June 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] soc: fix typos in comment
by Ren Yu 13 Jun '22

13 Jun '22

change 'slection' to 'selection' Signed-off-by: Ren Yu <renyu(a)nfschina.com> --- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/rt5631.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index bd0078e4499b..2eb43e93778e 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1201,7 +1201,7 @@ static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, case 12288000: case 26000000: case 19200000: - /* pll clk slection */ + /* pll clk selection */ pll_clk = 0; break; case 48000: diff --git a/sound/soc/codecs/rt5631.h b/sound/soc/codecs/rt5631.h index 8a6b99a48c7c..0caca2e1d31b 100644 --- a/sound/soc/codecs/rt5631.h +++ b/sound/soc/codecs/rt5631.h @@ -373,7 +373,7 @@ */ #define RT5631_SDP_DAC_DATA_L_R_SWAP (0x1 << 4) -/* Data Length Slection */ +/* Data Length Selection */ #define RT5631_SDP_I2S_DL_MASK (0x3 << 2) #define RT5631_SDP_I2S_DL_16 (0x0 << 2) #define RT5631_SDP_I2S_DL_20 (0x1 << 2) -- 2.11.0

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[PATCH v7 0/8] ASoC: mediatek: Add support for MT8186 SoC
by Jiaxin Yu 13 Jun '22

13 Jun '22

This series of patches adds support for Mediatek AFE of MT8186 Soc. Patches are based on broonie tree "for-next" branch. Changes since v6: - The Makefile and Kconfig updates are moved into the driver patches so that can keep independence of each patch. Changes since v5: - fix build error about function snd_soc_card_jack_new that modified by: Link: https://lore.kernel.org/r/20220408041114.6024-1-akihiko.odaki@gmail.com - some have been accepted, details are in the link below: Link: https://lore.kernel.org/all/165459931885.399031.2621579592368573898.b4-ty@k… Changes since v4: - [v5 07/20] - remove unsusd controls - [v5 09/20] - correct indent error - [v5 10/20] - [v5 13/20] - [v5 14/20] - fix the return value if the value is different from the previous value in mixer controls - [v5 17/20] - [v5 19/20] - correct the compatible name with '_' instead of '-' - [v5 18/20] - [v5 20/20] - correct the yaml after 'pip3 install dtschema --upgrade' Changes since v3: - [v4 09/18] - remove DEBUG_COEFF debugging code - [v4 10/18] - simplify the logic of the code - [v4 13/18] - split out the MT6366 bits into mt8186-mt6366-commom.c - fix build error of "error: 'struct dev_pm_info' has no member named 'runtime_error'" - fix bug of adda dai driver - add route for pcm interface channel 2. - [v4 15/18] - [v4 17/18] - commonize the configuration of the codec - move MT6366 common bits into mt8186-mt6366-common.c Changes since v2: - add a new compatible string "mediatek,mt6366-sound" - modify the log level for simplicity - use dev_err_probe(...) instead of dev_err(...) in dev probe() - optimized the logic of some code - use BIT() and GENMASK() macros to descript the registers Thanks for AngeloGioacchino's careful reviews. Changes since v1: [v2 01/17] - add a new ID to the existing mt6358 codec driver [v2 03/17] - modify log level in DAPM events - use standard numeric control with name ending in Switch - return 1 when the value changed in mixer control's .get callback [v2 05/17] - ending in Switch to the standard on/off controls - change to "HW Gain 1 Volume" and "HW Gain 2 Volume" [v2 09/17] - return an error in the default case rather than just picking one of the behaviours when do .set_fmt - use the new defines that are _PROVIDER_MASK, _DAIFMT_CBP_CFP and _DAIFMT_CBC_CFC [v2 10/17] [v2 11/17] - the clock and gpio are aplit out into separate patches The source file's GPL comment use c++ style, and the header fils's GPL comment use c style. We have added "Switch" after the names of all the controls that just are simple on/off. Jiaxin Yu (8): dt-bindings: mediatek: mt6358: add new compatible for using mt6366 ASoC: mediatek: mt8186: add platform driver ASoC: mediatek: mt8186: add mt8186-mt6366 common driver dt-bindings: mediatek: mt8186: add audio afe document ASoC: mediatek: mt8186: add machine driver with mt6366, da7219 and max98357 dt-bindings: mediatek: mt8186: add mt8186-mt6366-da7219-max98357 document ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682s dt-bindings: mediatek: mt8186: add mt8186-mt6366-rt1019-rt5682s document .../devicetree/bindings/sound/mt6358.txt | 4 +- .../bindings/sound/mt8186-afe-pcm.yaml | 175 + .../sound/mt8186-mt6366-da7219-max98357.yaml | 75 + .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 75 + sound/soc/mediatek/Kconfig | 44 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8186/Makefile | 22 + sound/soc/mediatek/mt8186/mt8186-afe-common.h | 235 ++ .../soc/mediatek/mt8186/mt8186-afe-control.c | 261 ++ sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3009 +++++++++++++++++ .../mediatek/mt8186/mt8186-mt6366-common.c | 59 + .../mediatek/mt8186/mt8186-mt6366-common.h | 17 + .../mt8186/mt8186-mt6366-da7219-max98357.c | 1002 ++++++ .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 978 ++++++ 14 files changed, 5956 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml create mode 100644 sound/soc/mediatek/mt8186/Makefile create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c -- 2.18.0

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[PATCH] soc: There are several spelling mistakes. Fix them
by Ren Yu 13 Jun '22

13 Jun '22

Signed-off-by: Ren Yu <renyu(a)nfschina.com> --- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/rt5631.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index bd0078e4499b..2eb43e93778e 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1201,7 +1201,7 @@ static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, case 12288000: case 26000000: case 19200000: - /* pll clk slection */ + /* pll clk selection */ pll_clk = 0; break; case 48000: diff --git a/sound/soc/codecs/rt5631.h b/sound/soc/codecs/rt5631.h index 8a6b99a48c7c..0caca2e1d31b 100644 --- a/sound/soc/codecs/rt5631.h +++ b/sound/soc/codecs/rt5631.h @@ -373,7 +373,7 @@ */ #define RT5631_SDP_DAC_DATA_L_R_SWAP (0x1 << 4) -/* Data Length Slection */ +/* Data Length Selection */ #define RT5631_SDP_I2S_DL_MASK (0x3 << 2) #define RT5631_SDP_I2S_DL_16 (0x0 << 2) #define RT5631_SDP_I2S_DL_20 (0x1 << 2) -- 2.11.0

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[PATCH] ALSA: usb-audio: US16x08: Move overflow check before array access
by Daniil Dementev 13 Jun '22

13 Jun '22

Buffer overflow could occur in the loop "while", due to accessing an array element before checking the index. Found by Linux Verification Center (linuxtesting.org) with SVACE. Signed-off-by: Daniil Dementev <d.dementev(a)ispras.ru> Reviewed-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> --- sound/usb/mixer_us16x08.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index b7b6f3834ed5..6eb7d93b358d 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -637,10 +637,10 @@ static int snd_get_meter_comp_index(struct snd_us16x08_meter_store *store) } } else { /* skip channels with no compressor active */ - while (!store->comp_store->val[ + while (store->comp_index <= SND_US16X08_MAX_CHANNELS + && !store->comp_store->val[ COMP_STORE_IDX(SND_US16X08_ID_COMP_SWITCH)] - [store->comp_index - 1] - && store->comp_index <= SND_US16X08_MAX_CHANNELS) { + [store->comp_index - 1]) { store->comp_index++; } ret = store->comp_index++; -- 2.35.1.windows.2

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Asus Z690-E: No Profiles Available in alsa-lib 1.2.7-2
by GitHub issues - opened 11 Jun '22

11 Jun '22

alsa-project/alsa-ucm-conf issue #168 was opened from BrothersRM: Upgrading alsa-lib from 1.2.6.1-1 to 1.2.7-2 seems to have broken support for my motherboard's sound card. Downgrading the package did resolve the issue. Device is listed in `lsusb -v` as : `Bus 001 Device 002: ID 0b05:1996 ASUSTek Computer, Inc. USB Audio` My hardware configuration uses speakers attached to the front, surround and subwoofer ports on my motherboard. Some interesting things I noticed while it was broken: - Plugging headphones into the front-panel headphone port allowed me to turn on the rear speakers, though they didn't seem to use the correct profile. Max volume was far lower - Though the device was listed as unavailable, I could manually enable the rear speakers using pavucontrol when front headphones weren't plugged in - Trying to run speaker-test on the device failed with 'no valid configuration' Please let me know if I can provide more information / do any testing. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/168 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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1.2.7 aplay doesn't compile
by GitHub issues - opened 11 Jun '22

11 Jun '22

alsa-project/alsa-utils issue #153 was opened from LoneFox78: I'm getting this error: ``` Making all in aplay make[1]: Entering directory '/mnt/pkg/pkgmk_work/alsa-utils/src/alsa-utils-1.2.7/aplay' gcc -DHAVE_CONFIG_H -I. -I../include -I../include -s -Wl,-s -Wall -O2 -march=native -ftree-vectorize -pipe -MT aplay.o -MD -MP -MF .deps/aplay.Tpo -c -o aplay.o aplay.c aplay.c:44:10: fatal error: alsa/use-case.h: No such file or directory 44 | #include <alsa/use-case.h> | ^~~~~~~~~~~~~~~~~ compilation terminated. make[1]: *** [Makefile:495: aplay.o] Error 1 make[1]: Leaving directory '/mnt/pkg/pkgmk_work/alsa-utils/src/alsa-utils-1.2.7/aplay' make: *** [Makefile:461: all-recursive] Error 1 ``` The configure script already looks for that header, but I guess someone forgot to add the corresponding <code>#ifdef</code>s in this file. Issue URL : https://github.com/alsa-project/alsa-utils/issues/153 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] ALSA: pcm: add 24KHz sample rate definition
by Patrick Lai 11 Jun '22

11 Jun '22

Add 24KHz sample rate definition as some hardware can support 24KHz sample rate PCM data Signed-off-by: Patrick Lai <quic_plai(a)quicinc.com> --- include/sound/pcm.h | 24 +++++++++++++----------- sound/core/pcm_native.c | 4 ++-- 2 files changed, 15 insertions(+), 13 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 6b99310b5b88..a761a4459091 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -111,23 +111,25 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_11025 (1<<2) /* 11025Hz */ #define SNDRV_PCM_RATE_16000 (1<<3) /* 16000Hz */ #define SNDRV_PCM_RATE_22050 (1<<4) /* 22050Hz */ -#define SNDRV_PCM_RATE_32000 (1<<5) /* 32000Hz */ -#define SNDRV_PCM_RATE_44100 (1<<6) /* 44100Hz */ -#define SNDRV_PCM_RATE_48000 (1<<7) /* 48000Hz */ -#define SNDRV_PCM_RATE_64000 (1<<8) /* 64000Hz */ -#define SNDRV_PCM_RATE_88200 (1<<9) /* 88200Hz */ -#define SNDRV_PCM_RATE_96000 (1<<10) /* 96000Hz */ -#define SNDRV_PCM_RATE_176400 (1<<11) /* 176400Hz */ -#define SNDRV_PCM_RATE_192000 (1<<12) /* 192000Hz */ -#define SNDRV_PCM_RATE_352800 (1<<13) /* 352800Hz */ -#define SNDRV_PCM_RATE_384000 (1<<14) /* 384000Hz */ +#define SNDDV_PCM_RATE_24000 (1<<5) /* 24000Hz */ +#define SNDRV_PCM_RATE_32000 (1<<6) /* 32000Hz */ +#define SNDRV_PCM_RATE_44100 (1<<7) /* 44100Hz */ +#define SNDRV_PCM_RATE_48000 (1<<8) /* 48000Hz */ +#define SNDRV_PCM_RATE_64000 (1<<9) /* 64000Hz */ +#define SNDRV_PCM_RATE_88200 (1<<10) /* 88200Hz */ +#define SNDRV_PCM_RATE_96000 (1<<11) /* 96000Hz */ +#define SNDRV_PCM_RATE_176400 (1<<12) /* 176400Hz */ +#define SNDRV_PCM_RATE_192000 (1<<13) /* 192000Hz */ +#define SNDRV_PCM_RATE_352800 (1<<14) /* 352800Hz */ +#define SNDRV_PCM_RATE_384000 (1<<15) /* 384000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1<<30) /* continuous range */ #define SNDRV_PCM_RATE_KNOT (1<<31) /* supports more non-continuos rates */ #define SNDRV_PCM_RATE_8000_44100 (SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_11025|\ SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_22050|\ - SNDRV_PCM_RATE_32000|SNDRV_PCM_RATE_44100) + SNDRV_PCM_RATE_24000|SNDRV_PCM_RATE_32000|\ + SNDRV_PCM_RATE_44100) #define SNDRV_PCM_RATE_8000_48000 (SNDRV_PCM_RATE_8000_44100|SNDRV_PCM_RATE_48000) #define SNDRV_PCM_RATE_8000_96000 (SNDRV_PCM_RATE_8000_48000|SNDRV_PCM_RATE_64000|\ SNDRV_PCM_RATE_88200|SNDRV_PCM_RATE_96000) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 4adaee62ef33..72e3eab09518 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2442,12 +2442,12 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 +#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 13 #error "Change this table" #endif static const unsigned int rates[] = { - 5512, 8000, 11025, 16000, 22050, 32000, 44100, + 5512, 8000, 11025, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, 352800, 384000 }; -- 2.36.1

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[PATCH v3 0/2] [PATCH v3 0/2] Add pinctrl support adsp bypass platforms
by Srinivasa Rao Mandadapu 11 Jun '22

11 Jun '22

This patch set is to make clock voting optinal for adsp bypass sc7280 platforms. Changes Since V2: -- Remove redundant lpi pincontrol variant data structure and compatible entry. -- Add adsp bypass mode boolean param check. -- Remove compatible name in dt bindings. -- Update dt binding bypass mode boolean param. Changes Since V1: -- Update commit message. Srinivasa Rao Mandadapu (2): dt-bindings: pinctrl: qcom: sc7280: Add boolean param for ADSP bypass platforms pinctrl: qcom: sc7280: Add clock optional check for ADSP bypass targets .../bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 9 ++++++++- drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 3 +++ drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 2 +- 3 files changed, 12 insertions(+), 2 deletions(-) -- 2.7.4

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[PATCH] ASoC: samsung: i2s: Check before clk_unregister() not needed
by Yihao Han 10 Jun '22

10 Jun '22

clk_unregister() already checks the clk ptr using !clk || WARN_ON_ONCE(IS_ERR(clk)) so there is no need to check it again before calling it. Signed-off-by: Yihao Han <hanyihao(a)vivo.com> --- sound/soc/samsung/i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 70c827162be4..84e21f7fc179 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1247,8 +1247,7 @@ static void i2s_unregister_clocks(struct samsung_i2s_priv *priv) int i; for (i = 0; i < priv->clk_data.clk_num; i++) { - if (!IS_ERR(priv->clk_table[i])) - clk_unregister(priv->clk_table[i]); + clk_unregister(priv->clk_table[i]); } } -- 2.17.1

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[PATCH] ASoC: qdsp6: q6apm-dai: unprepare stream if its already prepared
by Srinivas Kandagatla 10 Jun '22

10 Jun '22

prepare callback can be called multiple times, so unprepare the stream if its already prepared. Without this DSP is not happy to setting the params on a already prepared graph. Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support") Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6apm-dai.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 19c4a90ec1ea..ee59ef36b85a 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -147,6 +147,12 @@ static int q6apm_dai_prepare(struct snd_soc_component *component, cfg.num_channels = runtime->channels; cfg.bit_width = prtd->bits_per_sample; + if (prtd->state) { + /* clear the previous setup if any */ + q6apm_graph_stop(prtd->graph); + q6apm_unmap_memory_regions(prtd->graph, substream->stream); + } + prtd->pcm_count = snd_pcm_lib_period_bytes(substream); prtd->pos = 0; /* rate and channels are sent to audio driver */ -- 2.25.1

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