November 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH v3 1/1] ALSA: compress: allow pause and resume during draining
by Gyeongtaek Lee 27 Nov '20

27 Nov '20

With a stream with low bitrate, user can't pause or resume the stream near the end of the stream because current ALSA doesn't allow it. If the stream has very low bitrate enough to store whole stream into the buffer, user can't do anything except stop the stream and then restart it from the first because most of applications call draining after sending last frame to the kernel. If pause, resume are allowed during draining, user experience can be enhanced. To prevent malfunction in HW drivers which don't support pause during draining, pause during draining will only work if HW driver enable this feature explicitly by calling snd_compr_use_pause_in_draining(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- include/sound/compress_driver.h | 16 ++++++++++++++ sound/core/compress_offload.c | 39 ++++++++++++++++++++++++++------- 2 files changed, 47 insertions(+), 8 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 70cbc5095e72..277087f635f3 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -67,6 +67,7 @@ struct snd_compr_runtime { * @metadata_set: metadata set flag, true when set * @next_track: has userspace signal next track transition, true when set * @partial_drain: undergoing partial_drain for stream, true when set + * @pause_in_draining: paused during draining state, true when set * @private_data: pointer to DSP private data * @dma_buffer: allocated buffer if any */ @@ -80,6 +81,7 @@ struct snd_compr_stream { bool metadata_set; bool next_track; bool partial_drain; + bool pause_in_draining; void *private_data; struct snd_dma_buffer dma_buffer; }; @@ -142,6 +144,7 @@ struct snd_compr_ops { * @direction: Playback or capture direction * @lock: device lock * @device: device id + * @use_pause_in_draining: allow pause in draining, true when set */ struct snd_compr { const char *name; @@ -152,6 +155,7 @@ struct snd_compr { unsigned int direction; struct mutex lock; int device; + bool use_pause_in_draining; #ifdef CONFIG_SND_VERBOSE_PROCFS /* private: */ char id[64]; @@ -166,6 +170,18 @@ int snd_compress_deregister(struct snd_compr *device); int snd_compress_new(struct snd_card *card, int device, int type, const char *id, struct snd_compr *compr); +/** + * snd_compr_use_pause_in_draining - Allow pause and resume in draining state + * @substream: compress substream to set + * + * Allow pause and resume in draining state. + * Only HW driver supports this transition can call this API. + */ +static inline void snd_compr_use_pause_in_draining(struct snd_compr_stream *substream) +{ + substream->device->use_pause_in_draining = true; +} + /* dsp driver callback apis * For playback: driver should call snd_compress_fragment_elapsed() to let the * framework know that a fragment has been consumed from the ring buffer diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index c1fec932c49d..debc30fcf5b3 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -709,11 +709,22 @@ static int snd_compr_pause(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_RUNNING: + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->device->use_pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->pause_in_draining = true; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + } return retval; } @@ -721,11 +732,22 @@ static int snd_compr_resume(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_PAUSED: + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->pause_in_draining = false; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + } return retval; } @@ -768,6 +790,7 @@ static int snd_compr_stop(struct snd_compr_stream *stream) /* clear flags and stop any drain wait */ stream->partial_drain = false; stream->metadata_set = false; + stream->pause_in_draining = false; snd_compr_drain_notify(stream); stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; -- 2.21.0

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[PATCH] Asoc: qcom: Fix for problem in resume with CRAS
by Srinivasa Rao Mandadapu 27 Nov '20

27 Nov '20

To support playback continuation after resume problem in chrome audio server: Prepare device in platform trigger callback. Make I2s and DMA control registers as non volatile. Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 8 ++------ sound/soc/qcom/lpass-platform.c | 5 +++-- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af684fd..c99be03 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -454,20 +454,16 @@ static bool lpass_cpu_regmap_volatile(struct device *dev, unsigned int reg) struct lpass_variant *v = drvdata->variant; int i; - for (i = 0; i < v->i2s_ports; ++i) - if (reg == LPAIF_I2SCTL_REG(v, i)) - return true; for (i = 0; i < v->irq_ports; ++i) if (reg == LPAIF_IRQSTAT_REG(v, i)) return true; for (i = 0; i < v->rdma_channels; ++i) - if (reg == LPAIF_RDMACURR_REG(v, i) || reg == LPAIF_RDMACTL_REG(v, i)) + if (reg == LPAIF_RDMACURR_REG(v, i)) return true; for (i = 0; i < v->wrdma_channels; ++i) - if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start) || - reg == LPAIF_WRDMACTL_REG(v, i + v->wrdma_channel_start)) + if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start)) return true; return false; diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 80b09de..2b0a7c1 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -481,8 +481,9 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, return -ENOTRECOVERABLE; } switch (cmd) { - case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + lpass_platform_pcmops_prepare(component, substream); + case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ret = regmap_fields_write(dmactl->enable, id, LPAIF_DMACTL_ENABLE_ON); @@ -592,7 +593,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, break; } - return 0; + return ret; } static snd_pcm_uframes_t lpass_platform_pcmops_pointer( -- 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 v3] ASoC: qcom: Fix playback recover problem in suspend resume
by Srinivasa Rao Mandadapu 27 Nov '20

27 Nov '20

To support playback continuation after hard suspend(bypass powerd) and resume: Prepare device in platform trigger callback. Make I2s and DMA control registers as non volatile. Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes Since v1 and v2: -- Subject lines changed sound/soc/qcom/lpass-cpu.c | 8 ++------ sound/soc/qcom/lpass-platform.c | 5 +++-- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af684fd..c99be03 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -454,20 +454,16 @@ static bool lpass_cpu_regmap_volatile(struct device *dev, unsigned int reg) struct lpass_variant *v = drvdata->variant; int i; - for (i = 0; i < v->i2s_ports; ++i) - if (reg == LPAIF_I2SCTL_REG(v, i)) - return true; for (i = 0; i < v->irq_ports; ++i) if (reg == LPAIF_IRQSTAT_REG(v, i)) return true; for (i = 0; i < v->rdma_channels; ++i) - if (reg == LPAIF_RDMACURR_REG(v, i) || reg == LPAIF_RDMACTL_REG(v, i)) + if (reg == LPAIF_RDMACURR_REG(v, i)) return true; for (i = 0; i < v->wrdma_channels; ++i) - if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start) || - reg == LPAIF_WRDMACTL_REG(v, i + v->wrdma_channel_start)) + if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start)) return true; return false; diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 80b09de..2b0a7c1 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -481,8 +481,9 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, return -ENOTRECOVERABLE; } switch (cmd) { - case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + lpass_platform_pcmops_prepare(component, substream); + case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ret = regmap_fields_write(dmactl->enable, id, LPAIF_DMACTL_ENABLE_ON); @@ -592,7 +593,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, break; } - return 0; + return ret; } static snd_pcm_uframes_t lpass_platform_pcmops_pointer( -- 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 v1 0/1] Fix 32 bit format for adau7002
by xuyuqing 27 Nov '20

27 Nov '20

the microphone is attached to external codec(adau7002) instead of rt5682.We need to always use 32 bit format on sc7180 to meet the clock requirement of adau7002: The ADAU7002 requires a BCLK rate that is a minimum of 64× the LRCLK sample rate xuyuqing (1): ASoC: qcom: sc7180: fix 32 bit format for adau7002 sound/soc/qcom/sc7180.c | 3 +++ 1 file changed, 3 insertions(+) -- 2.25.1

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[PATCH 0/2] ASoC: soc-core: share same logic code
by Kuninori Morimoto 27 Nov '20

27 Nov '20

Hi Mark soc-core is using almost same code at some point. The difference is only used parameter. It is just duplicate code, and we can share it. This patch-set sharing code for mute and for suspend/resume. Kuninori Morimoto (2): ASoC: soc-core: add soc_playback_digital_mute() ASoC: soc-core: add soc_dapm_suspend_resume() sound/soc/soc-core.c | 84 ++++++++++++++++++++------------------------ 1 file changed, 38 insertions(+), 46 deletions(-) -- 2.25.1

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[PATCH 1/2] ALSA: hda/generic: Add option to enforce preferred_dacs pairs
by Takashi Iwai 27 Nov '20

27 Nov '20

The generic parser accepts the preferred_dacs[] pairs as a hint for assigning a DAC to each pin, but this hint doesn't work always effectively. Currently it's merely a secondary choice after the trial with the path index failed. This made sometimes it difficult to assign DACs without mimicking the connection list and/or the badness table. This patch adds a new flag, obey_preferred_dacs, that changes the behavior of the parser. As its name stands, the parser obeys the given preferred_dacs[] pairs by skipping the path index matching and giving a high penalty if no DAC is assigned by the pairs. This mode will help for assigning the fixed DACs forcibly from the codec driver. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 12 ++++++++---- sound/pci/hda/hda_generic.h | 1 + 2 files changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index bbb17481159e..8060cc86dfea 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1364,16 +1364,20 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs, struct nid_path *path; hda_nid_t pin = pins[i]; - path = snd_hda_get_path_from_idx(codec, path_idx[i]); - if (path) { - badness += assign_out_path_ctls(codec, path); - continue; + if (!spec->obey_preferred_dacs) { + path = snd_hda_get_path_from_idx(codec, path_idx[i]); + if (path) { + badness += assign_out_path_ctls(codec, path); + continue; + } } dacs[i] = get_preferred_dac(codec, pin); if (dacs[i]) { if (is_dac_already_used(codec, dacs[i])) badness += bad->shared_primary; + } else if (spec->obey_preferred_dacs) { + badness += BAD_NO_PRIMARY_DAC; } if (!dacs[i]) diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index a43f0bb77dae..0886bc81f40b 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -237,6 +237,7 @@ struct hda_gen_spec { unsigned int power_down_unused:1; /* power down unused widgets */ unsigned int dac_min_mute:1; /* minimal = mute for DACs */ unsigned int suppress_vmaster:1; /* don't create vmaster kctls */ + unsigned int obey_preferred_dacs:1; /* obey preferred_dacs assignment */ /* other internal flags */ unsigned int no_analog:1; /* digital I/O only */ -- 2.26.2

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[GIT PULL] ASoC fixes for v5.10-rc5
by Mark Brown 27 Nov '20

27 Nov '20

The following changes since commit 879ee8b6f2bae0cc4a25536f8841db1dbc969523: ASOC: Intel: kbl_rt5663_rt5514_max98927: Do not try to disable disabled clock (2020-11-18 17:59:23 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.10-rc5 for you to fetch changes up to aa4cb898b80a28a610e26d1513e6dd42d995c225: ASoC: rt5682: change SAR voltage threshold (2020-11-26 13:06:18 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.10 A small set of driver specific fixes, plus a new platform quirk from Hans. ---------------------------------------------------------------- Hans de Goede (1): ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 Detachable quirks Luo Meng (1): ASoC: wm_adsp: fix error return code in wm_adsp_load() Shuming Fan (1): ASoC: rt5682: change SAR voltage threshold Srinivasa Rao Mandadapu (1): ASoC: qcom: Fix enabling BCLK and LRCLK in LPAIF invalid state sound/soc/codecs/rt5682.c | 1 + sound/soc/codecs/wm_adsp.c | 1 + sound/soc/intel/boards/bytcr_rt5640.c | 17 ++++++++-- sound/soc/qcom/lpass-cpu.c | 62 +++++++++++++++++------------------ sound/soc/qcom/lpass-lpaif-reg.h | 7 ++++ sound/soc/qcom/lpass-platform.c | 20 +++++++++-- sound/soc/qcom/lpass.h | 1 + 7 files changed, 73 insertions(+), 36 deletions(-)

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[PATCH] ALSA: usb-audio: US16x08: fix value count for level meters
by Hector Martin 27 Nov '20

27 Nov '20

The level meter control returns 34 integers of info. This fixes: snd-usb-audio 3-1:1.0: control 2:0:0:Level Meter:0: access overflow Cc: stable(a)vger.kernel.org Signed-off-by: Hector Martin <marcan(a)marcan.st> --- sound/usb/mixer_us16x08.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index 92b1a6d9c931..bd63a9ce6a70 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -607,7 +607,7 @@ static int snd_us16x08_eq_put(struct snd_kcontrol *kcontrol, static int snd_us16x08_meter_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - uinfo->count = 1; + uinfo->count = 34; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.max = 0x7FFF; uinfo->value.integer.min = 0; -- 2.28.0

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pcm: ioplug: Limit transfer size to buffer boundary
by GitHub pull_request - opened 27 Nov '20

27 Nov '20

alsa-project/alsa-lib pull request #103 was opened from aditpape: Commit 1714332719fc91507ca24dd3567e50d7094b3001 introduced 2nd transfer() call to transfer all remaining available frames. If the prior calculated avail value exceeds the buffer size a too large size value is passed to the underlaying plugin and results in memory corruption if not blocked by plugin internally. Avail values > buffer size can happen if e.g. xrun detection is disabled, as avail is calculated by pure difference between hw and app position. This patch limits 2nd transfer call to remaining rest of a buffer size. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/103 Patch URL : https://github.com/alsa-project/alsa-lib/pull/103.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v2] Asoc: qcom: Fix plaback recover problem in suspend resume
by Srinivasa Rao Mandadapu 27 Nov '20

27 Nov '20

To support playback continuation after hard suspend(bypass powerd) and resume: Prepare device in platform trigger callback. Make I2s and DMA control registers as non volatile. Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes Since v1: -- Subject lines changed sound/soc/qcom/lpass-cpu.c | 8 ++------ sound/soc/qcom/lpass-platform.c | 5 +++-- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af684fd..c99be03 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -454,20 +454,16 @@ static bool lpass_cpu_regmap_volatile(struct device *dev, unsigned int reg) struct lpass_variant *v = drvdata->variant; int i; - for (i = 0; i < v->i2s_ports; ++i) - if (reg == LPAIF_I2SCTL_REG(v, i)) - return true; for (i = 0; i < v->irq_ports; ++i) if (reg == LPAIF_IRQSTAT_REG(v, i)) return true; for (i = 0; i < v->rdma_channels; ++i) - if (reg == LPAIF_RDMACURR_REG(v, i) || reg == LPAIF_RDMACTL_REG(v, i)) + if (reg == LPAIF_RDMACURR_REG(v, i)) return true; for (i = 0; i < v->wrdma_channels; ++i) - if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start) || - reg == LPAIF_WRDMACTL_REG(v, i + v->wrdma_channel_start)) + if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start)) return true; return false; diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 80b09de..2b0a7c1 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -481,8 +481,9 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, return -ENOTRECOVERABLE; } switch (cmd) { - case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + lpass_platform_pcmops_prepare(component, substream); + case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ret = regmap_fields_write(dmactl->enable, id, LPAIF_DMACTL_ENABLE_ON); @@ -592,7 +593,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, break; } - return 0; + return ret; } static snd_pcm_uframes_t lpass_platform_pcmops_pointer( -- 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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