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[PATCH v2 0/9] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 12 Oct '21

12 Oct '21

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. This patch set depends on: -- https://patchwork.kernel.org/project/alsa-devel/list/?series=549577 -- https://patchwork.kernel.org/project/alsa-devel/list/?series=548765 -- https://patchwork.kernel.org/project/alsa-devel/list/?series=543829 Chnages Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (9): ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: qcom: Add support for codec dma driver ASoC: dt-bindings: Add SC7280 sound card bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio ASoc: qcom: lpass: Add suspend and resume for sc7280 platform ASoC: qcom: SC7280: Update config for building codec dma drivers .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 69 ++- sound/soc/qcom/Kconfig | 13 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 209 +++++++ sound/soc/qcom/lpass-cpu.c | 274 +++++++++- sound/soc/qcom/lpass-lpaif-reg.h | 103 +++- sound/soc/qcom/lpass-platform.c | 497 +++++++++++++++-- sound/soc/qcom/lpass-sc7280.c | 602 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 156 ++++++ 9 files changed, 1870 insertions(+), 57 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH] ALSA: hda/realtek: Fix the mic type detection issue for ASUS G551JW
by Hui Wang 12 Oct '21

12 Oct '21

We need to define the codec pin 0x1b to be the mic, but somehow the mic doesn't support hot plugging detection, and Windows also has this issue, so we set it to phantom headset-mic. Also the determine_headset_type() often returns the omtp type by a mistake when we plug a ctia headset, this makes the mic can't record sound at all. Because most of the headset are ctia type nowadays and some machines have the fixed ctia type audio jack, it is possible this machine has the fixed ctia jack too. Here we set this mic jack to fixed ctia type, this could avoid the mic type detection mistake and make the ctia headset work stable. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214537 Reported-and-tested-by: msd <msd.mmq(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bca5830ff706..22d27b12c4e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10197,6 +10197,9 @@ enum { ALC671_FIXUP_HP_HEADSET_MIC2, ALC662_FIXUP_ACER_X2660G_HEADSET_MODE, ALC662_FIXUP_ACER_NITRO_HEADSET_MODE, + ALC668_FIXUP_ASUS_NO_HEADSET_MIC, + ALC668_FIXUP_HEADSET_MIC, + ALC668_FIXUP_MIC_DET_COEF, }; static const struct hda_fixup alc662_fixups[] = { @@ -10580,6 +10583,29 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC662_FIXUP_USI_FUNC }, + [ALC668_FIXUP_ASUS_NO_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x04a1112c }, + { } + }, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_MIC + }, + [ALC668_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_headset_mic, + .chained = true, + .chain_id = ALC668_FIXUP_MIC_DET_COEF + }, + [ALC668_FIXUP_MIC_DET_COEF] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x15 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0d60 }, + {} + }, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -10615,6 +10641,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51), + SND_PCI_QUIRK(0x1043, 0x185d, "ASUS G551JW", ALC668_FIXUP_ASUS_NO_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71SL", ALC662_FIXUP_ASUS_MODE8), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), -- 2.25.1

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[PATCH] ASoC: cs42l42: Ensure 0dB full scale volume is used for headsets
by Richard Fitzgerald 12 Oct '21

12 Oct '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> Ensure the default 0dB playback path is always used. The code that set FULL_SCALE_VOL based on LOAD_DET_RCSTAT was spurious, and resulted in a -6dB attenuation being accidentally inserted into the playback path. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d484ec09eb2e..8de23e4732b3 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -922,7 +922,6 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); unsigned int regval; - u8 fullScaleVol; int ret; if (mute) { @@ -993,20 +992,11 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) cs42l42->stream_use |= 1 << stream; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* Read the headphone load */ - regval = snd_soc_component_read(component, CS42L42_LOAD_DET_RCSTAT); - if (((regval & CS42L42_RLA_STAT_MASK) >> CS42L42_RLA_STAT_SHIFT) == - CS42L42_RLA_STAT_15_OHM) { - fullScaleVol = CS42L42_HP_FULL_SCALE_VOL_MASK; - } else { - fullScaleVol = 0; - } - - /* Un-mute the headphone, set the full scale volume flag */ + /* Un-mute the headphone */ snd_soc_component_update_bits(component, CS42L42_HP_CTL, CS42L42_HP_ANA_AMUTE_MASK | - CS42L42_HP_ANA_BMUTE_MASK | - CS42L42_HP_FULL_SCALE_VOL_MASK, fullScaleVol); + CS42L42_HP_ANA_BMUTE_MASK, + 0); } } -- 2.11.0

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[PATCH 0/8] arm: ep93xx: CCF conversion
by Nikita Shubin 12 Oct '21

12 Oct '21

This series series of patches converts ep93xx to Common Clock Framework. It consists of preparation patches to use clk_prepare_enable where it is needed, instead of clk_enable used in ep93xx drivers prior to CCF and a patch converting mach-ep93xx/clock.c to CCF. Link: https://lore.kernel.org/patchwork/cover/1445563/ Link: https://lore.kernel.org/patchwork/patch/1435884/ Alexander Sverdlin (7): iio: ep93xx: Prepare clock before using it spi: spi-ep93xx: Prepare clock before using it Input: ep93xx_keypad: Prepare clock before using it video: ep93xx: Prepare clock before using it dmaengine: ep93xx: Prepare clock before using it ASoC: cirrus: i2s: Prepare clock before using it pwm: ep93xx: Prepare clock before using it Nikita Shubin (1): ep93xx: clock: convert in-place to COMMON_CLK arch/arm/Kconfig | 2 +- arch/arm/mach-ep93xx/clock.c | 975 ++++++++++++++----------- arch/arm/mach-ep93xx/core.c | 2 +- arch/arm/mach-ep93xx/soc.h | 42 +- drivers/dma/ep93xx_dma.c | 6 +- drivers/iio/adc/ep93xx_adc.c | 6 +- drivers/input/keyboard/ep93xx_keypad.c | 4 +- drivers/pwm/pwm-ep93xx.c | 12 +- drivers/spi/spi-ep93xx.c | 4 +- drivers/video/fbdev/ep93xx-fb.c | 4 +- sound/soc/cirrus/ep93xx-i2s.c | 12 +- 11 files changed, 605 insertions(+), 464 deletions(-) base-commit: 64376a981a0e2e57c46efa63197c2ebb7dab35df -- 2.26.2

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[PATCH v3 0/5] Update Lpass digital codec macro drivers
by Srinivasa Rao Mandadapu 12 Oct '21

12 Oct '21

This patch set is to add support for lpass sc7280 based targets. Upadate compatible name and change of bulk clock voting to optional clock voting in digital codecs va, rx, tx macro drivers. Changes Since V2: -- Add Tx macro deafults for lpass sc7280 Changes Since V1: -- Removed individual clock voting and used bulk clock optional. -- Removed volatile changes and fixed default values. -- Typo errors. Srinivasa Rao Mandadapu (5): ASoC: qcom: Add compatible names in va,wsa,rx,tx codec drivers for sc7280 ASoC: qcom: dt-bindings: Add compatible names for lpass sc7280 digital codecs ASoC: codecs: tx-macro: Enable tx top soundwire mic clock ASoC: codecs: tx-macro: Update tx default values ASoC: codecs: Change bulk clock voting to optional voting in digital codecs .../bindings/sound/qcom,lpass-rx-macro.yaml | 4 +++- .../bindings/sound/qcom,lpass-tx-macro.yaml | 4 +++- .../bindings/sound/qcom,lpass-va-macro.yaml | 4 +++- .../bindings/sound/qcom,lpass-wsa-macro.yaml | 4 +++- sound/soc/codecs/lpass-rx-macro.c | 3 ++- sound/soc/codecs/lpass-tx-macro.c | 25 +++++++++++++++++++--- sound/soc/codecs/lpass-va-macro.c | 3 ++- sound/soc/codecs/lpass-wsa-macro.c | 1 + 8 files changed, 39 insertions(+), 9 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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Add ALSA UCM support for Pinebook Pro's audio
by Jian-Hong Pan 12 Oct '21

12 Oct '21

Hi, I prepared the pull request "ucm2: Add UCM support for rockchip_es8316 on Pinebook Pro" [1] to enable Pinebook Pro's audio generically. Also, the corresponding validator pull request "ucm-validator: Upload rockchip_es8316's dummy json and alsa-info" [2]. The original information comes from Manjaro ARM's pinebookpro-post-install's asound.state [3] and pinebookpro-audio's audio_jack_plugged_in.sh [4]. To make it available on most distros, I transfer the asound.state and the shell script into ALSA UCM conf. It will be appreciated if someone can review the pull requests and give some feedback. [1]: https://github.com/alsa-project/alsa-ucm-conf/pull/112 [2]: https://github.com/alsa-project/alsa-tests/pull/22 [3]: https://gitlab.manjaro.org/manjaro-arm/packages/community/pinebookpro-post-… [4]: https://gitlab.manjaro.org/manjaro-arm/packages/community/pinebookpro-audio… Jian-Hong Pan

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[PATCH v2 0/8] ASoC: Add common modules support for ACP hw block
by Ajit Kumar Pandey 12 Oct '21

12 Oct '21

Changes since v1: - Fixes Kconfig indent as per kernel coding style guidelines. - Updated _GPL suffix for all exported symbol. - As suggested added separate machine ops for each codecs in machine driver - Add sound card support for newer codecs Ajit Kumar Pandey (8): ASoC: amd: Add common framework to support I2S on ACP SOC ASoC: amd: acp: Add I2S support on Renoir platform ASoC: amd: acp: Add callback for machine driver on ACP ASoC: amd: acp: Add generic machine driver support for ACP cards ASoC: amd: acp: Add legacy sound card support for Chrome audio ASoC: amd: acp: Add SOF audio support on Chrome board ASoC: amd: acp: Add support for Maxim amplifier codec ASoC: amd: acp: Add support for RT5682-VS codec sound/soc/amd/Kconfig | 2 + sound/soc/amd/Makefile | 1 + sound/soc/amd/acp/Kconfig | 50 +++ sound/soc/amd/acp/Makefile | 26 ++ sound/soc/amd/acp/acp-i2s.c | 340 +++++++++++++++ sound/soc/amd/acp/acp-legacy-mach.c | 104 +++++ sound/soc/amd/acp/acp-mach-common.c | 596 +++++++++++++++++++++++++++ sound/soc/amd/acp/acp-mach.h | 57 +++ sound/soc/amd/acp/acp-platform.c | 315 ++++++++++++++ sound/soc/amd/acp/acp-renoir.c | 144 +++++++ sound/soc/amd/acp/acp-sof-mach.c | 131 ++++++ sound/soc/amd/acp/amd.h | 146 +++++++ sound/soc/amd/acp/chip_offset_byte.h | 76 ++++ 13 files changed, 1988 insertions(+) create mode 100644 sound/soc/amd/acp/Kconfig create mode 100644 sound/soc/amd/acp/Makefile create mode 100644 sound/soc/amd/acp/acp-i2s.c create mode 100644 sound/soc/amd/acp/acp-legacy-mach.c create mode 100644 sound/soc/amd/acp/acp-mach-common.c create mode 100644 sound/soc/amd/acp/acp-mach.h create mode 100644 sound/soc/amd/acp/acp-platform.c create mode 100644 sound/soc/amd/acp/acp-renoir.c create mode 100644 sound/soc/amd/acp/acp-sof-mach.c create mode 100644 sound/soc/amd/acp/amd.h create mode 100644 sound/soc/amd/acp/chip_offset_byte.h -- 2.25.1

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[PATCH] ASoC: simple-card: Fill in driver name
by Guido Günther 12 Oct '21

12 Oct '21

alsa-ucm groups by driver name so fill that in as well. Otherwise the presented information is redundant and doesn't reflect the used driver. We can't just use 'asoc-simple-card' since the driver name is restricted to 15 characters. Before: # cat /proc/asound/cards 0 [Devkit ]: Librem_5_Devkit - Librem 5 Devkit Librem 5 Devkit After: 0 [Devkit ]: simple-card - Librem 5 Devkit Librem 5 Devkit Signed-off-by: Guido Günther <agx(a)sigxcpu.org> --- This came out of a discussion about adding alsa-ucm profiles for the Librem 5 Devkit at https://github.com/alsa-project/alsa-ucm-conf/pull/102 sound/soc/generic/simple-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 0015f534d42d..a3a7990b5cb6 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -621,6 +621,7 @@ static int asoc_simple_probe(struct platform_device *pdev) card->owner = THIS_MODULE; card->dev = dev; card->probe = simple_soc_probe; + card->driver_name = "simple-card"; li = devm_kzalloc(dev, sizeof(*li), GFP_KERNEL); if (!li) -- 2.30.2

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[PATCH v3 0/2] ASoC: rt9120: Add Richtek RT9120 supprot
by cy_huang 12 Oct '21

12 Oct '21

From: ChiYuan Huang <cy_huang(a)richtek.com> This patch series Add the Richtek RT9120 support. In v3: - Add dvdd regulator binding to check the dvdd voltage domain. - Refine sdo_select_text. - Use switch case in 'internal_power_event' function. - Remove the volume and mute initially write in component probe. - Remove the mute API. It's no need by HW design. In v2: - Add missing #sound-dai-cells property. ChiYuan Huang (2): ASoC: dt-bindings: rt9120: Add initial bindings ASoC: rt9120: Add rt9210 audio amplifier support .../devicetree/bindings/sound/richtek,rt9120.yaml | 59 +++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt9120.c | 489 +++++++++++++++++++++ 4 files changed, 560 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/richtek,rt9120.yaml create mode 100644 sound/soc/codecs/rt9120.c -- 2.7.4

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[PATCH] ASoC: soc-core: fix null-ptr-deref in snd_soc_del_component_unlocked()
by Yang Yingliang 11 Oct '21

11 Oct '21

'component' is allocated in snd_soc_register_component(), but component->list is not initalized, this may cause snd_soc_del_component_unlocked() deref null ptr in the error handing case. KASAN: null-ptr-deref in range [0x0000000000000000-0x0000000000000007] RIP: 0010:__list_del_entry_valid+0x81/0xf0 Call Trace: snd_soc_del_component_unlocked+0x69/0x1b0 [snd_soc_core] snd_soc_add_component.cold+0x54/0x6c [snd_soc_core] snd_soc_register_component+0x70/0x90 [snd_soc_core] devm_snd_soc_register_component+0x5e/0xd0 [snd_soc_core] tas2552_probe+0x265/0x320 [snd_soc_tas2552] ? tas2552_component_probe+0x1e0/0x1e0 [snd_soc_tas2552] i2c_device_probe+0xa31/0xbe0 Fix by adding INIT_LIST_HEAD() to snd_soc_component_initialize(). Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c830e96afba2..80ca260595fd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2599,6 +2599,7 @@ int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->dai_list); INIT_LIST_HEAD(&component->dobj_list); INIT_LIST_HEAD(&component->card_list); + INIT_LIST_HEAD(&component->list); mutex_init(&component->io_mutex); component->name = fmt_single_name(dev, &component->id); -- 2.25.1

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