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

[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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[PATCH] ASoC: Remove unused hw_write_t type
by Amadeusz Sławiński 10 Jun '22

10 Jun '22

Commit 81da8a0b7975 ("ASoC: remove codec hw_write/control_data") removed use of hw_write_t in struct snd_soc_codec, but it left type definition. Fully clean it up. Fixes: 81da8a0b7975 ("ASoC: remove codec hw_write/control_data") Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/soc.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f20f5f890794..b276dcb5d4e8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -408,8 +408,6 @@ struct snd_soc_jack_pin; struct snd_soc_jack_gpio; -typedef int (*hw_write_t)(void *,const char* ,int); - enum snd_soc_pcm_subclass { SND_SOC_PCM_CLASS_PCM = 0, SND_SOC_PCM_CLASS_BE = 1, -- 2.25.1

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[PATCH] ASoC: Intel: avs: Fix parsing UUIDs in topology
by Amadeusz Sławiński 10 Jun '22

10 Jun '22

Use correct type for parsing UUIDs, this eliminates warning present, when compiling with W=1. Fixes: 34ae2cd53673 ("ASoC: Intel: avs: Add topology parsing infrastructure") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/avs/topology.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index 0d11cc8aab0b..6a06fe387d13 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -128,10 +128,10 @@ struct avs_tplg_token_parser { static int avs_parse_uuid_token(struct snd_soc_component *comp, void *elem, void *object, u32 offset) { - struct snd_soc_tplg_vendor_value_elem *tuple = elem; + struct snd_soc_tplg_vendor_uuid_elem *tuple = elem; guid_t *val = (guid_t *)((u8 *)object + offset); - guid_copy((guid_t *)val, (const guid_t *)&tuple->value); + guid_copy((guid_t *)val, (const guid_t *)&tuple->uuid); return 0; } -- 2.25.1

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