM SUB-ARCHITECTURES: Use fallthrough; M68K ON APPLE MACINTOSH: Use fallthrough; PCI SUBSYSTEM: Use fallthrough; FPU EMULATOR: Use fallthrough; X86 MM: Use fallthrough; NETWORKING [IPv4/IPv6]: Use fallthrough; USER-MODE LINUX (UML): Use fallthrough; ACPI COMPONENT ARCHITECTURE (ACPICA): Use fallthrough; FIRMWARE LOADER (request_firmware): Use fallthrough; ATA OVER ETHERNET (AOE) DRIVER: Use fallthrough; DRBD DRIVER: Use fallthrough; PARIDE DRIVERS FOR PARALLEL PORT IDE DEVICES: Use fallthrough; FLASH ADAPTER DRIVER (IBM Flash Adapter 900GB Full Height PCI Flash Card): Use fallthrough; XEN BLOCK SUBSYSTEM: Use fallthrough; AGPGART DRIVER: Use fallthrough; IPMI SUBSYSTEM: Use fallthrough; TI DAVINCI SERIES CLOCK DRIVER: Use fallthrough; H8/300 ARCHITECTURE: Use fallthrough; VIRTIO CORE AND NET DRIVERS: Use fallthrough; DRM DRIVERS AND MISC GPU PATCHES: Use fallthrough; USB HID/HIDBP DRIVERS (USB KEYBOARDS, MICE, REMOTE CONTROLS, ...): Use fallthrough; ARM/CORESIGHT FRAMEWORK AND DRIVERS: Use fallthrough; INTEL(R) TRACE HUB: Use fallthrough; I2C SUBSYSTEM HOST DRIVERS: Use fallthrough; ARM/ZYNQ ARCHITECTURE: Use fallthrough; I3C DRIVER FOR SYNOPSYS DESIGNWARE: Use fallthrough; SENSIRION SPS30 AIR POLLUTION SENSOR DRIVER: Use fallthrough; IIO DIGITAL POTENTIOMETER DAC: Use fallthrough; ASAHI KASEI AK8974 DRIVER: Use fallthrough; HFI1 DRIVER: Use fallthrough; RDMAVT - RDMA verbs software: Use fallthrough; ISCSI EXTENSIONS FOR RDMA (ISER) TARGET: Use fallthrough; OPA-VNIC DRIVER: Use fallthrough; FLYSKY FSIA6B RC RECEIVER: Use fallthrough; WACOM PROTOCOL 4 SERIAL TABLETS: Use fallthrough; ATMEL MAXTOUCH DRIVER: Use fallthrough; ISDN/mISDN SUBSYSTEM: Use fallthrough; BCACHE (BLOCK LAYER CACHE): Use fallthrough; DEVICE-MAPPER (LVM): Use fallthrough; TI FLASH MEDIA MEMORYSTICK/MMC DRIVERS: Use fallthrough; LSILOGIC MPT FUSION DRIVERS (FC/SAS/SPI): Use fallthrough; INTEL MIC DRIVERS (mic): Use fallthrough; SGI GRU DRIVER: Use fallthrough; SGI XP/XPC/XPNET DRIVER: Use fallthrough; BLOCK2MTD DRIVER: Use fallthrough; PHRAM MTD DRIVER: Use fallthrough; NAND FLASH SUBSYSTEM: Use fallthrough; SPI NOR SUBSYSTEM: Use fallthrough; UNSORTED BLOCK IMAGES (UBI): Use fallthrough; APPLETALK NETWORK LAYER: Use fallthrough; ARCNET NETWORK LAYER: Use fallthrough; BONDING DRIVER: Use fallthrough; MEDIATEK SWITCH DRIVER: Use fallthrough; NETWORKING [DSA]: Use fallthrough; BAYCOM/HDLCDRV DRIVERS FOR AX.25: Use fallthrough; IEEE 802.15.4 SUBSYSTEM: Use fallthrough; NETDEVSIM: Use fallthrough; ANALOG DEVICES INC ADIN DRIVER: Use fallthrough; PTP HARDWARE CLOCK SUPPORT: Use fallthrough; SFF/SFP/SFP+ MODULE SUPPORT: Use fallthrough; ETHERNET PHY LIBRARY: Use fallthrough; USB LAN78XX ETHERNET DRIVER: Use fallthrough; USB PEGASUS DRIVER: Use fallthrough; USB RTL8150 DRIVER: Use fallthrough; USB "USBNET" DRIVER FRAMEWORK: Use fallthrough; USB NETWORKING DRIVERS: Use fallthrough; FRAME RELAY DLCI/FRAD (Sangoma drivers too): Use fallthrough; MAC80211: Use fallthrough; RAYLINK/WEBGEAR 802.11 WIRELESS LAN DRIVER: Use fallthrough; NETWORKING DRIVERS (WIRELESS): Use fallthrough; XEN NETWORK BACKEND DRIVER: Use fallthrough; NVM EXPRESS FC TRANSPORT DRIVERS: Use fallthrough; NVM EXPRESS DRIVER: Use fallthrough; NVM EXPRESS TARGET DRIVER: Use fallthrough; ACPI: Use fallthrough; BROADCOM BCM2711/BCM2835 ARM ARCHITECTURE: Use fallthrough; PIN CONTROLLER - INTEL: Use fallthrough; MIPS: Use fallthrough; X86 PLATFORM DRIVERS: Use fallthrough; CHROMEOS EC SUBDRIVERS: Use fallthrough; POWER SUPPLY CLASS/SUBSYSTEM and DRIVERS: Use fallthrough; S390 COMMON I/O LAYER: Use fallthrough; S390 VFIO AP DRIVER: Use fallthrough; S390 ZCRYPT DRIVER: Use fallthrough; S390 IUCV NETWORK LAYER: Use fallthrough; S390 NETWORK DRIVERS: Use fallthrough; S390 ZFCP DRIVER: Use fallthrough; AACRAID SCSI RAID DRIVER: Use fallthrough; AIC7XXX / AIC79XX SCSI DRIVER: Use fallthrough; NCR 5380 SCSI DRIVERS: Use fallthrough; Emulex 10Gbps iSCSI - OneConnect DRIVER: Use fallthrough; BROCADE BFA FC SCSI DRIVER: Use fallthrough; BROADCOM BNX2FC 10 GIGABIT FCOE DRIVER: Use fallthrough; CXGB3 ISCSI DRIVER (CXGB3I): Use fallthrough; CXGB4 ISCSI DRIVER (CXGB4I): Use fallthrough; CXLFLASH (IBM Coherent Accelerator Processor Interface CAPI Flash) SCSI DRIVER: Use fallthrough; HISILICON SAS Controller: Use fallthrough; IBM Power Virtual SCSI Device Target Driver: Use fallthrough; INTEL C600 SERIES SAS CONTROLLER DRIVER: Use fallthrough; EMULEX/BROADCOM LPFC FC/FCOE SCSI DRIVER: Use fallthrough; MEGARAID SCSI/SAS DRIVERS: Use fallthrough; NINJA SCSI-3 / NINJA SCSI-32Bi (16bit/CardBus) PCMCIA SCSI HOST ADAPTER DRIVER: Use fallthrough; QLOGIC QLA2XXX FC-SCSI DRIVER: Use fallthrough; QLOGIC QLA4XXX iSCSI DRIVER: Use fallthrough; MICROSEMI SMART ARRAY SMARTPQI DRIVER (smartpqi): Use fallthrough; LSILOGIC/SYMBIOS/NCR 53C8XX and 53C1010 PCI-SCSI drivers: Use fallthrough; UNIVERSAL FLASH STORAGE HOST CONTROLLER DRIVER: Use fallthrough; STAGING - COMEDI: Use fallthrough; DPAA2 ETHERNET SWITCH DRIVER: Use fallthrough; QLOGIC QLGE 10Gb ETHERNET DRIVER: Use fallthrough; STAGING - REALTEK RTL8188EU DRIVERS: Use fallthrough; STAGING - REALTEK RTL8712U DRIVERS: Use fallthrough; STAGING - VIA VT665X DRIVERS: Use fallthrough; HYPERVISOR VIRTUAL CONSOLE DRIVER: Use fallthrough; SYNOPSYS ARC ARCHITECTURE: Use fallthrough; MICROCHIP AT91 SERIAL DRIVER: Use fallthrough; KGDB / KDB /debug_core: Use fallthrough; TEGRA SERIAL DRIVER: Use fallthrough; TTY LAYER: Use fallthrough; CONEXANT ACCESSRUNNER USB DRIVER: Use fallthrough; USB CYPRESS C67X00 DRIVER: Use fallthrough; DESIGNWARE USB3 DRD IP DRIVER: Use fallthrough; USB GADGET/PERIPHERAL SUBSYSTEM: Use fallthrough; USB EHCI DRIVER: Use fallthrough; USB ISP116X DRIVER: Use fallthrough; USB OHCI DRIVER: Use fallthrough; USB XHCI DRIVER: Use fallthrough; MEDIATEK USB3 DRD IP DRIVER: Use fallthrough; MUSB MULTIPOINT HIGH SPEED DUAL-ROLE CONTROLLER: Use fallthrough; USB PHY LAYER: Use fallthrough; USB SERIAL SUBSYSTEM: Use fallthrough; USB ATTACHED SCSI: Use fallthrough; USB MASS STORAGE DRIVER: Use fallthrough; USB OVER IP DRIVER: Use fallthrough; ADP8860 BACKLIGHT DRIVER (ADP8860/ADP8861/ADP8863): Use fallthrough; MICROCHIP LCDFB DRIVER: Use fallthrough; FREESCALE DIU FRAMEBUFFER DRIVER: Use fallthrough; SAMSUNG FRAMEBUFFER DRIVER: Use fallthrough; VIRTUAL BOX GUEST DEVICE DRIVER: Use fallthrough; FANOTIFY: Use fallthrough; MULTIFUNCTION DEVICES (MFD): Use fallthrough; MULTIPLEXER SUBSYSTEM: Use fallthrough; VOLTAGE AND CURRENT REGULATOR FRAMEWORK: Use fallthrough; RESET CONTROLLER FRAMEWORK: Use fallthrough; SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC): Use fallthrough; CEPH COMMON CODE (LIBCEPH): Use fallthrough; GPIO SUBSYSTEM: Use fallthrough; GREYBUS SUBSYSTEM: Use fallthrough; INPUT (KEYBOARD, MOUSE, JOYSTICK, TOUCHSCREEN) DRIVERS: Use fallthrough; MEMORY TECHNOLOGY DEVICES (MTD): Use fallthrough; ETHERNET BRIDGE: Use fallthrough; PIN CONTROL SUBSYSTEM: Use fallthrough; NFC SUBSYSTEM: Use fallthrough; REAL TIME CLOCK (RTC) SUBSYSTEM: Use fallthrough; CPU FREQUENCY SCALING FRAMEWORK: Use fallthrough; NOHZ, DYNTICKS SUPPORT: Use fallthrough; SONICS SILICON BACKPLANE DRIVER (SSB): Use fallthrough; USB SUBSYSTEM: Use fallthrough; 9P FILE SYSTEM: Use fallthrough; BLUETOOTH SUBSYSTEM: Use fallthrough; NETFILTER: Use fallthrough; PHONET PROTOCOL: Use fallthrough; SCTP PROTOCOL: Use fallthrough; AFS FILESYSTEM: Use fallthrough; EROFS FILE SYSTEM: Use fallthrough; F2FS FILE SYSTEM: Use fallthrough; FILESYSTEM DIRECT ACCESS (DAX): Use fallthrough; HARDWARE MONITORING: Use fallthrough; INFINIBAND SUBSYSTEM: Use fallthrough; NILFS2 FILESYSTEM: Use fallthrough; RXRPC SOCKETS (AF_RXRPC): Use fallthrough; ATM: Use fallthrough; AUDIT SUBSYSTEM: Use fallthrough; BTRFS FILE SYSTEM: Use fallthrough; CAPABILITIES: Use fallthrough; HEWLETT-PACKARD SMART ARRAY RAID DRIVER (hpsa): Use fallthrough; DCCP PROTOCOL: Use fallthrough; DEVLINK: Use fallthrough; XFS FILESYSTEM: Use fallthrough; FRAMEBUFFER LAYER: Use fallthrough; FILE LOCKING (flock() and fcntl()/lockf()): Use fallthrough; FIREWIRE SUBSYSTEM: Use fallthrough; FILESYSTEMS (VFS and infrastructure): Use fallthrough; FUTEX SUBSYSTEM: Use fallthrough; HID CORE LAYER: Use fallthrough; I2C SUBSYSTEM: Use fallthrough; NETWORKING DRIVERS: Use fallthrough; NETWORKING [GENERAL]: Use fallthrough; KEYS/KEYRINGS: Use fallthrough; LIGHTNVM PLATFORM SUPPORT: Use fallthrough; PARALLEL PORT SUBSYSTEM: Use fallthrough; SCHEDULER: Use fallthrough; SELINUX SECURITY MODULE: Use fallthrough; THERMAL: Use fallthrough; TIMEKEEPING, CLOCKSOURCE CORE, NTP, ALARMTIMER: Use fallthrough; VFIO DRIVER: Use fallthrough; VIRTIO BLOCK AND SCSI DRIVERS: Use fallthrough; WATCHDOG DEVICE DRIVERS: Use fallthrough; SOUND: Use fallthrough; INTEGRITY MEASUREMENT ARCHITECTURE (IMA): Use fallthrough; ALPHA PORT: Use fallthrough; ARM PORT: Use fallthrough; ARM64 PORT (AARCH64 ARCHITECTURE): Use fallthrough; C6X ARCHITECTURE: Use fallthrough; C-SKY ARCHITECTURE: Use fallthrough; QUALCOMM HEXAGON ARCHITECTURE: Use fallthrough; IA64 (Itanium) PLATFORM: Use fallthrough; MICROBLAZE ARCHITECTURE: Use fallthrough; ANDES ARCHITECTURE: Use fallthrough; OPENRISC ARCHITECTURE: Use fallthrough; RISC-V ARCHITECTURE: Use fallthrough; S390: Use fallthrough; SUPERH: Use fallthrough; SPARC + UltraSPARC (sparc/sparc64): Use fallthrough; UNICORE32 ARCHITECTURE: Use fallthrough; X86 ARCHITECTURE (32-BIT AND 64-BIT): Use fallthrough; TENSILICA XTENSA PORT (xtensa): Use fallthrough; ASYNCHRONOUS TRANSFERS/TRANSFORMS (IOAT) API: Use fallthrough; LIBATA SATA AHCI PLATFORM devices support: Use fallthrough; LIBATA PATA DRIVERS: Use fallthrough; LIBATA SATA PROMISE TX2/TX4 CONTROLLER DRIVER: Use fallthrough; LIBATA SUBSYSTEM (Serial and Parallel ATA drivers): Use fallthrough; PARALLEL LCD/KEYPAD PANEL DRIVER: Use fallthrough; FLOPPY DRIVER: Use fallthrough; RADOS BLOCK DEVICE (RBD): Use fallthrough; STEC S1220 SKD DRIVER: Use fallthrough; BLOCK LAYER: Use fallthrough; BLUETOOTH DRIVERS: Use fallthrough; CHAR and MISC DRIVERS: Use fallthrough; COMMON CLK FRAMEWORK: Use fallthrough; EDAC-AMD64: Use fallthrough; EDAC-PND2: Use fallthrough; IDE/ATAPI DRIVERS: Use fallthrough; IDE SUBSYSTEM: Use fallthrough; IIO SUBSYSTEM AND DRIVERS: Use fallthrough; AMD IOMMU (AMD-VI): Use fallthrough; ARM SMMU DRIVERS: Use fallthrough; INTEL IOMMU (VT-d): Use fallthrough; VIRTIO IOMMU DRIVER: Use fallthrough; IRQCHIP DRIVERS: Use fallthrough; OMAP GENERAL PURPOSE MEMORY CONTROLLER SUPPORT: Use fallthrough; SONY MEMORYSTICK SUBSYSTEM: Use fallthrough; NTB DRIVER CORE: Use fallthrough; PCMCIA SUBSYSTEM: Use fallthrough; RAPIDIO SUBSYSTEM: Use fallthrough; 53C700 AND 53C700-66 SCSI DRIVER: Use fallthrough; BUSLOGIC SCSI DRIVER: Use fallthrough; ATTO EXPRESSSAS SAS/SATA RAID SCSI DRIVER: Use fallthrough; MYLEX DAC960 PCI RAID Controller: Use fallthrough; SCSI CDROM DRIVER: Use fallthrough; SCSI TAPE DRIVER: Use fallthrough; VMware PVSCSI driver: Use fallthrough; SCSI SUBSYSTEM: Use fallthrough; STAGING SUBSYSTEM: Use fallthrough; SCSI TARGET SUBSYSTEM: Use fallthrough; THUNDERBOLT DRIVER: Use fallthrough; M68K ARCHITECTURE: Use fallthrough; EXTRA BOOT CONFIG: Use fallthrough; HIGH-RESOLUTION TIMERS, CLOCKEVENTS: Use fallthrough; DYNAMIC INTERRUPT MODERATION: Use fallthrough; MEMORY MANAGEMENT: Use fallthrough; SIPHASH PRF ROUTINES: Use fallthrough; POSIX CLOCKS and TIMERS: Use fallthrough; GCOV BASED KERNEL PROFILING: Use fallthrough; IRQ SUBSYSTEM: Use fallthrough; APPARMOR SECURITY MODULE: Use fallthrough; SMACK SECURITY MODULE: Use fallthrough; TOMOYO SECURITY MODULE: Use fallthrough; VSPRINTF: Use fallthrough; THE REST: Use fallthrough; arch/alpha/kernel/module.c | 2 +- arch/alpha/kernel/signal.c | 2 +- arch/alpha/kernel/traps.c | 6 +- arch/arc/kernel/disasm.c | 3 +- arch/arc/kernel/signal.c | 3 +- arch/arc/kernel/unwind.c | 6 +- arch/arm/kernel/hw_breakpoint.c | 10 +- arch/arm/kernel/signal.c | 2 +- arch/arm/mach-ep93xx/crunch.c | 3 +- arch/arm/mach-mmp/pm-mmp2.c | 8 +- arch/arm/mach-mmp/pm-pxa910.c | 10 +- arch/arm/mach-omap2/id.c | 8 -- arch/arm/mach-omap2/omap_device.c | 2 +- arch/arm/mach-orion5x/dns323-setup.c | 2 +- arch/arm/mach-rpc/riscpc.c | 2 +- arch/arm/mach-tegra/reset.c | 2 +- arch/arm/mm/alignment.c | 6 +- arch/arm/plat-omap/dma.c | 6 +- arch/arm/probes/decode.c | 2 +- arch/arm/probes/kprobes/core.c | 2 +- arch/arm64/kernel/cpufeature.c | 2 +- arch/arm64/kernel/cpuinfo.c | 2 +- arch/arm64/kernel/hw_breakpoint.c | 8 +- arch/arm64/kernel/module.c | 8 +- arch/arm64/kernel/smp.c | 2 +- arch/arm64/kvm/handle_exit.c | 2 +- arch/arm64/kvm/hyp/debug-sr.c | 60 ++++----- arch/arm64/mm/context.c | 2 +- arch/c6x/kernel/signal.c | 5 +- arch/csky/kernel/signal.c | 2 +- arch/h8300/kernel/signal.c | 2 +- arch/hexagon/kernel/module.c | 2 +- arch/hexagon/kernel/signal.c | 2 +- arch/ia64/kernel/crash.c | 2 +- arch/ia64/kernel/module.c | 2 +- arch/ia64/kernel/perfmon.c | 2 +- arch/ia64/kernel/signal.c | 2 +- arch/ia64/kernel/unaligned.c | 6 +- arch/ia64/kernel/unwind.c | 2 +- arch/m68k/amiga/config.c | 4 +- arch/m68k/atari/atakeyb.c | 3 +- arch/m68k/kernel/signal.c | 2 +- arch/m68k/mac/config.c | 2 +- arch/m68k/mac/via.c | 2 +- arch/m68k/mm/fault.c | 2 +- arch/microblaze/kernel/signal.c | 2 +- arch/mips/alchemy/devboards/db1550.c | 2 +- arch/mips/ar7/setup.c | 2 +- arch/mips/ath79/setup.c | 3 +- arch/mips/bcm63xx/cpu.c | 2 +- arch/mips/bcm63xx/dev-flash.c | 2 +- arch/mips/cavium-octeon/executive/cvmx-pko.c | 2 +- arch/mips/cavium-octeon/octeon-platform.c | 4 +- arch/mips/cavium-octeon/octeon-usb.c | 2 +- arch/mips/dec/tc.c | 2 +- arch/mips/include/asm/fpu.h | 2 +- arch/mips/include/asm/octeon/cvmx-sli-defs.h | 2 +- arch/mips/include/asm/page.h | 2 +- arch/mips/include/asm/unroll.h | 64 +++++----- arch/mips/kernel/branch.c | 26 ++-- arch/mips/kernel/cpu-probe.c | 22 ++-- arch/mips/kernel/idle.c | 2 +- arch/mips/kernel/mips-r2-to-r6-emul.c | 2 +- arch/mips/kernel/perf_event_mipsxx.c | 6 +- arch/mips/kernel/signal.c | 2 +- arch/mips/kernel/traps.c | 3 +- arch/mips/kernel/watch.c | 26 ++-- arch/mips/kvm/emulate.c | 8 +- arch/mips/math-emu/cp1emu.c | 28 ++--- arch/mips/math-emu/dp_add.c | 3 +- arch/mips/math-emu/dp_div.c | 3 +- arch/mips/math-emu/dp_fmax.c | 6 +- arch/mips/math-emu/dp_fmin.c | 6 +- arch/mips/math-emu/dp_maddf.c | 3 +- arch/mips/math-emu/dp_mul.c | 3 +- arch/mips/math-emu/dp_sqrt.c | 5 +- arch/mips/math-emu/dp_sub.c | 3 +- arch/mips/math-emu/sp_add.c | 3 +- arch/mips/math-emu/sp_div.c | 3 +- arch/mips/math-emu/sp_fdp.c | 3 +- arch/mips/math-emu/sp_fmax.c | 6 +- arch/mips/math-emu/sp_fmin.c | 6 +- arch/mips/math-emu/sp_maddf.c | 3 +- arch/mips/math-emu/sp_mul.c | 3 +- arch/mips/math-emu/sp_sub.c | 3 +- arch/mips/mm/c-r4k.c | 6 +- arch/mips/mm/tlbex.c | 2 +- arch/mips/oprofile/op_model_mipsxx.c | 26 ++-- arch/mips/pci/fixup-sni.c | 3 +- arch/mips/pci/ops-bcm63xx.c | 2 +- arch/nds32/kernel/fpu.c | 12 +- arch/nds32/kernel/signal.c | 4 +- arch/openrisc/kernel/signal.c | 2 +- arch/parisc/kernel/signal.c | 2 +- arch/parisc/kernel/traps.c | 10 +- arch/parisc/mm/fault.c | 5 +- arch/powerpc/kernel/align.c | 8 +- arch/powerpc/kvm/book3s_32_mmu.c | 2 +- arch/powerpc/kvm/book3s_64_mmu.c | 2 +- arch/powerpc/kvm/book3s_pr.c | 2 +- arch/powerpc/kvm/booke.c | 6 +- arch/powerpc/kvm/powerpc.c | 1 - arch/powerpc/platforms/cell/spufs/switch.c | 2 +- arch/powerpc/platforms/powermac/feature.c | 2 +- arch/powerpc/platforms/powernv/opal-async.c | 2 +- arch/powerpc/platforms/pseries/hvcserver.c | 2 +- arch/powerpc/xmon/xmon.c | 2 +- arch/riscv/kernel/signal.c | 2 +- arch/riscv/net/bpf_jit_comp32.c | 5 +- arch/s390/kernel/signal.c | 4 +- arch/s390/kernel/topology.c | 2 +- arch/s390/kvm/gaccess.c | 23 ++-- arch/s390/kvm/interrupt.c | 2 +- arch/s390/kvm/kvm-s390.c | 4 +- arch/s390/mm/fault.c | 13 +- arch/s390/mm/gmap.c | 6 +- arch/s390/mm/pgalloc.c | 2 +- arch/sh/drivers/platform_early.c | 2 +- arch/sh/kernel/disassemble.c | 4 +- arch/sh/kernel/kgdb.c | 2 +- arch/sh/kernel/signal_32.c | 2 +- arch/sh/kernel/signal_64.c | 2 +- arch/sparc/kernel/auxio_64.c | 1 - arch/sparc/kernel/central.c | 2 +- arch/sparc/kernel/kgdb_32.c | 3 +- arch/sparc/kernel/kgdb_64.c | 3 +- arch/sparc/kernel/pcr.c | 2 +- arch/sparc/kernel/prom_32.c | 3 +- arch/sparc/kernel/signal32.c | 4 +- arch/sparc/kernel/signal_32.c | 4 +- arch/sparc/kernel/signal_64.c | 4 +- arch/sparc/math-emu/math_32.c | 8 +- arch/sparc/net/bpf_jit_comp_32.c | 2 +- arch/um/kernel/signal.c | 2 +- arch/unicore32/kernel/signal.c | 2 +- arch/x86/boot/cmdline.c | 6 +- arch/x86/boot/compressed/kaslr.c | 2 +- arch/x86/events/intel/core.c | 7 +- arch/x86/events/intel/lbr.c | 2 +- arch/x86/kernel/alternative.c | 4 +- arch/x86/kernel/apic/io_apic.c | 4 +- arch/x86/kernel/apic/probe_32.c | 2 +- arch/x86/kernel/cpu/cacheinfo.c | 2 +- arch/x86/kernel/cpu/mce/inject.c | 2 +- arch/x86/kernel/cpu/mce/intel.c | 4 +- arch/x86/kernel/cpu/mtrr/cyrix.c | 2 +- arch/x86/kernel/hw_breakpoint.c | 2 +- arch/x86/kernel/kgdb.c | 4 +- arch/x86/kernel/mpparse.c | 4 +- arch/x86/kernel/ptrace.c | 3 +- arch/x86/kernel/reboot.c | 3 +- arch/x86/kernel/signal.c | 2 +- arch/x86/kernel/uprobes.c | 4 +- arch/x86/kvm/emulate.c | 2 +- arch/x86/kvm/hyperv.c | 2 +- arch/x86/kvm/irq_comm.c | 2 +- arch/x86/kvm/lapic.c | 6 +- arch/x86/kvm/mmu/mmu.c | 2 +- arch/x86/kvm/svm.c | 2 +- arch/x86/kvm/vmx/vmx.c | 15 ++- arch/x86/kvm/x86.c | 12 +- arch/x86/lib/cmdline.c | 12 +- arch/x86/lib/insn-eval.c | 8 +- arch/x86/math-emu/errors.c | 2 +- arch/x86/math-emu/fpu_trig.c | 2 +- arch/x86/mm/ioremap.c | 3 +- arch/xtensa/kernel/signal.c | 2 +- block/badblocks.c | 2 +- block/bfq-iosched.c | 4 +- block/blk-wbt.c | 2 +- crypto/drbg.c | 3 +- crypto/tcrypt.c | 114 +++++++++--------- .../accessibility/braille/braille_console.c | 2 +- drivers/acpi/ac.c | 2 +- drivers/acpi/acpi_processor.c | 3 +- drivers/acpi/acpica/dscontrol.c | 4 +- drivers/acpi/acpica/dswexec.c | 3 +- drivers/acpi/acpica/dswload.c | 4 +- drivers/acpi/acpica/dswload2.c | 4 +- drivers/acpi/acpica/exfldio.c | 4 +- drivers/acpi/acpica/exresop.c | 7 +- drivers/acpi/acpica/exstore.c | 8 +- drivers/acpi/acpica/hwgpe.c | 4 +- drivers/acpi/acpica/utdelete.c | 4 +- drivers/acpi/acpica/utprint.c | 3 +- drivers/acpi/button.c | 2 +- drivers/acpi/dock.c | 2 +- drivers/acpi/processor_idle.c | 3 +- drivers/acpi/resource.c | 2 +- drivers/acpi/spcr.c | 4 +- drivers/ata/ahci_brcm.c | 2 +- drivers/ata/libahci_platform.c | 2 +- drivers/ata/libata-core.c | 16 +-- drivers/ata/libata-eh.c | 6 +- drivers/ata/libata-scsi.c | 2 +- drivers/ata/pata_atp867x.c | 4 +- drivers/ata/pata_serverworks.c | 2 +- drivers/ata/sata_mv.c | 12 +- drivers/ata/sata_promise.c | 8 +- drivers/ata/sata_sx4.c | 2 +- drivers/atm/firestream.c | 2 +- drivers/atm/fore200e.c | 24 ++-- drivers/atm/he.c | 4 +- drivers/atm/idt77105.c | 2 +- drivers/atm/lanai.c | 2 +- drivers/atm/zatm.c | 2 +- drivers/auxdisplay/panel.c | 6 +- drivers/base/firmware_loader/fallback.c | 4 +- drivers/block/aoe/aoecmd.c | 2 +- drivers/block/ataflop.c | 2 +- drivers/block/drbd/drbd_int.h | 2 +- drivers/block/drbd/drbd_main.c | 2 +- drivers/block/drbd/drbd_nl.c | 2 +- drivers/block/drbd/drbd_receiver.c | 13 +- drivers/block/drbd/drbd_req.c | 5 +- drivers/block/floppy.c | 4 +- drivers/block/loop.c | 4 +- drivers/block/paride/pd.c | 4 +- drivers/block/rbd.c | 8 +- drivers/block/rsxx/core.c | 2 +- drivers/block/skd_main.c | 3 +- drivers/block/xen-blkback/blkback.c | 2 +- drivers/block/xen-blkfront.c | 5 +- drivers/bluetooth/bcm203x.c | 2 +- drivers/bluetooth/bluecard_cs.c | 2 - drivers/bluetooth/hci_ll.c | 2 +- drivers/bluetooth/hci_qca.c | 8 +- drivers/bus/ti-sysc.c | 2 +- drivers/char/agp/ali-agp.c | 2 +- drivers/char/ipmi/kcs_bmc.c | 3 +- drivers/char/lp.c | 4 +- drivers/char/mem.c | 2 +- drivers/char/nvram.c | 2 +- drivers/clk/davinci/pll.c | 2 +- drivers/clk/imx/clk-pllv3.c | 4 +- drivers/clk/samsung/clk-s3c2443.c | 2 +- drivers/clocksource/timer-cadence-ttc.c | 5 +- drivers/cpufreq/p4-clockmod.c | 2 +- drivers/cpufreq/speedstep-lib.c | 2 +- drivers/cpufreq/ti-cpufreq.c | 4 +- drivers/crypto/axis/artpec6_crypto.c | 3 +- drivers/crypto/cavium/cpt/cptvf_reqmanager.c | 4 +- drivers/crypto/chelsio/chcr_ktls.c | 4 +- drivers/crypto/qat/qat_common/adf_pf2vf_msg.c | 2 +- drivers/crypto/qat/qat_common/qat_uclo.c | 6 +- drivers/crypto/ux500/cryp/cryp.c | 18 +-- drivers/dma/amba-pl08x.c | 10 +- drivers/dma/fsldma.c | 2 +- drivers/dma/imx-dma.c | 2 +- drivers/dma/iop-adma.h | 12 +- drivers/dma/nbpfaxi.c | 2 +- drivers/dma/pl330.c | 13 +- drivers/dma/sh/shdma-base.c | 2 +- drivers/edac/amd64_edac.c | 2 +- drivers/edac/pnd2_edac.c | 2 +- drivers/firewire/core-device.c | 2 +- drivers/firewire/core-iso.c | 3 +- drivers/firewire/core-topology.c | 2 +- drivers/firewire/core-transaction.c | 6 +- drivers/firewire/ohci.c | 5 +- drivers/gpio/gpio-aspeed-sgpio.c | 6 +- drivers/gpio/gpio-aspeed.c | 6 +- drivers/gpio/gpio-ath79.c | 2 +- drivers/gpio/gpio-eic-sprd.c | 4 +- drivers/gpio/gpio-stmpe.c | 4 +- drivers/gpio/gpiolib-acpi.c | 2 +- .../drm/amd/amdgpu/amdgpu_amdkfd_arcturus.c | 2 +- drivers/gpu/drm/amd/amdgpu/gfx_v8_0.c | 3 +- drivers/gpu/drm/amd/amdgpu/gmc_v9_0.c | 2 +- drivers/gpu/drm/amd/amdgpu/si_dpm.c | 4 +- .../drm/amd/display/dc/bios/bios_parser2.c | 4 +- drivers/gpu/drm/amd/display/dc/dce/dce_aux.c | 2 +- .../drm/amd/display/dc/dce/dce_mem_input.c | 2 +- .../gpu/drm/amd/powerplay/hwmgr/smu7_hwmgr.c | 6 +- drivers/gpu/drm/arm/malidp_hw.c | 6 +- drivers/gpu/drm/ast/ast_main.c | 2 +- .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 4 +- drivers/gpu/drm/bridge/ti-sn65dsi86.c | 6 +- drivers/gpu/drm/drm_bufs.c | 2 +- drivers/gpu/drm/drm_dp_helper.c | 2 +- drivers/gpu/drm/drm_modes.c | 2 +- drivers/gpu/drm/drm_vm.c | 4 +- drivers/gpu/drm/exynos/exynos_drm_dsi.c | 10 +- drivers/gpu/drm/fsl-dcu/fsl_dcu_drm_plane.c | 6 +- drivers/gpu/drm/i915/display/icl_dsi.c | 6 +- drivers/gpu/drm/i915/display/intel_bios.c | 6 +- drivers/gpu/drm/i915/display/intel_cdclk.c | 10 +- .../gpu/drm/i915/display/intel_combo_phy.c | 6 +- drivers/gpu/drm/i915/display/intel_ddi.c | 4 +- drivers/gpu/drm/i915/display/intel_display.c | 20 +-- drivers/gpu/drm/i915/display/intel_dpll_mgr.c | 8 +- drivers/gpu/drm/i915/display/intel_sdvo.c | 12 +- drivers/gpu/drm/i915/display/intel_sprite.c | 22 ++-- drivers/gpu/drm/i915/display/intel_tc.c | 2 +- drivers/gpu/drm/i915/gem/i915_gem_mman.c | 2 +- drivers/gpu/drm/i915/gem/i915_gem_pages.c | 2 +- drivers/gpu/drm/i915/gem/i915_gem_stolen.c | 6 +- drivers/gpu/drm/i915/gt/intel_engine_cs.c | 2 +- drivers/gpu/drm/i915/gt/intel_ggtt.c | 2 +- drivers/gpu/drm/i915/gt/intel_lrc.c | 2 +- .../gpu/drm/i915/gt/intel_ring_submission.c | 2 +- drivers/gpu/drm/i915/gvt/handlers.c | 2 +- drivers/gpu/drm/i915/i915_gpu_error.c | 2 +- drivers/gpu/drm/i915/i915_pmu.c | 2 +- drivers/gpu/drm/i915/intel_device_info.c | 4 +- drivers/gpu/drm/imx/ipuv3-plane.c | 3 +- drivers/gpu/drm/ingenic/ingenic-drm.c | 2 +- drivers/gpu/drm/meson/meson_crtc.c | 4 +- drivers/gpu/drm/meson/meson_osd_afbcd.c | 2 +- drivers/gpu/drm/meson/meson_overlay.c | 4 +- drivers/gpu/drm/msm/adreno/a5xx_gpu.c | 4 +- drivers/gpu/drm/msm/adreno/a6xx_gmu.c | 2 +- drivers/gpu/drm/msm/adreno/a6xx_gpu.c | 2 +- drivers/gpu/drm/msm/adreno/adreno_gpu.c | 2 +- drivers/gpu/drm/nouveau/dispnv50/disp.c | 2 +- drivers/gpu/drm/nouveau/nouveau_bo.c | 2 +- drivers/gpu/drm/nouveau/nouveau_connector.c | 4 +- .../gpu/drm/nouveau/nvkm/engine/disp/hdmi.c | 34 +++--- .../drm/nouveau/nvkm/engine/dma/usernv04.c | 2 +- .../gpu/drm/nouveau/nvkm/engine/fifo/nv04.c | 4 +- .../gpu/drm/nouveau/nvkm/engine/fifo/nv40.c | 2 +- .../gpu/drm/nouveau/nvkm/subdev/bios/dcb.c | 3 +- drivers/gpu/drm/nouveau/nvkm/subdev/bios/dp.c | 2 +- .../gpu/drm/nouveau/nvkm/subdev/bios/perf.c | 2 +- .../gpu/drm/nouveau/nvkm/subdev/bios/pll.c | 2 +- .../gpu/drm/nouveau/nvkm/subdev/bios/timing.c | 10 +- .../gpu/drm/nouveau/nvkm/subdev/clk/base.c | 2 +- .../gpu/drm/nouveau/nvkm/subdev/clk/mcp77.c | 2 +- .../drm/nouveau/nvkm/subdev/devinit/nv04.c | 18 ++- .../gpu/drm/nouveau/nvkm/subdev/fb/ramnv40.c | 4 +- .../gpu/drm/nouveau/nvkm/subdev/mxm/nv50.c | 2 +- drivers/gpu/drm/omapdrm/dss/venc.c | 2 +- drivers/gpu/drm/radeon/ci_dpm.c | 4 +- drivers/gpu/drm/radeon/r300.c | 4 +- drivers/gpu/drm/radeon/r420.c | 2 +- drivers/gpu/drm/radeon/r600_cs.c | 4 +- drivers/gpu/drm/radeon/radeon_uvd.c | 3 +- drivers/gpu/drm/radeon/si_dpm.c | 4 +- drivers/gpu/drm/radeon/uvd_v1_0.c | 3 +- drivers/gpu/drm/savage/savage_state.c | 10 +- drivers/gpu/drm/sti/sti_hdmi.c | 6 +- drivers/gpu/drm/sun4i/sun4i_tcon.c | 4 +- drivers/gpu/drm/sun4i/sun6i_mipi_dsi.c | 3 +- drivers/gpu/drm/tegra/dc.c | 2 +- drivers/gpu/drm/tilcdc/tilcdc_crtc.c | 2 +- drivers/gpu/drm/ttm/ttm_bo_vm.c | 2 +- drivers/gpu/drm/via/via_dmablit.c | 8 +- drivers/gpu/drm/xen/xen_drm_front.c | 3 - drivers/gpu/ipu-v3/ipu-dc.c | 2 +- drivers/greybus/es2.c | 2 +- drivers/greybus/interface.c | 2 +- drivers/hid/hid-lg-g15.c | 2 +- drivers/hid/hid-logitech-dj.c | 2 +- drivers/hid/hid-microsoft.c | 3 - drivers/hid/hid-rmi.c | 1 - drivers/hid/hid-roccat-kone.c | 2 +- drivers/hid/hid-uclogic-params.c | 2 +- drivers/hid/hid-wiimote-core.c | 2 - drivers/hid/usbhid/hiddev.c | 1 - drivers/hid/wacom_wac.c | 47 +++----- drivers/hsi/clients/ssi_protocol.c | 4 +- drivers/hsi/controllers/omap_ssi_core.c | 2 +- drivers/hv/hv_kvp.c | 4 +- drivers/hv/vmbus_drv.c | 2 +- drivers/hwmon/adt7462.c | 8 +- drivers/hwmon/emc1403.c | 4 +- drivers/hwmon/f71882fg.c | 4 +- drivers/hwmon/hwmon-vid.c | 4 +- drivers/hwmon/ina3221.c | 3 +- drivers/hwmon/nct6775.c | 2 +- drivers/hwmon/occ/common.c | 6 +- drivers/hwmon/w83627hf.c | 2 +- drivers/hwmon/w83781d.c | 2 +- drivers/hwmon/w83795.c | 2 +- .../hwtracing/coresight/coresight-cpu-debug.c | 4 +- drivers/hwtracing/coresight/coresight-etm4x.c | 1 - .../hwtracing/coresight/coresight-platform.c | 3 +- drivers/hwtracing/coresight/coresight-tmc.c | 2 - drivers/hwtracing/intel_th/sth.c | 6 +- drivers/i2c/busses/i2c-amd8111.c | 2 +- drivers/i2c/busses/i2c-aspeed.c | 4 +- drivers/i2c/busses/i2c-designware-pcidrv.c | 2 +- drivers/i2c/busses/i2c-digicolor.c | 2 +- drivers/i2c/busses/i2c-i801.c | 8 +- drivers/i2c/busses/i2c-mv64xxx.c | 9 +- drivers/i2c/busses/i2c-omap.c | 1 - drivers/i2c/busses/i2c-opal.c | 2 +- drivers/i2c/busses/i2c-s3c2410.c | 3 +- drivers/i2c/busses/i2c-synquacer.c | 4 +- drivers/i2c/busses/i2c-viapro.c | 2 +- drivers/i2c/busses/scx200_acb.c | 3 +- drivers/i2c/i2c-slave-eeprom.c | 2 +- drivers/i3c/master/dw-i3c-master.c | 2 +- drivers/ide/hpt366.c | 6 +- drivers/ide/ide-cd.c | 4 +- drivers/ide/ide-floppy.c | 2 +- drivers/ide/ide-probe.c | 2 +- drivers/ide/ide-taskfile.c | 12 +- drivers/ide/sis5513.c | 2 +- drivers/iio/accel/mma8452.c | 2 +- drivers/iio/adc/ab8500-gpadc.c | 2 +- drivers/iio/adc/cpcap-adc.c | 2 +- drivers/iio/chemical/sps30.c | 2 +- drivers/iio/dac/ad5592r-base.c | 2 - drivers/iio/dac/dpot-dac.c | 2 +- drivers/iio/health/max30102.c | 4 +- drivers/iio/imu/adis.c | 6 +- drivers/iio/industrialio-core.c | 2 +- drivers/iio/light/si1145.c | 2 +- drivers/iio/magnetometer/ak8974.c | 2 +- drivers/infiniband/core/cm.c | 14 +-- drivers/infiniband/core/cma.c | 3 +- drivers/infiniband/core/ucma.c | 4 +- drivers/infiniband/core/uverbs_ioctl.c | 5 +- drivers/infiniband/hw/bnxt_re/ib_verbs.c | 2 +- drivers/infiniband/hw/bnxt_re/qplib_fp.c | 2 +- drivers/infiniband/hw/cxgb4/cm.c | 4 +- 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fs/nfsd/nfs4layouts.c | 2 +- fs/nfsd/nfs4proc.c | 2 +- fs/nfsd/nfs4state.c | 12 +- fs/nfsd/nfsfh.c | 4 +- fs/nfsd/nfsproc.c | 3 +- fs/nfsd/nfssvc.c | 2 +- fs/nfsd/vfs.c | 4 +- fs/nilfs2/bmap.c | 2 +- fs/nilfs2/recovery.c | 2 +- fs/nilfs2/segment.c | 19 +-- fs/notify/fanotify/fanotify_user.c | 2 +- fs/ocfs2/cluster/quorum.c | 2 +- fs/quota/quota.c | 2 +- fs/seq_file.c | 2 +- fs/signalfd.c | 2 +- fs/ubifs/lprops.c | 4 +- fs/udf/symlink.c | 2 +- fs/ufs/util.h | 12 +- fs/vboxsf/utils.c | 2 +- fs/xfs/libxfs/xfs_ag_resv.c | 4 +- fs/xfs/libxfs/xfs_alloc.c | 2 +- fs/xfs/libxfs/xfs_da_btree.c | 2 +- fs/xfs/libxfs/xfs_inode_buf.c | 4 +- fs/xfs/scrub/bmap.c | 2 +- fs/xfs/scrub/btree.c | 2 +- fs/xfs/scrub/common.c | 6 +- fs/xfs/scrub/dabtree.c | 2 +- fs/xfs/scrub/repair.c | 2 +- fs/xfs/xfs_bmap_util.c | 3 +- fs/xfs/xfs_export.c | 4 +- fs/xfs/xfs_file.c | 2 +- fs/xfs/xfs_fsmap.c | 2 +- fs/xfs/xfs_inode.c | 2 +- fs/xfs/xfs_ioctl.c | 4 +- fs/xfs/xfs_iomap.c | 2 +- fs/xfs/xfs_log.c | 2 +- fs/xfs/xfs_trans_buf.c | 2 +- fs/zonefs/super.c | 2 +- include/linux/compat.h | 6 +- include/linux/filter.h | 2 +- include/linux/jhash.h | 39 ++++-- include/linux/mm.h | 9 +- include/linux/signal.h | 12 +- include/linux/skbuff.h | 12 +- include/math-emu/op-common.h | 6 +- init/do_mounts_md.c | 4 +- ipc/sem.c | 4 +- ipc/shm.c | 4 +- kernel/auditfilter.c | 2 +- kernel/bpf/cgroup.c | 4 +- kernel/bpf/syscall.c | 2 +- kernel/bpf/verifier.c | 4 +- kernel/capability.c | 2 +- kernel/compat.c | 6 +- kernel/debug/gdbstub.c | 6 +- kernel/debug/kdb/kdb_keyboard.c | 4 +- kernel/debug/kdb/kdb_support.c | 6 +- kernel/events/core.c | 3 +- kernel/futex.c | 4 +- kernel/gcov/gcc_3_4.c | 6 +- kernel/irq/handle.c | 3 +- kernel/irq/manage.c | 5 +- kernel/kallsyms.c | 4 +- kernel/power/hibernate.c | 2 +- kernel/power/qos.c | 4 +- kernel/sched/core.c | 2 +- kernel/sched/topology.c | 6 +- kernel/signal.c | 2 +- kernel/sys.c | 3 +- kernel/time/hrtimer.c | 2 +- kernel/time/posix-timers.c | 4 +- kernel/time/tick-broadcast.c | 2 +- kernel/time/timer.c | 2 +- kernel/trace/blktrace.c | 2 +- kernel/trace/trace_events_filter.c | 4 +- lib/asn1_decoder.c | 6 +- lib/assoc_array.c | 2 +- lib/bootconfig.c | 4 +- lib/cmdline.c | 10 +- lib/dim/net_dim.c | 2 +- lib/dim/rdma_dim.c | 4 +- lib/glob.c | 2 +- lib/nlattr.c | 5 +- lib/siphash.c | 54 ++++++--- lib/ts_fsm.c | 3 +- lib/vsprintf.c | 18 ++- lib/xz/xz_dec_lzma2.c | 8 +- lib/xz/xz_dec_stream.c | 32 ++--- lib/zlib_inflate/inflate.c | 24 ++-- lib/zstd/bitstream.h | 10 +- lib/zstd/compress.c | 3 +- lib/zstd/decompress.c | 17 +-- lib/zstd/huf_compress.c | 4 +- mm/gup.c | 2 +- mm/hugetlb_cgroup.c | 6 +- mm/ksm.c | 3 +- mm/list_lru.c | 2 +- mm/memcontrol.c | 2 +- mm/mempolicy.c | 3 - mm/mmap.c | 5 +- mm/shmem.c | 2 +- mm/zsmalloc.c | 2 +- net/8021q/vlan_dev.c | 2 +- net/9p/trans_xen.c | 2 +- net/atm/common.c | 4 +- net/atm/lec.c | 2 +- net/atm/resources.c | 8 +- net/bluetooth/hci_event.c | 6 +- net/bluetooth/hci_sock.c | 4 +- net/bluetooth/l2cap_core.c | 22 ++-- net/bluetooth/l2cap_sock.c | 4 +- net/bluetooth/mgmt.c | 4 +- net/bluetooth/rfcomm/core.c | 3 +- net/bluetooth/rfcomm/sock.c | 3 +- net/bluetooth/smp.c | 2 +- net/bpf/test_run.c | 2 +- net/bridge/netfilter/ebtables.c | 2 +- net/can/j1939/socket.c | 2 +- net/can/j1939/transport.c | 21 ++-- net/ceph/ceph_hash.c | 20 +-- net/ceph/crush/mapper.c | 2 +- net/ceph/messenger.c | 4 +- net/ceph/mon_client.c | 2 +- net/ceph/osd_client.c | 6 +- net/core/dev.c | 4 +- net/core/dev_ioctl.c | 2 +- net/core/devlink.c | 2 +- net/core/drop_monitor.c | 2 +- net/core/filter.c | 2 +- net/core/pktgen.c | 2 +- net/core/skmsg.c | 3 +- net/core/sock.c | 2 +- net/dccp/ccids/ccid3.c | 2 +- net/dccp/feat.c | 3 +- net/dccp/input.c | 10 +- net/dccp/options.c | 2 +- net/dccp/output.c | 9 +- net/dccp/proto.c | 8 +- net/decnet/af_decnet.c | 6 +- net/decnet/dn_nsp_in.c | 2 +- net/decnet/dn_table.c | 2 +- net/decnet/sysctl_net_decnet.c | 2 +- net/dsa/slave.c | 2 +- net/ieee802154/6lowpan/reassembly.c | 3 +- net/ieee802154/6lowpan/rx.c | 6 +- net/ipv4/af_inet.c | 4 +- net/ipv4/ah4.c | 2 +- net/ipv4/arp.c | 2 +- net/ipv4/devinet.c | 6 +- net/ipv4/fib_semantics.c | 4 +- net/ipv4/icmp.c | 2 +- net/ipv4/ip_output.c | 2 +- net/ipv4/ipmr.c | 2 +- net/ipv4/netfilter/nf_log_ipv4.c | 2 +- net/ipv4/netfilter/nf_nat_pptp.c | 4 +- net/ipv4/nexthop.c | 2 +- net/ipv4/tcp.c | 2 +- net/ipv4/tcp_input.c | 6 +- net/ipv4/tcp_ipv4.c | 4 +- net/ipv4/udp.c | 2 +- net/ipv6/addrconf.c | 6 +- net/ipv6/ah6.c | 2 +- net/ipv6/exthdrs.c | 2 +- net/ipv6/icmp.c | 2 +- net/ipv6/ip6_fib.c | 8 +- net/ipv6/ip6mr.c | 2 +- net/ipv6/ndisc.c | 2 +- net/ipv6/netfilter/nf_log_ipv6.c | 2 +- net/ipv6/raw.c | 8 +- net/ipv6/route.c | 2 +- net/ipv6/tcp_ipv6.c | 2 +- net/iucv/af_iucv.c | 13 +- net/l2tp/l2tp_netlink.c | 2 +- net/mac80211/cfg.c | 6 +- net/mac80211/chan.c | 2 +- net/mac80211/ht.c | 4 +- net/mac80211/ibss.c | 4 +- net/mac80211/iface.c | 11 +- net/mac80211/key.c | 3 +- net/mac80211/mesh.c | 4 +- net/mac80211/mesh_hwmp.c | 2 +- net/mac80211/mesh_plink.c | 2 +- net/mac80211/mlme.c | 4 +- net/mac80211/offchannel.c | 4 +- net/mac80211/rx.c | 4 +- net/mac80211/tdls.c | 8 +- net/mac80211/tx.c | 12 +- net/mac80211/util.c | 11 +- net/mac80211/wme.c | 2 +- net/mpls/af_mpls.c | 2 +- net/mptcp/protocol.c | 2 +- net/ncsi/ncsi-manage.c | 4 +- net/netfilter/ipset/ip_set_core.c | 2 +- net/netfilter/ipvs/ip_vs_proto_tcp.c | 2 +- net/netfilter/ipvs/ip_vs_proto_udp.c | 2 +- net/netfilter/nf_conntrack_h323_asn1.c | 6 +- net/netfilter/nf_conntrack_proto.c | 2 +- net/netfilter/nf_conntrack_proto_tcp.c | 2 +- net/netfilter/nf_conntrack_standalone.c | 2 +- net/netfilter/nf_nat_core.c | 10 +- net/netfilter/nf_synproxy_core.c | 6 +- net/netfilter/nf_tables_api.c | 8 +- net/netfilter/nf_tables_core.c | 2 +- net/netfilter/nfnetlink_cttimeout.c | 2 +- net/netfilter/nft_cmp.c | 4 +- net/netfilter/nft_ct.c | 4 +- net/netfilter/nft_fib.c | 2 +- net/netfilter/nft_payload.c | 2 +- net/netfilter/utils.c | 8 +- net/netfilter/x_tables.c | 2 +- net/netrom/nr_in.c | 2 +- net/netrom/nr_route.c | 8 +- net/openvswitch/conntrack.c | 4 +- net/openvswitch/flow.c | 2 +- net/packet/af_packet.c | 3 +- net/phonet/pep.c | 10 +- net/rds/send.c | 3 +- net/rose/rose_in.c | 2 +- net/rose/rose_route.c | 4 +- net/rxrpc/af_rxrpc.c | 7 +- net/rxrpc/call_accept.c | 2 +- net/rxrpc/conn_client.c | 2 +- net/rxrpc/input.c | 6 +- net/rxrpc/local_object.c | 2 +- net/rxrpc/recvmsg.c | 2 +- net/rxrpc/sendmsg.c | 6 +- net/sched/act_csum.c | 3 +- net/sched/act_ct.c | 2 +- net/sched/sch_cake.c | 2 +- net/sched/sch_cbq.c | 2 +- net/sched/sch_drr.c | 2 +- net/sched/sch_ets.c | 2 +- net/sched/sch_fq_codel.c | 2 +- net/sched/sch_fq_pie.c | 2 +- net/sched/sch_hfsc.c | 2 +- net/sched/sch_htb.c | 2 +- net/sched/sch_multiq.c | 2 +- net/sched/sch_prio.c | 2 +- net/sched/sch_qfq.c | 2 +- net/sched/sch_sfb.c | 2 +- net/sched/sch_sfq.c | 2 +- net/sctp/ipv6.c | 2 +- net/sctp/outqueue.c | 6 +- net/sctp/sm_make_chunk.c | 2 +- net/sctp/sm_sideeffect.c | 3 +- net/sctp/sm_statefuns.c | 2 +- net/socket.c | 6 +- net/sunrpc/auth_gss/gss_krb5_wrap.c | 2 +- net/sunrpc/clnt.c | 22 ++-- net/sunrpc/xprt.c | 2 +- net/sunrpc/xprtsock.c | 8 +- net/tipc/bearer.c | 2 +- net/tipc/group.c | 2 +- net/tipc/link.c | 2 +- net/tipc/socket.c | 4 +- net/unix/af_unix.c | 2 +- net/wireless/chan.c | 4 +- net/wireless/mlme.c | 2 +- net/wireless/nl80211.c | 20 +-- net/wireless/scan.c | 2 +- net/wireless/sme.c | 4 +- net/wireless/util.c | 4 +- net/wireless/wext-compat.c | 4 +- net/x25/x25_facilities.c | 2 +- net/x25/x25_in.c | 2 +- net/xfrm/xfrm_policy.c | 2 +- samples/bpf/hbm.c | 2 +- security/apparmor/domain.c | 2 +- security/apparmor/lib.c | 4 +- security/integrity/ima/ima_appraise.c | 4 +- security/integrity/ima/ima_policy.c | 8 +- security/integrity/ima/ima_template_lib.c | 2 +- security/keys/process_keys.c | 6 +- security/keys/request_key.c | 12 +- security/selinux/hooks.c | 8 +- security/selinux/ss/mls.c | 3 +- security/smack/smack_lsm.c | 2 +- security/tomoyo/common.c | 18 +-- security/tomoyo/file.c | 2 +- sound/atmel/ac97c.c | 16 +-- sound/core/memalloc.c | 2 +- sound/core/oss/pcm_oss.c | 2 +- sound/core/oss/pcm_plugin.c | 2 +- sound/core/pcm_native.c | 8 +- sound/core/seq/oss/seq_oss_timer.c | 2 +- sound/core/seq/seq_midi_emul.c | 2 +- sound/drivers/opl3/opl3_midi.c | 4 +- sound/isa/galaxy/galaxy.c | 6 +- sound/isa/msnd/msnd_pinnacle_mixer.c | 3 +- sound/isa/opti9xx/miro.c | 12 +- sound/isa/opti9xx/opti92x-ad1848.c | 13 +- sound/isa/sb/sb8_main.c | 10 +- sound/oss/dmasound/dmasound_atari.c | 2 +- sound/oss/dmasound/dmasound_core.c | 4 +- sound/pci/ac97/ac97_codec.c | 4 +- sound/pci/atiixp.c | 6 +- sound/pci/azt3328.c | 2 +- sound/pci/echoaudio/echoaudio.c | 4 +- sound/pci/emu10k1/emupcm.c | 5 +- sound/pci/hda/hda_beep.c | 2 +- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_realtek.c | 4 +- sound/pci/ice1712/delta.c | 2 +- sound/pci/intel8x0.c | 8 +- sound/pci/mixart/mixart.c | 2 +- sound/pci/mixart/mixart_core.c | 2 +- sound/pci/nm256/nm256.c | 6 +- sound/pci/oxygen/oxygen_pcm.c | 2 +- sound/pci/ymfpci/ymfpci_main.c | 2 +- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 4 +- sound/ppc/snd_ps3.c | 4 +- sound/soc/atmel/mchp-i2s-mcc.c | 2 +- sound/soc/codecs/ab8500-codec.c | 2 +- sound/soc/codecs/adau1761.c | 4 +- sound/soc/codecs/adau17x1.c | 4 +- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/ak4613.c | 6 +- sound/soc/codecs/es8328.c | 4 +- sound/soc/codecs/jz4770.c | 2 +- sound/soc/codecs/max9860.c | 2 +- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- sound/soc/codecs/rt274.c | 4 +- sound/soc/codecs/rt5640.c | 4 +- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/sta32x.c | 2 +- sound/soc/codecs/sta350.c | 2 +- sound/soc/codecs/tas2552.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 3 +- sound/soc/codecs/tpa6130a2.c | 2 +- sound/soc/codecs/wm8753.c | 6 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 5 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8993.c | 4 +- sound/soc/codecs/wm8994.c | 4 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/hisilicon/hi6210-i2s.c | 4 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 4 +- sound/soc/intel/skylake/skl-pcm.c | 3 +- sound/soc/meson/axg-tdm-interface.c | 8 +- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 6 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/soc-core.c | 3 +- sound/soc/soc-topology.c | 4 +- sound/soc/sof/intel/hda-dai.c | 6 +- sound/soc/sof/pcm.c | 5 +- sound/soc/sof/topology.c | 2 +- sound/soc/ti/davinci-i2s.c | 2 +- sound/soc/ti/n810.c | 2 +- sound/soc/ti/omap-dmic.c | 4 +- sound/soc/ti/omap-mcpdm.c | 8 +- sound/soc/ti/rx51.c | 2 +- sound/soc/zte/zx-i2s.c | 4 +- sound/soc/zte/zx-spdif.c | 2 +- sound/sparc/dbri.c | 8 +- sound/usb/caiaq/audio.c | 2 +- sound/usb/caiaq/device.c | 2 +- sound/usb/card.c | 3 +- sound/usb/clock.c | 2 +- sound/usb/midi.c | 2 +- sound/usb/pcm.c | 2 +- sound/usb/stream.c | 3 +- sound/xen/xen_snd_front.c | 3 - sound/xen/xen_snd_front_evtchnl.c | 4 - virt/kvm/arm/hyp/vgic-v3-sr.c | 16 +-- 1831 files changed, 4064 insertions(+), 4709 deletions(-) -- 2.24.0

3 6

[PATCH v2 0/6] ASoC: Merge CPU/Codec DAIs
by Kuninori Morimoto 12 Mar '20

12 Mar '20

Hi Mark Current ALSA SoC has CPU/Codec categorized DAIs. But we want to have non-categorized DAIs in the future. This was indicated by Lars-Peter before at ELCE. But, we *can't* do it right now. Because many drivers are considering CPU and Codec separately. Currently, rtd has both CPU/Codec DAIs array. rtd->cpu_dais = [][][][][][][][][] rtd->codec_dais = [][][][][][][][][] This patch merges these, like below. It still can use rtd->cpu_dais, rtd->codec_dais. Of course for_each_rtd_cpu/codec_dais() macro too. So we can use/keep existing code/method same as before. rtd->dais = [][][][][][][][][][][][][][][][][][] ^cpu_dais ^codec_dais |--- num_cpus ---|--- num_codecs --| rtd->cpu_dais = &rtd->dais[0]; rtd->codec_dais = &rtd->dais[dai_link->num_cpus]; There are some code which is doing something *only* for CPU or Codec. This patch-set do nothing to such code. Maybe it can be updated in the future, maybe not (can't). I hope this patch-set can be 1st step for non-categorized DAIs. But the main purpose so far is that replace the duplicate code for borh CPU and Codec. - for_each_rtd_cpu_dais() { - ... - } - for_each_rtd_codec_dais() { - ... - } + for_each_rtd_dais() { + ... + } Kuninori Morimoto (6): ASoC: soc-core: Merge CPU/Codec DAIs ASoC: soc-core: Merge for_each_rtd_cpu/codec_dais() ASoC: soc-dapm: Merge for_each_rtd_cpu/codec_dais() ASoC: soc-pcm: Merge for_each_rtd_cpu/codec_dais() ASoC: soc-core: Merge CPU/Codec for soc_dai_pcm_new() ASoC: soc-pcm: Merge CPU/Codec MSB at soc_pcm_apply_msb() include/sound/soc.h | 7 +- sound/soc/soc-core.c | 66 ++++----- sound/soc/soc-dapm.c | 9 +- sound/soc/soc-pcm.c | 340 +++++++++++-------------------------------- 4 files changed, 116 insertions(+), 306 deletions(-) -- 2.17.1

1 7

[PATCH 0/7] ASoC: Merge CPU/Codec DAIs
by Kuninori Morimoto 12 Mar '20

12 Mar '20

Hi Mark Current ALSA SoC has CPU/Codec categorized DAIs. But it will be no longer good much for modern device. Currently, rtd has both CPU/Codec DAIs array. rtd->cpu_dais = [][][][][][][][][] rtd->codec_dais = [][][][][][][][][] This patch merges these, like below. It still keeps rtd->cpu_dais, rtd->codec_dais rtd->dais = [][][][][][][][][][][][][][][][][][] ^cpu_dais ^codec_dais |--- num_cpus ---|--- num_codecs --| After this merging, we can merge for_each_rtd_cpu/codec_dais(). - for_each_rtd_cpu_dais() { - ... - } - for_each_rtd_codec_dais() { - ... - } + for_each_rtd_dais() { + ... + } Kuninori Morimoto (7): ASoC: soc-core: Merge CPU/Codec DAIs ASoC: soc-core: Merge for_each_rtd_cpu/codec_dais() ASoC: soc-dapm: Merge for_each_rtd_cpu/codec_dais() ASoC: soc-pcm: Merge for_each_rtd_cpu/codec_dais() ASoC: soc-core: Merge CPU/Codec for soc_dai_pcm_new() ASoC: soc-pcm: Merge CPU/Codec MSB at soc_pcm_apply_msb() ASoC: soc-pcm: Merge CPU/Codec at soc_pcm_pointer() include/sound/soc.h | 7 +- sound/soc/soc-core.c | 66 +++----- sound/soc/soc-dapm.c | 9 +- sound/soc/soc-pcm.c | 367 +++++++++++-------------------------------- 4 files changed, 123 insertions(+), 326 deletions(-) -- 2.17.1

4 18

Applied "ASoC: zte: zx-tdm: remove redundant variables dev" to the asoc tree
by Mark Brown 11 Mar '20

11 Mar '20

The patch ASoC: zte: zx-tdm: remove redundant variables dev 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 103ae95513803102d2a2c91458cfac5dfbaad124 Mon Sep 17 00:00:00 2001 From: tangbin <tangbin(a)cmss.chinamobile.com> Date: Wed, 11 Mar 2020 22:46:46 +0800 Subject: [PATCH] ASoC: zte: zx-tdm: remove redundant variables dev In this function, the variable 'dev' is assigned to '&pdev->dev', but in the following code, all the assignments to 'struce device' are used '&pdev->dev' instead of 'dev',except 'zx_tdm->dev'. So,the variable 'dev' in this function is redundant and can be replaced by '&pdev->dev' as elsewhere. Signed-off-by: tangbin <tangbin(a)cmss.chinamobile.com> Link: https://lore.kernel.org/r/20200311144646.11292-1-tangbin@cmss.chinamobile.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/zte/zx-tdm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c index 0e5a05b25a77..4f787185d630 100644 --- a/sound/soc/zte/zx-tdm.c +++ b/sound/soc/zte/zx-tdm.c @@ -371,7 +371,6 @@ static struct snd_soc_dai_driver zx_tdm_dai = { static int zx_tdm_probe(struct platform_device *pdev) { - struct device *dev = &pdev->dev; struct of_phandle_args out_args; unsigned int dma_reg_offset; struct zx_tdm_info *zx_tdm; @@ -384,7 +383,7 @@ static int zx_tdm_probe(struct platform_device *pdev) if (!zx_tdm) return -ENOMEM; - zx_tdm->dev = dev; + zx_tdm->dev = &pdev->dev; zx_tdm->dai_wclk = devm_clk_get(&pdev->dev, "wclk"); if (IS_ERR(zx_tdm->dai_wclk)) { -- 2.20.1

1 0

[alsa-devel] [PATCH v3] ASoC: dt-bindings: stm32: convert spdfirx to json-schema
by Olivier Moysan 11 Mar '20

11 Mar '20

Convert the STM32 SPDIFRX bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- Changes in v2: - Add "additionalProperties: false" - Also change minItems to 2 for dmas property, as both DMAs are required. Changes in v3: - Drop minItems/maxItems for dmas property, remove ref to stm32-dma.txt. --- .../bindings/sound/st,stm32-spdifrx.txt | 56 ------------- .../bindings/sound/st,stm32-spdifrx.yaml | 80 +++++++++++++++++++ 2 files changed, 80 insertions(+), 56 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt deleted file mode 100644 index 33826f2459fa..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt +++ /dev/null @@ -1,56 +0,0 @@ -STMicroelectronics STM32 S/PDIF receiver (SPDIFRX). - -The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with -IEC-60958 and IEC-61937. - -Required properties: - - compatible: should be "st,stm32h7-spdifrx" - - reg: cpu DAI IP base address and size - - clocks: must contain an entry for kclk (used as S/PDIF signal reference) - - clock-names: must contain "kclk" - - interrupts: cpu DAI interrupt line - - dmas: DMA specifiers for audio data DMA and iec control flow DMA - See STM32 DMA bindings, Documentation/devicetree/bindings/dma/stm32-dma.txt - - dma-names: two dmas have to be defined, "rx" and "rx-ctrl" - -Optional properties: - - resets: Reference to a reset controller asserting the SPDIFRX - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -spdifrx: spdifrx@40004000 { - compatible = "st,stm32h7-spdifrx"; - reg = <0x40004000 0x400>; - clocks = <&rcc SPDIFRX_CK>; - clock-names = "kclk"; - interrupts = <97>; - dmas = <&dmamux1 2 93 0x400 0x0>, - <&dmamux1 3 94 0x400 0x0>; - dma-names = "rx", "rx-ctrl"; - pinctrl-0 = <&spdifrx_pins>; - pinctrl-names = "default"; - - spdifrx_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - }; - }; -}; - -spdif_in: spdif-in { - compatible = "linux,spdif-dir"; - - codec_port: port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; - -soundcard { - compatible = "audio-graph-card"; - dais = <&spdifrx_port>; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml new file mode 100644 index 000000000000..b7f7dc452231 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-spdifrx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 S/PDIF receiver (SPDIFRX) + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: | + The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with + IEC-60958 and IEC-61937. + +properties: + compatible: + enum: + - st,stm32h7-spdifrx + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: kclk + + interrupts: + maxItems: 1 + + dmas: + items: + - description: audio data capture DMA + - description: IEC status bits capture DMA + + dma-names: + items: + - const: rx + - const: rx-ctrl + + resets: + maxItems: 1 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + spdifrx: spdifrx@40004000 { + compatible = "st,stm32h7-spdifrx"; + #sound-dai-cells = <0>; + reg = <0x40004000 0x400>; + clocks = <&rcc SPDIF_K>; + clock-names = "kclk"; + interrupts = <GIC_SPI 97 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmamux1 2 93 0x400 0x0>, + <&dmamux1 3 94 0x400 0x0>; + dma-names = "rx", "rx-ctrl"; + pinctrl-0 = <&spdifrx_pins>; + pinctrl-names = "default"; + }; + +... -- 2.17.1

3 6

[PATCH 0/7] ASoC: Intel: Skylake: Fix HDaudio and Dmic
by Cezary Rojewski 11 Mar '20

11 Mar '20

Following is the list of fixes and updates targeting HDaudio +/- Dmic configuration on Intel DSP platforms. - ASoC: Intel: Skylake: Remove superfluous chip initialization ASoC: Intel: Skylake: Select hda configuration permissively First patch addresses race condition issue between i915 and hda controller. This is done by yielding priority to i915 so iDisp codec can enumerate properly: same codec_mask is now observed regardless of driver chosen (snd_hda_intel vs snd_soc_skl). Second patch is a consequence of the first, to prevent driver from incorrectly aborting probe - rather than reorganizing Skylake's boot flow, small changed has been proposed. - ASoC: Intel: Skylake: Shield against no-NHLT configurations Some hardware has no NHLT exposed by BIOS (or an equivalent). Changes have been made to ensure driver is shielded against null-dereferences and such which occur when said table is absent. - ASoC: Intel: skl_hda_dsp: Enable Dmic configuration While DMIC is available on some production stuff, Intel platforms with Skylake driver do not treat it as a valid option if no additional I2S codec in present onboard. Update skl_hda_dsp board to expose Dmic connections too. - ASoC: Intel: Allow for ROM init retry on CNL platforms ASoC: Intel: Skylake: Await purge request ack on CNL Both address rom init timeouts during CNL/ CFL/ CML/ WHL boot up sequences. These provide retry mechanism and ensure purge request is acked before proceding with FW load. bxt-sst.c has had these fixes appended long ago - somehow someone forgotten about CNL family. ***** Note: topology update is also needed to enable HDA +/- Dmic configuration as existing ones do not contain any routes or widgets required to enable it - these care about I2S only. We have prepared corresponding UCM files too. Will be sharing them shortly. This patchset has been prepared internally for topmost linux-stable 5.5 and 4.20 (no 4.19 as skl_hda_dsp did not exist there yet). Apart from our RVPs, we have run tests also on: - KBL Lenovo Carbon X1 - SKL Dell XPS 9350 - WHL Acer Swift 5 Honestly, I'd see HDaudio related patches being backport as low as 4.20 (although some changes had to be adjusted due to base differences between 4.20 and 5.5, can share these too). One could argue HDA + Dmic configuration should be available on 4.19 too - it's an LTS after all. However, that time, some changes could be counted as "feature" rather than fixes. Awaiting your replies and thoughts on that. In consequence, I've appended "Fixes" only for last two patches for now - once decisions are made, can append adequate tags wherever necessary. Cezary Rojewski (6): ASoC: Intel: Skylake: Remove superfluous chip initialization ASoC: Intel: Skylake: Select hda configuration permissively ASoC: Intel: Skylake: Enable codec wakeup during chip init ASoC: Intel: Skylake: Shield against no-NHLT configurations ASoC: Intel: Allow for ROM init retry on CNL platforms ASoC: Intel: Skylake: Await purge request ack on CNL Mateusz Gorski (1): ASoC: Intel: skl_hda_dsp: Enable Dmic configuration sound/soc/intel/boards/skl_hda_dsp_generic.c | 3 ++ sound/soc/intel/skylake/bxt-sst.c | 3 -- sound/soc/intel/skylake/cnl-sst.c | 35 ++++++++++++++++---- sound/soc/intel/skylake/skl-nhlt.c | 3 +- sound/soc/intel/skylake/skl-sst-dsp.h | 2 ++ sound/soc/intel/skylake/skl.c | 29 ++++++++-------- 6 files changed, 48 insertions(+), 27 deletions(-) -- 2.17.1

4 41

Applied "ASoC: Convert cirrus, cs42l51 to json-schema" to the asoc tree
by Mark Brown 11 Mar '20

11 Mar '20

The patch ASoC: Convert cirrus,cs42l51 to json-schema has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.7 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 97249a89c17e8f1288fed1ebc617ea2e9e88d501 Mon Sep 17 00:00:00 2001 From: Benjamin Gaignard <benjamin.gaignard(a)st.com> Date: Fri, 28 Feb 2020 16:27:06 +0100 Subject: [PATCH] ASoC: Convert cirrus,cs42l51 to json-schema Convert cirrus,cs42l51 to yaml format. Signed-off-by: Benjamin Gaignard <benjamin.gaignard(a)st.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Link: https://lore.kernel.org/r/20200228152706.29749-1-benjamin.gaignard@st.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/cirrus,cs42l51.yaml | 69 +++++++++++++++++++ .../devicetree/bindings/sound/cs42l51.txt | 33 --------- 2 files changed, 69 insertions(+), 33 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml delete mode 100644 Documentation/devicetree/bindings/sound/cs42l51.txt diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml new file mode 100644 index 000000000000..efce847a3408 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs42l51.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: CS42L51 audio codec DT bindings + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +properties: + compatible: + const: cirrus,cs42l51 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: MCLK + + reset-gpios: + maxItems: 1 + + VL-supply: + description: phandle to voltage regulator of digital interface section + + VD-supply: + description: phandle to voltage regulator of digital internal section + + VA-supply: + description: phandle to voltage regulator of analog internal section + + VAHP-supply: + description: phandle to voltage regulator of headphone + +required: + - compatible + - reg + - "#sound-dai-cells" + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c@0 { + #address-cells = <1>; + #size-cells = <0>; + + cs42l51@4a { + compatible = "cirrus,cs42l51"; + reg = <0x4a>; + #sound-dai-cells = <0>; + clocks = <&mclk_prov>; + clock-names = "MCLK"; + VL-supply = <&reg_audio>; + VD-supply = <&reg_audio>; + VA-supply = <&reg_audio>; + VAHP-supply = <&reg_audio>; + reset-gpios = <&gpiog 9 GPIO_ACTIVE_LOW>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/cs42l51.txt b/Documentation/devicetree/bindings/sound/cs42l51.txt deleted file mode 100644 index acbd68ddd2cb..000000000000 --- a/Documentation/devicetree/bindings/sound/cs42l51.txt +++ /dev/null @@ -1,33 +0,0 @@ -CS42L51 audio CODEC - -Required properties: - - - compatible : "cirrus,cs42l51" - - - reg : the I2C address of the device for I2C. - -Optional properties: - - VL-supply, VD-supply, VA-supply, VAHP-supply: power supplies for the device, - as covered in Documentation/devicetree/bindings/regulator/regulator.txt. - - - reset-gpios : GPIO specification for the reset pin. If specified, it will be - deasserted before starting the communication with the codec. - - - clocks : a list of phandles + clock-specifiers, one for each entry in - clock-names - - - clock-names : must contain "MCLK" - -Example: - -cs42l51: cs42l51@4a { - compatible = "cirrus,cs42l51"; - reg = <0x4a>; - clocks = <&mclk_prov>; - clock-names = "MCLK"; - VL-supply = <&reg_audio>; - VD-supply = <&reg_audio>; - VA-supply = <&reg_audio>; - VAHP-supply = <&reg_audio>; - reset-gpios = <&gpiog 9 GPIO_ACTIVE_LOW>; -}; -- 2.20.1

1 0

[RFC PATCH] soundwire: bus: Add flag to mark DPN_BlockCtrl1 as readonly
by Srinivas Kandagatla 11 Mar '20

11 Mar '20

According to SoundWire Specification Version 1.2. "A Data Port number X (in the range 0-14) which supports only one value of WordLength may implement the WordLength field in the DPX_BlockCtrl1 Register as Read-Only, returning the fixed value of WordLength in response to reads." As WSA881x interfaces in PDM mode making the only field "WordLength" in DPX_BlockCtrl1" fixed and read-only. Behaviour of writing to this register on WSA881x soundwire slave with Qualcomm Soundwire Controller is throwing up an error. Not sure how other controllers deal with writing to readonly registers, but this patch provides a way to avoid writes to DPN_BlockCtrl1 register by providing a ro_blockctrl1_reg flag in struct sdw_port_runtime. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- I will send patch for WSA881x to include this change once this patch is accepted. drivers/soundwire/bus.h | 2 ++ drivers/soundwire/stream.c | 17 ++++++++++------- 2 files changed, 12 insertions(+), 7 deletions(-) diff --git a/drivers/soundwire/bus.h b/drivers/soundwire/bus.h index 204204a26db8..791e8d14093e 100644 --- a/drivers/soundwire/bus.h +++ b/drivers/soundwire/bus.h @@ -79,6 +79,7 @@ int sdw_find_col_index(int col); * @num: Port number. For audio streams, valid port number ranges from * [1,14] * @ch_mask: Channel mask + * @ro_blockctrl1_reg: Read Only flag for DPN_BlockCtrl1 register * @transport_params: Transport parameters * @port_params: Port parameters * @port_node: List node for Master or Slave port_list @@ -89,6 +90,7 @@ int sdw_find_col_index(int col); struct sdw_port_runtime { int num; int ch_mask; + bool ro_blockctrl1_reg; struct sdw_transport_params transport_params; struct sdw_port_params port_params; struct list_head port_node; diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 00348d1fc606..4491643aeb4a 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -167,13 +167,15 @@ static int sdw_program_slave_port_params(struct sdw_bus *bus, return ret; } - /* Program DPN_BlockCtrl1 register */ - ret = sdw_write(s_rt->slave, addr2, (p_params->bps - 1)); - if (ret < 0) { - dev_err(&s_rt->slave->dev, - "DPN_BlockCtrl1 register write failed for port %d\n", - t_params->port_num); - return ret; + if (!p_rt->ro_blockctrl1_reg) { + /* Program DPN_BlockCtrl1 register */ + ret = sdw_write(s_rt->slave, addr2, (p_params->bps - 1)); + if (ret < 0) { + dev_err(&s_rt->slave->dev, + "DPN_BlockCtrl1 register write failed for port %d\n", + t_params->port_num); + return ret; + } } /* Program DPN_SampleCtrl1 register */ @@ -1195,6 +1197,7 @@ static struct sdw_port_runtime p_rt->ch_mask = port_config[port_index].ch_mask; p_rt->num = port_config[port_index].num; + p_rt->ro_blockctrl1_reg = port_config[port_index].ro_blockctrl1_reg; return p_rt; } -- 2.21.0

3 5

[PATCH] ALSA: hda/realtek: Fix pop noise on ALC225
by Kai-Heng Feng 11 Mar '20

11 Mar '20

Commit 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") makes the ALC225 have pop noise on S3 resume and cold boot. So partially revert this commit for ALC225 to fix the regression. Fixes: 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") BugLink: https://bugs.launchpad.net/bugs/1866357 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0ac06ff1a17c..7b83b020ac3c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8051,6 +8051,8 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; break; case 0x10ec0225: + codec->power_save_node = 1; + /* fall through */ case 0x10ec0295: case 0x10ec0299: spec->codec_variant = ALC269_TYPE_ALC225; -- 2.17.1

2 2