Alsa-devel January 2022

alsa-devel@alsa-project.org

134 participants
273 discussions

[PATCH v2] ASoC: soc-compress: prevent the potentially use of null pointer
by Jiasheng Jiang 02 Mar '22

02 Mar '22

There is one call trace that snd_soc_register_card() ->snd_soc_bind_card()->soc_init_pcm_runtime() ->snd_soc_dai_compress_new()->snd_soc_new_compress(). In the trace the 'codec_dai' transfers from card->dai_link, and we can see from the snd_soc_add_pcm_runtime() in snd_soc_bind_card() that, if value of card->dai_link->num_codecs is 0, then 'codec_dai' could be null pointer caused by index out of bound in 'asoc_rtd_to_codec(rtd, 0)'. And snd_soc_register_card() is called by various platforms. Therefore, it is better to add the check in the case of misusing. And because 'cpu_dai' has already checked in soc_init_pcm_runtime(), there is no need to check again. Adding the check as follow, then if 'codec_dai' is null, snd_soc_new_compress() will not pass through the check 'if (playback + capture != 1)', avoiding the leftover use of 'codec_dai'. Fixes: 467fece ("ASoC: soc-dai: move snd_soc_dai_stream_valid() to soc-dai.c") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/soc-compress.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index b4f5935..67c3df1 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -535,12 +535,14 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* check client and interface hw capabilities */ - if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) - playback = 1; - if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) - capture = 1; + if (codec_dai) { + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) + playback = 1; + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) + capture = 1; + } /* * Compress devices are unidirectional so only one of the directions -- 2.7.4

2 1

[PATCH v10 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 24 Feb '22

24 Feb '22

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V9: -- Change individual clock voting to bulk clock voting of lpass-sc7280 platform driver. -- Remove redundant clocks in lpass variant structure. -- Add mclk for MI2S based headset path. -- Remove unused lpass variant structure members in lpass header. Changes Since V8: -- Fix errors in sc7280 lpass cpu dt-bindings. -- Move to quicinc domain email id's. Changes Since V7: -- Fix indentation errors. -- Bisect patches to avoid interdependency. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: SC7280: Update config for building codec dma drivers ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add regmap config support for codec dma driver ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 75 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 275 +++++++++ sound/soc/qcom/lpass-cpu.c | 244 +++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 617 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 445 +++++++++++++++ sound/soc/qcom/lpass.h | 145 +++++ 9 files changed, 1856 insertions(+), 87 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- 2.7.4

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[RESEND, v9 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 24 Feb '22

24 Feb '22

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V8: -- Fix errors in sc7280 lpass cpu dt-bindings. -- Move to quicinc domain email id's. Changes Since V7: -- Fix indentation errors. -- Bisect patches to avoid interdependency. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: SC7280: Update config for building codec dma drivers ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add regmap config support for codec dma driver ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 75 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 275 +++++++++ sound/soc/qcom/lpass-cpu.c | 244 +++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 617 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 441 +++++++++++++++ sound/soc/qcom/lpass.h | 162 ++++++ 9 files changed, 1869 insertions(+), 87 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- 2.7.4

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[RFC PATCH 1/3] ASoC: simple-card-utils: add support for componants provideing jack events via set_jack
by Carl Philipp Klemm 20 Feb '22

20 Feb '22

This allows componants that want a jack to report state on to do so by calling set_jack on components implementing this function. Im not entirely sure this is the right way to do this so RFC Signed-off-by: Carl Philipp Klemm <philipp(a)uvos.xyz> --- include/sound/simple_card_utils.h | 6 ++-- sound/soc/generic/simple-card-utils.c | 47 +++++++++++++++++++-------- 2 files changed, 36 insertions(+), 17 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 51b3b485a92e..547ad537613d 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -57,8 +57,8 @@ struct asoc_simple_priv { struct prop_nums num; unsigned int mclk_fs; } *dai_props; - struct asoc_simple_jack hp_jack; - struct asoc_simple_jack mic_jack; + struct asoc_simple_jack *hp_jack; + struct asoc_simple_jack *mic_jack; struct snd_soc_dai_link *dai_link; struct asoc_simple_dai *dais; struct snd_soc_dai_link_component *dlcs; @@ -173,7 +173,7 @@ int asoc_simple_parse_pin_switches(struct snd_soc_card *card, char *prefix); int asoc_simple_init_jack(struct snd_soc_card *card, - struct asoc_simple_jack *sjack, + struct asoc_simple_jack **sjack, int is_hp, char *prefix, char *pin); int asoc_simple_init_priv(struct asoc_simple_priv *priv, struct link_info *li); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 10c63b73900c..1899feba16cc 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -395,6 +395,7 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *props = simple_priv_to_props(priv, rtd->num); struct asoc_simple_dai *dai; + struct snd_soc_component *component; int i, ret; for_each_prop_dai_codec(props, i, dai) { @@ -412,6 +413,21 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; + for_each_rtd_components(rtd, i, component) { + if (component->driver->set_jack) { + if (!priv->hp_jack) { + priv->hp_jack = devm_kzalloc(priv->snd_card.dev, + sizeof(*priv->hp_jack), GFP_KERNEL); + snd_soc_card_jack_new(&priv->snd_card, + "Headphones", + SND_JACK_HEADPHONE, + &priv->hp_jack->jack, + NULL, 0); + } + snd_soc_component_set_jack(component, &priv->hp_jack->jack, NULL); + } + } + return 0; } EXPORT_SYMBOL_GPL(asoc_simple_dai_init); @@ -554,7 +570,7 @@ int asoc_simple_parse_pin_switches(struct snd_soc_card *card, EXPORT_SYMBOL_GPL(asoc_simple_parse_pin_switches); int asoc_simple_init_jack(struct snd_soc_card *card, - struct asoc_simple_jack *sjack, + struct asoc_simple_jack **sjack, int is_hp, char *prefix, char *pin) { @@ -569,8 +585,6 @@ int asoc_simple_init_jack(struct snd_soc_card *card, if (!prefix) prefix = ""; - sjack->gpio.gpio = -ENOENT; - if (is_hp) { snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix); pin_name = pin ? pin : "Headphones"; @@ -588,21 +602,26 @@ int asoc_simple_init_jack(struct snd_soc_card *card, return -EPROBE_DEFER; if (gpio_is_valid(det)) { - sjack->pin.pin = pin_name; - sjack->pin.mask = mask; + struct asoc_simple_jack *sjack_d; + + sjack = devm_kzalloc(dev, sizeof(*(*sjack)), GFP_KERNEL); + sjack_d = *sjack; + + sjack_d->pin.pin = pin_name; + sjack_d->pin.mask = mask; - sjack->gpio.name = gpio_name; - sjack->gpio.report = mask; - sjack->gpio.gpio = det; - sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); - sjack->gpio.debounce_time = 150; + sjack_d->gpio.name = gpio_name; + sjack_d->gpio.report = mask; + sjack_d->gpio.gpio = det; + sjack_d->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); + sjack_d->gpio.debounce_time = 150; snd_soc_card_jack_new(card, pin_name, mask, - &sjack->jack, - &sjack->pin, 1); + &sjack_d->jack, + &sjack_d->pin, 1); - snd_soc_jack_add_gpios(&sjack->jack, 1, - &sjack->gpio); + snd_soc_jack_add_gpios(&sjack_d->jack, 1, + &sjack_d->gpio); } return 0; -- 2.34.1

3 14

[PATCH 0/1] sound: add quirk for Huawei D15
by Mauro Carvalho Chehab 18 Feb '22

18 Feb '22

Hi Pierre-Louis, Based on my tests, Huawei D15 (Intel) uses SSP0 on es8336. Add a quirk for it. Please notice that, currently, only the internal speaker is working. The topology for the internal microphone and for the headphones is wrong. Enabling/disabling the other two quirks (GPIO and/or DMIC) doesn't cause any audible results, nor change the devices listed on pavucontrol (tested with pipewire-pulse). Mauro Carvalho Chehab (1): ASoC: Intel: sof_es8336: add quirk for Huawei D15 2021 sound/soc/intel/boards/sof_es8336.c | 8 ++++++++ 1 file changed, 8 insertions(+) -- 2.33.1

3 4

[PATCH v3 0/5] Tegra234 APE support
by Sameer Pujar 17 Feb '22

17 Feb '22

This adds APE support on Tegra234 and enables sound card on Jetson AGX Orin platform. Changelog ========= v2->v3 ------ * Fixed DT build error related to power-domain and iommu reference. Reported-by: kernel test robot <lkp(a)intel.com> v1->v2 ------ * Rebased patches on latest linux-next Mohan Kumar (1): ASoC: tegra: Update AHUB driver for Tegra234 Sameer Pujar (4): dt-bindings: Add Tegra234 APE support dt-bindings: Document Tegra234 APE support arm64: tegra: Add audio devices on Tegra234 arm64: tegra: APE sound card for Jetson AGX Orin .../bindings/bus/nvidia,tegra210-aconnect.yaml | 1 + .../bindings/dma/nvidia,tegra210-adma.yaml | 4 +- .../bindings/interrupt-controller/arm,gic.yaml | 1 + .../bindings/sound/nvidia,tegra186-dspk.yaml | 4 +- .../bindings/sound/nvidia,tegra210-admaif.yaml | 4 +- .../bindings/sound/nvidia,tegra210-adx.yaml | 1 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 1 + .../bindings/sound/nvidia,tegra210-amx.yaml | 3 + .../bindings/sound/nvidia,tegra210-dmic.yaml | 1 + .../bindings/sound/nvidia,tegra210-i2s.yaml | 1 + .../bindings/sound/nvidia,tegra210-mixer.yaml | 1 + .../bindings/sound/nvidia,tegra210-mvc.yaml | 1 + .../bindings/sound/nvidia,tegra210-sfc.yaml | 1 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 1783 ++++++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra234.dtsi | 432 +++++ include/dt-bindings/clock/tegra234-clock.h | 74 +- include/dt-bindings/memory/tegra234-mc.h | 7 + include/dt-bindings/power/tegra234-powergate.h | 9 + sound/soc/tegra/tegra210_ahub.c | 146 +- sound/soc/tegra/tegra210_ahub.h | 4 +- 20 files changed, 2473 insertions(+), 6 deletions(-) create mode 100644 include/dt-bindings/power/tegra234-powergate.h -- 2.7.4

3 8

[PATCH v17 0/2] mtk: add ADSP mailbox controller for MT8195
by allen-kh.cheng 15 Feb '22

15 Feb '22

From: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> Mediatek ADSP IPC is used to send notification or short message between processors with dsp. It will place the message to the share buffer and will access the ADSP mailbox registers to kick dsp. Two mailboxes used to send notification or short message between processors with dsp. This patchset was tested and confirmed on MT8195 cherry board. Based on linux-next changes since v16: - fix WARNING: modpost: missing MODULE_LICENSE() in drivers/mailbox /mtk-adsp-mailbox.o. Add MODULE_LICENSE in the last line. - Due to WARNING: Missing or malformed SPDX-License-Identifier tag in line 1 in checkpatch, we don't remove SPDX-License in line 1. changes since v15: - add paragraphs to describe the config symbols fully changes since v14: - add inline for get_mtk_adsp_mbox_priv changes since v13: - rebase on v5.16-rc8 - add back ptr check from of_device_get_match_data changes since v12: - remove of_device_get_match_data ptr check: of_device_get_match_data(dev) will never going to return NULL. driver probe with compatible mediatek,mt8195-adsp-mbox. changes since v11: - remove useless MODULE_LICENSE changes since v10: - split up v9 into two separate submissions changes since v9: - rename adsp_mbox_chan_ops to mtk_adsp_mbox_chan_ops changes since v8: - remove struct adsp_mbox_ch_info - add get_mtk_adsp_mbox_priv - use mtk_adsp_mbox_priv va_mboxreg address in adsp mbox driver - add struct mtk_adsp_mbox_cfg for DSP mbox register offset - remove adsp mbox register offset hard code define - remove mtk-adsp-ipc.h reference in adsp mbox driver changes since v7: - add mtk prefix for adsp ipc functions - rename adsp_mbox_ch_info to mtk_adsp_mbox_ch_info - remove incorrect reviewers in commit message changes since v6: - dt-bindings: change example dtsi node. - rename config MTK_ADSP_IPC_MBOX to MTK_ADSP_MBOX - remove unused variable - add reviewers changes since v5: - remove some redundant code changes since v4: - use switch ... case in adsp_ipc_recv - add error handling path for chan_name pointer - refine some code to be concise changes since v3: - reorder MTK_ADSP_IPC_MBOX config - remove some redundant code - remove lock in mtk-adsp-mailbox changes since v2: - separate adsp_mailbox into two instances changes since v1: - fix dt_binding_check error Allen-KH Cheng (2): dt-bindings: mailbox: add mtk adsp-mbox document mailbox: mediatek: add support for adsp mailbox controller .../bindings/mailbox/mtk,adsp-mbox.yaml | 52 ++++++ drivers/mailbox/Kconfig | 9 + drivers/mailbox/Makefile | 2 + drivers/mailbox/mtk-adsp-mailbox.c | 176 ++++++++++++++++++ 4 files changed, 239 insertions(+) create mode 100644 Documentation/devicetree/bindings/mailbox/mtk,adsp-mbox.yaml create mode 100644 drivers/mailbox/mtk-adsp-mailbox.c -- 2.18.0

2 4

[PATCH] ALSA: pcm: accept the OPEN state for snd_pcm_stop()
by Jaroslav Kysela 14 Feb '22

14 Feb '22

It may be useful to completely invalidate streaming under some circumstances like the USB gadget detach. This case is a bit different than the complete disconnection. The applications should be notified that the PCM streaming is no longer available, but the recovery may be expected. This patch adds support for SNDRV_PCM_STATE_OPEN passed to snd_pcm_stop() function. Only the hw_free ioctl is allowed to free the buffers in this state for a full recovery operation or the PCM file handle must be closed. In the OPEN state, ioctls return EBADFD error (with the added hw_free exception above). The applications which are not aware about this new state transition (and recovery) will fail with an error. This operation is expected. Link: https://lore.kernel.org/alsa-devel/4e17c99b-1d8a-8ebe-972c-9b1299952ff8@ivi… Cc: Pavel Hofman <pavel.hofman(a)ivitera.com> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/pcm.h | 1 + sound/core/pcm.c | 1 + sound/core/pcm_native.c | 12 +++++++++++- 3 files changed, 13 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 33451f8ff755..4de1c2c91109 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -467,6 +467,7 @@ struct snd_pcm_substream { /* -- assigned files -- */ int ref_count; atomic_t mmap_count; + atomic_t queued_hw_free; unsigned int f_flags; void (*pcm_release)(struct snd_pcm_substream *); struct pid *pid; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 6fd3677685d7..8dc7e99b2113 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -694,6 +694,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) snd_pcm_group_init(&substream->self_group); list_add_tail(&substream->link_list, &substream->self_group.substreams); atomic_set(&substream->mmap_count, 0); + atomic_set(&substream->queued_hw_free, 0); prev = substream; } return 0; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 621883e71194..118ad3f26f4a 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -686,10 +686,13 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_stream_lock_irq(substream); switch (runtime->status->state) { case SNDRV_PCM_STATE_OPEN: + if (atomic_read(&substream->queued_hw_free)) + goto __badfd; case SNDRV_PCM_STATE_SETUP: case SNDRV_PCM_STATE_PREPARED: break; default: +__badfd: snd_pcm_stream_unlock_irq(substream); return -EBADFD; } @@ -829,6 +832,7 @@ static int do_hw_free(struct snd_pcm_substream *substream) result = substream->ops->hw_free(substream); if (substream->managed_buffer_alloc) snd_pcm_lib_free_pages(substream); + atomic_set(&substream->queued_hw_free, 0); return result; } @@ -1454,6 +1458,8 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, } wake_up(&runtime->sleep); wake_up(&runtime->tsleep); + if (state == SNDRV_PCM_STATE_OPEN) + atomic_set(&substream->queued_hw_free, 1); } static const struct action_ops snd_pcm_action_stop = { @@ -1469,6 +1475,9 @@ static const struct action_ops snd_pcm_action_stop = { * * The state of each stream is then changed to the given state unconditionally. * + * If the requested state is OPEN, the stream is invalidated and + * the application must call hw_free to recover the operation. + * * Return: Zero if successful, or a negative error code. */ int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t state) @@ -2637,7 +2646,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) snd_pcm_drop(substream); if (substream->hw_opened) { - if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN || + atomic_read(&substream->queued_hw_free)) do_hw_free(substream); substream->ops->close(substream); substream->hw_opened = 0; -- 2.33.1

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[PATCH 00/19] soundwire: stream: cleanup of 'stream' support
by Bard Liao 11 Feb '22

11 Feb '22

This series revisits the SoundWire 'sdw_stream' support to split allocation and configuration steps. This is necessary if for example the routines are called multiple times from the hw_params stage. This also helps with better error handling. Pierre-Louis Bossart (19): soundwire: stream: remove unused parameter in sdw_stream_add_slave soundwire: stream: add slave runtime to list earlier soundwire: stream: simplify check on port range soundwire: stream: add alloc/config/free helpers for ports soundwire: stream: split port allocation and configuration loops soundwire: stream: split alloc and config in two functions soundwire: stream: add 'slave' prefix for port range checks soundwire: stream: group sdw_port and sdw_master/slave_port functions soundwire: stream: simplify sdw_alloc_master_rt() soundwire: stream: split sdw_alloc_master_rt() in alloc and config soundwire: stream: move sdw_alloc_slave_rt() before 'master' helpers soundwire: stream: split sdw_alloc_slave_rt() in alloc and config soundwire: stream: group sdw_stream_ functions soundwire: stream: rename and move master/slave_rt_free routines soundwire: stream: move list addition to sdw_slave_alloc_rt() soundwire: stream: separate alloc and config within sdw_stream_add_xxx() soundwire: stream: introduce sdw_slave_rt_find() helper soundwire: stream: sdw_stream_add_ functions can be called multiple times soundwire: stream: make enable/disable/deprepare idempotent drivers/soundwire/stream.c | 960 +++++++++++++++++++++---------------- 1 file changed, 547 insertions(+), 413 deletions(-) -- 2.17.1

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[PATCH v2 0/6] ASoC: dt-bindings: samsung: convert to dtschema
by Krzysztof Kozlowski 10 Feb '22

10 Feb '22

Hi, The patchset is based on Rob's sound-dai changes: https://lore.kernel.org/all/20220126231427.1638089-1-robh@kernel.org/ Changes since v1: 1. Correct samsung,snow cpu/sound-dai. Best regards, Krzysztof Krzysztof Kozlowski (6): ASoC: dt-bindings: samsung,aries-wm8994: require sound-dai property ASoC: dt-bindings: samsung,arndale: convert to dtschema ASoC: dt-bindings: samsung,arndale: document ALC5631 ASoC: dt-bindings: samsung,smdk5250: convert to dtschema ASoC: dt-bindings: samsung,snow: convert to dtschema ASoC: dt-bindings: samsung,tm2: convert to dtschema .../devicetree/bindings/sound/arndale.txt | 25 ------ .../bindings/sound/samsung,aries-wm8994.yaml | 4 + .../bindings/sound/samsung,arndale.yaml | 45 +++++++++++ .../bindings/sound/samsung,smdk-wm8994.txt | 14 ---- .../bindings/sound/samsung,smdk5250.yaml | 38 +++++++++ .../bindings/sound/samsung,snow.yaml | 74 +++++++++++++++++ .../bindings/sound/samsung,tm2-audio.txt | 42 ---------- .../bindings/sound/samsung,tm2.yaml | 80 +++++++++++++++++++ .../devicetree/bindings/sound/snow.txt | 31 ------- 9 files changed, 241 insertions(+), 112 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/arndale.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,arndale.yaml delete mode 100644 Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml create mode 100644 Documentation/devicetree/bindings/sound/samsung,snow.yaml delete mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2.yaml delete mode 100644 Documentation/devicetree/bindings/sound/snow.txt -- 2.32.0

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Alsa-devel January 2022