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

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

1 0

[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

2 1

[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

2 5

[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

3 3

[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

2 1

[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

2 1

[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

2 1

[PATCH 0/6] ASoC: SOF: Add support ctx_save with IPC4
by Peter Ujfalusi 10 Jun '22

10 Jun '22

Hi, The context save functionality with IPC4 is triggered by sending a message to the firmware about the pending power down of the primary core by the host. In order to have this functionality implemented in a clean way we need to introduce a new IPC level PM ops for core state management and use that instead of open coding IPC messages here and there. The first patch updates the ctx store/ctx_restore documentation to clarify that they are optional. Regards, Peter --- Peter Ujfalusi (6): ASoC: SOF: make ctx_store and ctx_restore as optional ASoC: SOF: sof_ipc_pm_ops: Add support for DSP core power management ASoC: SOF: ipc3: Add set_core_state pm_ops implementation ASoC: SOF: ipc4: Add set_core_state pm_ops implementation ASoC: SOF: Intel: Switch to use the generic pm_ops.set_core_state ASoC: SOF: ipc4: implement pm ctx_save callback include/sound/sof/ipc4/header.h | 8 +++++++ sound/soc/sof/intel/hda-dsp.c | 15 +++++------- sound/soc/sof/intel/tgl.c | 30 ++++++++---------------- sound/soc/sof/ipc3.c | 18 +++++++++++++++ sound/soc/sof/ipc4.c | 41 +++++++++++++++++++++++++++++++++ sound/soc/sof/sof-priv.h | 6 +++-- 6 files changed, 87 insertions(+), 31 deletions(-) -- 2.36.1

2 7

[PATCH 0/3] ASoC: SOF: ipc3-dtrace: Handle race during initialization
by Peter Ujfalusi 10 Jun '22

10 Jun '22

Hi, This series handles the race which can result missing the first position update after the trace is enabled. In short: the firmware might send the position update (if we have enough trace data generated) after the dma-trace is enabled by the TRACE_DMA_PARAMS_EXT message. Depending on scheduling, load, preemption on Linux side we have seen that occasionally this first position update got missed and we missed reading it out. A new state and more strict handling of host_offset can overcome this issue, making the dtrace more reliable. Regards, Peter --- Peter Ujfalusi (3): ASoC: SOF: ipc3-dtrace: Introduce SOF_DTRACE_INITIALIZING state ASoC: SOF: ipc3-dtrace: Add helper function to update the sdev->host_offset ASoC: SOF: ipc3-dtrace: Return from dtrace_read if there is no new data available sound/soc/sof/ipc3-dtrace.c | 47 ++++++++++++++++++++++++++++++------- 1 file changed, 38 insertions(+), 9 deletions(-) -- 2.36.1

2 4

[PATCH 0/2] ASoC: codecs: qualcomm move gain to S8_TLV
by Srinivas Kandagatla 10 Jun '22

10 Jun '22

Move all the digital and IIR gains form using SX_TLV to S8_TLV, these gains are actually 8 bit gains with 7th signed bit and ranges from -84dB to +40dB Tested on DB410c with Headset playback Thanks, Srini Srinivas Kandagatla (2): ASoC: codecs: msm8916-wcd-digital: move gains from SX_TLV to S8_TLV ASoC: codecs: wcd9335: move gains from SX_TLV to S8_TLV sound/soc/codecs/msm8916-wcd-digital.c | 46 +++++++-------- sound/soc/codecs/wcd9335.c | 81 ++++++++++++-------------- 2 files changed, 59 insertions(+), 68 deletions(-) -- 2.25.1

2 3