January 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: soc-compress.c: fixup private_data on snd_soc_new_compress()
by Kuninori Morimoto 28 Jan '23

28 Jan '23

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit d3268a40d4b19f ("ASoC: soc-compress.c: fix NULL dereference") enables DPCM capture, but it should independent from playback. This patch fixup it. Fixes: d3268a40d4b1 ("ASoC: soc-compress.c: fix NULL dereference") Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- > Need Intel's Tested-by ? sound/soc/soc-compress.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index b2a6bbb4e807..ed9e910659ab 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -616,7 +616,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->fe_compr = 1; if (rtd->dai_link->dpcm_playback) be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - else if (rtd->dai_link->dpcm_capture) + if (rtd->dai_link->dpcm_capture) be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else { -- 2.25.1

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[PATCH] ASoC: fsl-asoc-card: constify fsl_asoc_card_dai
by Luca Ceresoli 28 Jan '23

28 Jan '23

This is never modified and can be made const. Signed-off-by: Luca Ceresoli <luca.ceresoli(a)bootlin.com> --- sound/soc/fsl/fsl-asoc-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 2f25358196ee..00118a75d126 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -297,7 +297,7 @@ SND_SOC_DAILINK_DEFS(hifi_be, DAILINK_COMP_ARRAY(COMP_EMPTY()), DAILINK_COMP_ARRAY(COMP_DUMMY())); -static struct snd_soc_dai_link fsl_asoc_card_dai[] = { +static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { /* Default ASoC DAI Link*/ { .name = "HiFi", -- 2.34.1

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From 644473b181f0f310e428301a2ed459f912eec7ea Mon Sep 17 00:00:00 2001
by Peter Ujfalusi 28 Jan '23

28 Jan '23

Hi, The following series will enable multi-stream support for playback and capture streams. Currently only a single PCM can be connected to a DAI, with the multi-stream support it is possible to connect multiple PCMs to a single DAI. To achieve this we need to make sure that DAIs/AIF are only set up once since other stream could be connected to it later. We also need to introduce reference or use counting for widgets to make sure that they are not going to be destroyed while other streams are still using them. With the multi-stream support we also need to extend our current locking scheme which worked well for simple paths. The first patch was sent to the mailing list, but it is not yet applied and I have included in this series as it is a direct dependency: https://lore.kernel.org/alsa-devel/20230117121615.25690-1-peter.ujfalusi@li… Regards, Peter --- Peter Ujfalusi (3): ASoC: SOF: Avoid double decrementing use_count in sof_widget_setup on error ASoC: SOF: Protect swidget->use_count with mutex for kcontrol access race ASoC: SOF: ipc4-pcm: Do not run the trigger pipelines if no spipe is stored Ranjani Sridharan (15): ASoC: SOF: ipc4-topology: No need to unbind routes within a pipeline ASoC: soc-pcm: Export widget_in_list() ASoC: SOF: sof-audio: Set up/free DAI/AIF widgets only once ASoC: SOF: sof-audio: Only process widgets in the connected widget list ASoC: SOF: pcm: do not free widgets during suspend trigger ASoC: SOF: topology: Set IPC-specific trigger order for DAI links ASoC: SOF: Introduce PCM setup/free PCM IPC ops ASoC: SOF: ipc4-pcm: Define pcm_setup/free ops ASoC: SOF: ipc4: Add flag to skip triggering pipelines during FE DAI trigger ASoC: SOF: sof-audio: Populate the PCM stream pipeline_info ASoC: SOF: ipc4-pcm: Use the PCM stream's pipeline_info during trigger ASoC: SOF: Introduce struct snd_sof_pipeline ASoC: SOF: ipc4-pcm: Rename 'data' variable to trigger_list ASoC: SOF: ipc4-pcm: Implement pipeline trigger reference counting ASoC: SOF: ipc4-topology: Protect pipeline free with mutex include/sound/soc-dpcm.h | 2 + include/sound/sof/ipc4/header.h | 3 + sound/soc/soc-pcm.c | 3 +- sound/soc/sof/core.c | 1 + sound/soc/sof/intel/hda-dai.c | 92 ++------- sound/soc/sof/ipc3-control.c | 46 +++-- sound/soc/sof/ipc3-topology.c | 32 ++- sound/soc/sof/ipc4-control.c | 33 ++- sound/soc/sof/ipc4-pcm.c | 343 +++++++++++++++++++++++++------- sound/soc/sof/ipc4-priv.h | 2 + sound/soc/sof/ipc4-topology.c | 48 ++++- sound/soc/sof/ipc4-topology.h | 12 ++ sound/soc/sof/ipc4.c | 2 + sound/soc/sof/pcm.c | 5 +- sound/soc/sof/sof-audio.c | 226 ++++++++++++++------- sound/soc/sof/sof-audio.h | 54 ++++- sound/soc/sof/sof-priv.h | 1 + sound/soc/sof/topology.c | 114 +++++++---- 18 files changed, 714 insertions(+), 305 deletions(-) -- 2.39.1

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[PATCH] ASoC: soc-compress: Reposition and add pcm_mutex
by 강신형 28 Jan '23

28 Jan '23

If panic_on_warn is set and compress stream(DPCM) is started, then kernel panic occurred because card->pcm_mutex isn't held appropriately. In the following functions, warning were issued at this line "snd_soc_dpcm_mutex_assert_held". static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { ... snd_soc_dpcm_mutex_assert_held(fe); ... } void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { ... snd_soc_dpcm_mutex_assert_held(fe); ... } void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, int stream, int action) { ... snd_soc_dpcm_mutex_assert_held(rtd); ... } int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) { ... snd_soc_dpcm_mutex_assert_held(fe); ... } These functions are called by soc_compr_set_params_fe, soc_compr_open_fe and soc_compr_free_fe without pcm_mutex locking. And this is call stack. [ 414.527841][ T2179] pc : dpcm_process_paths+0x5a4/0x750 [ 414.527848][ T2179] lr : dpcm_process_paths+0x37c/0x750 [ 414.527945][ T2179] Call trace: [ 414.527949][ T2179] dpcm_process_paths+0x5a4/0x750 [ 414.527955][ T2179] soc_compr_open_fe+0xb0/0x2cc [ 414.527972][ T2179] snd_compr_open+0x180/0x248 [ 414.527981][ T2179] snd_open+0x15c/0x194 [ 414.528003][ T2179] chrdev_open+0x1b0/0x220 [ 414.528023][ T2179] do_dentry_open+0x30c/0x594 [ 414.528045][ T2179] vfs_open+0x34/0x44 [ 414.528053][ T2179] path_openat+0x914/0xb08 [ 414.528062][ T2179] do_filp_open+0xc0/0x170 [ 414.528068][ T2179] do_sys_openat2+0x94/0x18c [ 414.528076][ T2179] __arm64_sys_openat+0x78/0xa4 [ 414.528084][ T2179] invoke_syscall+0x48/0x10c [ 414.528094][ T2179] el0_svc_common+0xbc/0x104 [ 414.528099][ T2179] do_el0_svc+0x34/0xd8 [ 414.528103][ T2179] el0_svc+0x34/0xc4 [ 414.528125][ T2179] el0t_64_sync_handler+0x8c/0xfc [ 414.528133][ T2179] el0t_64_sync+0x1a0/0x1a4 [ 414.528142][ T2179] Kernel panic - not syncing: panic_on_warn set ... So, I reposition and add pcm_mutex to resolve lockdep error. Signed-off-by: Shinhyung Kang <s47.kang(a)samsung.com> --- sound/soc/soc-compress.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 870f13e1d389..7bce5088b455 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -149,6 +149,8 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) if (ret < 0) goto be_err; + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); + /* calculate valid and active FE <-> BE dpcms */ dpcm_process_paths(fe, stream, &list, 1); fe->dpcm[stream].runtime = fe_substream->runtime; @@ -184,7 +186,6 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; - mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_activate(fe, stream); mutex_unlock(&fe->card->pcm_mutex); @@ -215,7 +216,6 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_deactivate(fe, stream); - mutex_unlock(&fe->card->pcm_mutex); fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; @@ -234,6 +234,8 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) dpcm_be_disconnect(fe, stream); + mutex_unlock(&fe->card->pcm_mutex); + fe->dpcm[stream].runtime = NULL; snd_soc_link_compr_shutdown(cstream, 0); @@ -409,8 +411,9 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, ret = snd_soc_link_compr_set_params(cstream); if (ret < 0) goto out; - + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); + mutex_unlock(&fe->card->pcm_mutex); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; out: -- 2.21.0

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[PATCH] ASoC: SOF: ipc4-topology: No need to unbind routes within a pipeline
by Peter Ujfalusi 28 Jan '23

28 Jan '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> The FW currently ignores unbinding routes if the source and sink widgets belong to the same pipeline. So no need to send the IPC at all in the first place. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 59f4d42f9011..89f2dc4f564d 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1744,12 +1744,19 @@ static int sof_ipc4_route_free(struct snd_sof_dev *sdev, struct snd_sof_route *s struct sof_ipc4_fw_module *sink_fw_module = sink_widget->module_info; struct sof_ipc4_msg msg = {{ 0 }}; u32 header, extension; - int ret; + int ret = 0; dev_dbg(sdev->dev, "unbind modules %s:%d -> %s:%d\n", src_widget->widget->name, sroute->src_queue_id, sink_widget->widget->name, sroute->dst_queue_id); + /* + * routes belonging to the same pipeline will be disconnected by the FW when the pipeline + * is freed. So avoid sending this IPC which will be ignored by the FW anyway. + */ + if (src_widget->pipe_widget == sink_widget->pipe_widget) + goto out; + header = src_fw_module->man4_module_entry.id; header |= SOF_IPC4_MOD_INSTANCE(src_widget->instance_id); header |= SOF_IPC4_MSG_TYPE_SET(SOF_IPC4_MOD_UNBIND); @@ -1768,7 +1775,7 @@ static int sof_ipc4_route_free(struct snd_sof_dev *sdev, struct snd_sof_route *s if (ret < 0) dev_err(sdev->dev, "failed to unbind modules %s -> %s\n", src_widget->widget->name, sink_widget->widget->name); - +out: sof_ipc4_put_queue_id(sink_widget, sroute->dst_queue_id, SOF_PIN_TYPE_SINK); sof_ipc4_put_queue_id(src_widget, sroute->src_queue_id, SOF_PIN_TYPE_SOURCE); -- 2.39.0

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[GIT PULL] ASoC fixes for v6.2-rc5
by Mark Brown 28 Jan '23

28 Jan '23

The following changes since commit 242fc66ae6e1e2b8519daacc7590a73cd0e8a6e4: ASoC: fsl-asoc-card: Fix naming of AC'97 CODEC widgets (2023-01-10 12:45:22 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.2-rc5 for you to fetch changes up to e18c6da62edc780e4f4f3c9ce07bdacd69505182: ASoC: cs42l56: fix DT probe (2023-01-26 17:42:36 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.2 An unfortunately large batch of fixes here, the numbers amplified by several repeated fixes for patterns of bugs in multiple drivers. Most of this is in the x86 drivers which are very actively developed, the implementation of PCI shutdown is a fix for issues with spamming warnings into the logs with a leaked reference to the i915 driver. ---------------------------------------------------------------- Amadeusz Sławiński (1): ASoC: Intel: avs: Implement PCI shutdown Andy Shevchenko (6): ASoC: amd: acp-es8336: Drop reference count of ACPI device after use ASoC: Intel: bytcht_es8316: Drop reference count of ACPI device after use ASoC: Intel: bytcr_rt5651: Drop reference count of ACPI device after use ASoC: Intel: bytcr_rt5640: Drop reference count of ACPI device after use ASoC: Intel: bytcr_wm5102: Drop reference count of ACPI device after use ASoC: Intel: sof_es8336: Drop reference count of ACPI device after use Arnd Bergmann (1): ASoC: cs42l56: fix DT probe Bard Liao (3): ASoC: SOF: sof-audio: start with the right widget type ASoC: SOF: sof-audio: unprepare when swidget->use_count > 0 ASoC: SOF: keep prepare/unprepare widgets in sink path Dan Carpenter (1): ASoC: SOF: ipc4-mtrace: prevent underflow in sof_ipc4_priority_mask_dfs_write() Krzysztof Kozlowski (1): ASoC: codecs: wsa883x: correct playback min/max rates Kuninori Morimoto (1): ASoC: hdmi-codec: zero clear HDMI pdata Mark Brown (2): ASoC: SOF: sof-audio: Fixes for widget prepare and ASoC: Intel: set dpcm_capture for amps Peter Ujfalusi (1): ASoC: SOF: sof-audio: prepare_widgets: Check swidget for NULL on sink failure Pierre-Louis Bossart (4): ASoC: Intel: sof_rt5682: always set dpcm_capture for amplifiers ASoC: Intel: sof_cs42l42: always set dpcm_capture for amplifiers ASoC: Intel: sof_nau8825: always set dpcm_capture for amplifiers ASoC: Intel: sof_ssp_amp: always set dpcm_capture for amplifiers Ranjani Sridharan (1): ASoC: SOF: sof-audio: skip prepare/unprepare if swidget is NULL Syed Saba Kareem (1): ASoC: amd: yc: Add DMI support for new acer/emdoor platforms fengwk (1): ASoC: amd: yc: Add Xiaomi Redmi Book Pro 15 2022 into DMI table .../gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 1 + sound/soc/amd/acp-es8336.c | 6 ++++-- sound/soc/amd/yc/acp6x-mach.c | 21 +++++++++++++++++++ sound/soc/codecs/cs42l56.c | 6 ------ sound/soc/codecs/wsa883x.c | 4 ++-- sound/soc/intel/avs/core.c | 24 ++++++++++++++++++++++ sound/soc/intel/boards/bytcht_es8316.c | 20 ++++++++++-------- sound/soc/intel/boards/bytcr_rt5640.c | 12 +++++------ sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/sof_cs42l42.c | 3 +++ sound/soc/intel/boards/sof_es8336.c | 14 +++++++------ sound/soc/intel/boards/sof_nau8825.c | 5 +++-- sound/soc/intel/boards/sof_rt5682.c | 5 +++-- sound/soc/intel/boards/sof_ssp_amp.c | 5 ++--- sound/soc/sof/ipc4-mtrace.c | 7 ++++--- sound/soc/sof/sof-audio.c | 16 ++++++++------- 17 files changed, 104 insertions(+), 49 deletions(-)

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[PATCH 1/4] ASoC: codecs: tas5720: split a tas5720_mute_soc_component() function
by Alexandru Ardelean 28 Jan '23

28 Jan '23

This is to be re-used in tas5720_mute() (which is part of the dai_ops) and also in the tas5720_fault_check_work() hook. The benefit here isn't too great (now). It's only when we add support for a new device with a slightly different regmap that this becomes more useful. Signed-off-by: Alexandru Ardelean <alex(a)shruggie.ro> --- sound/soc/codecs/tas5720.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index 3885c0bf0b01..0587c9bb3c4e 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -199,9 +199,8 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, return ret; } -static int tas5720_mute(struct snd_soc_dai *dai, int mute, int direction) +static int tas5720_mute_soc_component(struct snd_soc_component *component, int mute) { - struct snd_soc_component *component = dai->component; int ret; ret = snd_soc_component_update_bits(component, TAS5720_DIGITAL_CTRL2_REG, @@ -214,6 +213,11 @@ static int tas5720_mute(struct snd_soc_dai *dai, int mute, int direction) return 0; } +static int tas5720_mute(struct snd_soc_dai *dai, int mute, int direction) +{ + return tas5720_mute_soc_component(dai->component, mute); +} + static void tas5720_fault_check_work(struct work_struct *work) { struct tas5720_data *tas5720 = container_of(work, struct tas5720_data, @@ -318,8 +322,7 @@ static int tas5720_codec_probe(struct snd_soc_component *component) expected_device_id, device_id); /* Set device to mute */ - ret = snd_soc_component_update_bits(component, TAS5720_DIGITAL_CTRL2_REG, - TAS5720_MUTE, TAS5720_MUTE); + ret = tas5720_mute_soc_component(component, 1); if (ret < 0) goto error_snd_soc_component_update_bits; -- 2.34.1

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[PATCH] scripts/spelling.txt: add "exsits" pattern and fix typo instances
by Luca Ceresoli 27 Jan '23

27 Jan '23

Fix typos and add the following to the scripts/spelling.txt: exsits||exists Signed-off-by: Luca Ceresoli <luca.ceresoli(a)bootlin.com> --- drivers/infiniband/ulp/iser/iscsi_iser.c | 2 +- net/openvswitch/flow_table.c | 2 +- scripts/spelling.txt | 1 + sound/soc/fsl/fsl-asoc-card.c | 2 +- 4 files changed, 4 insertions(+), 3 deletions(-) diff --git a/drivers/infiniband/ulp/iser/iscsi_iser.c b/drivers/infiniband/ulp/iser/iscsi_iser.c index 620ae5b2d80d..6b7603765383 100644 --- a/drivers/infiniband/ulp/iser/iscsi_iser.c +++ b/drivers/infiniband/ulp/iser/iscsi_iser.c @@ -446,7 +446,7 @@ iscsi_iser_conn_create(struct iscsi_cls_session *cls_session, * @is_leading: indicate if this is the session leading connection (MCS) * * Return: zero on success, $error if iscsi_conn_bind fails and - * -EINVAL in case end-point doesn't exsits anymore or iser connection + * -EINVAL in case end-point doesn't exists anymore or iser connection * state is not UP (teardown already started). */ static int iscsi_iser_conn_bind(struct iscsi_cls_session *cls_session, diff --git a/net/openvswitch/flow_table.c b/net/openvswitch/flow_table.c index 0a0e4c283f02..cfac54cbafdf 100644 --- a/net/openvswitch/flow_table.c +++ b/net/openvswitch/flow_table.c @@ -1012,7 +1012,7 @@ static int flow_mask_insert(struct flow_table *tbl, struct sw_flow *flow, mask = flow_mask_find(tbl, new); if (!mask) { - /* Allocate a new mask if none exsits. */ + /* Allocate a new mask if none exists. */ mask = mask_alloc(); if (!mask) return -ENOMEM; diff --git a/scripts/spelling.txt b/scripts/spelling.txt index ded8bcfc0247..0147bd8dc6e2 100644 --- a/scripts/spelling.txt +++ b/scripts/spelling.txt @@ -625,6 +625,7 @@ exeuction||execution existance||existence existant||existent exixt||exist +exsits||exists exlcude||exclude exlcusive||exclusive exmaple||example diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 8d14b5593658..2f25358196ee 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -811,7 +811,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.num_links = 1; if (asrc_pdev) { - /* DPCM DAI Links only if ASRC exsits */ + /* DPCM DAI Links only if ASRC exists */ priv->dai_link[1].cpus->of_node = asrc_np; priv->dai_link[1].platforms->of_node = asrc_np; priv->dai_link[2].codecs->dai_name = codec_dai_name; -- 2.34.1

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[PATCH 0/6] Fix default DMIC gain on AMD PDM drivers
by Mario Limonciello 27 Jan '23

27 Jan '23

It's been reported that a number of laptops have a low volume level from the digital microphone compared to Windows. AMD offers a register that can adjust the gain for PDM which is not configured at maximum gain by default. To fix this change the default for all 3 drivers to raise the gain but also offer a module parameter. The module parameter can be used for debugging if the gain is too high on a given laptop. This is intentionally split into multiple patches for default and parameter so that if the default really does behave better universally we can bring it back to stable too later. Mario Limonciello (6): ASoC: amd: yc: Adjust the gain for PDM DMIC ASoC: amd: yc: Add a module parameter to influence pdm_gain ASoC: amd: renoir: Adjust the gain for PDM DMIC ASoC: amd: renoir: Add a module parameter to influence pdm_gain ASoC: amd: ps: Adjust the gain for PDM DMIC ASoC: amd: ps: Add a module parameter to influence pdm_gain sound/soc/amd/ps/acp63.h | 2 +- sound/soc/amd/ps/ps-pdm-dma.c | 7 ++++++- sound/soc/amd/renoir/acp3x-pdm-dma.c | 7 ++++++- sound/soc/amd/renoir/rn_acp3x.h | 2 +- sound/soc/amd/yc/acp6x-pdm-dma.c | 7 ++++++- sound/soc/amd/yc/acp6x.h | 2 +- 6 files changed, 21 insertions(+), 6 deletions(-) -- 2.34.1

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[PATCH] pcm: hw: fix rate regression
by Jonas Holmberg 27 Jan '23

27 Jan '23

Add a line that seems to have been removed by mistake in commit 7f2d6c3aac3505ceee4b0d3e8b3ca423ce29b070. Signed-off-by: Jonas Holmberg <jonashg(a)axis.com> --- src/pcm/pcm_hw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 0588ce5e..daa3e1ff 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -356,7 +356,7 @@ static int snd_pcm_hw_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) if (hw->rates.min > 0) { err = _snd_pcm_hw_param_set_minmax(params, SND_PCM_HW_PARAM_RATE, hw->rates.min, 0, hw->rates.max + 1, -1); - + if (err < 0) return err; } -- 2.39.1

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