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

ALSA PM issues
by Rizwan Mubarak Chikodi 21 Jun '23

21 Jun '23

Hi ALSA team, We are using the standard ALSA driver framework for capture and playback in our Linux platform. In our platform arecord and aplay work as expected using I2S-CODEC, In one of our applications we have to do Suspend to RAM (S2R) operation. When Suspend to RAM triggers while playing audio through I2S-CODEC, The playback fails to resume-back once the system is resumed. We have thoroughly verified the drivers of the I2S controller and CODEC, But have not observed any issues, We are suspecting that the issue might be in the ASOC layer. Following are the details: * kernel version: 5.10.41 * Platform: RisCV64 We are playing audio in one terminal using " aplay (using standard alsa-utils package)" command and running S2R command " echo mem > /sys/power/state " in other terminals to suspend and resume the system. We are facing input/error once the system resumed back as mentioned below, --------------------------------------------------------------------------------------------- root@XX-SOC-BSP-31JAN23-REL12-v1:~# aplay -Dhw:2,0 3.wav Playing WAVE '3.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_write:2061: write error: Input/output error ---------------------------------------------------------------------------------------------- And we found that in " pcm_lib.c " the tout time is 2.5 seconds, once we do Suspend to resume we observe the tout is zero. if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { /* use wait time from substream if available */ if (substream->wait_time) { wait_time = substream->wait_time; } else { wait_time = 10; if (runtime->rate) { long t = runtime->period_size * 2 / runtime->rate; wait_time = max(t, wait_time); } wait_time = msecs_to_jiffies(wait_time * 1000); printk("******sound/core/pcm_lib.c ::: tout = %ld ****",wait_time); } } ....... tout = schedule_timeout(wait_time); Thanks & regards Rizwan Chikodi

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acp63: add initial support for AMD Pink Sardine - ACP63
by GitHub pull_request - opened 21 Jun '23

21 Jun '23

alsa-project/alsa-ucm-conf pull request #327 was opened from pma1: Link: #326 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/327 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/327.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 0/2] AMD Vangogh support for NAU8821/MAX98388
by Cristian Ciocaltea 21 Jun '23

21 Jun '23

This patch series extends the Vangogh machine driver to support a variant based on the Nuvoton NAU88L21 Codec and the Analog Devices MAX98388 Speaker Amplifier. Cristian Ciocaltea (2): ASoC: amd: vangogh: Add ACPI probe support ASoC: amd: vangogh: Add support for NAU8821/MAX98388 variant sound/soc/amd/Kconfig | 5 +- sound/soc/amd/vangogh/acp5x-mach.c | 155 ++++++++++++++++++++++++----- sound/soc/amd/vangogh/acp5x.h | 2 +- sound/soc/amd/vangogh/pci-acp5x.c | 3 - 4 files changed, 134 insertions(+), 31 deletions(-) -- 2.41.0

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[PATCH] sound: make all 'class' structures const
by Greg Kroah-Hartman 21 Jun '23

21 Jun '23

From: Ivan Orlov <ivan.orlov0322(a)gmail.com> Now that the driver core allows for struct class to be in read-only memory, making all 'class' structures to be declared at build time placing them into read-only memory, instead of having to be dynamically allocated at load time. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Ivan Orlov <ivan.orlov0322(a)gmail.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Geoff Levand <geoff(a)infradead.org> Cc: Thierry Reding <treding(a)nvidia.com> Cc: "Uwe Kleine-König" <u.kleine-koenig(a)pengutronix.de> Cc: alsa-devel(a)alsa-project.org Suggested-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- include/sound/core.h | 2 +- sound/core/control_led.c | 2 +- sound/core/init.c | 4 ++-- sound/sound_core.c | 23 ++++++++++++----------- 4 files changed, 16 insertions(+), 15 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 3edc4ab08774..bbcecfff723d 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -232,7 +232,7 @@ static inline struct device *snd_card_get_device_link(struct snd_card *card) extern int snd_major; extern int snd_ecards_limit; -extern struct class *sound_class; +extern const struct class sound_class; #ifdef CONFIG_SND_DEBUG extern struct dentry *sound_debugfs_root; #endif diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 3cadd40100f3..ee77547bf8dc 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -737,7 +737,7 @@ static int __init snd_ctl_led_init(void) unsigned int group; device_initialize(&snd_ctl_led_dev); - snd_ctl_led_dev.class = sound_class; + snd_ctl_led_dev.class = &sound_class; snd_ctl_led_dev.release = snd_ctl_led_dev_release; dev_set_name(&snd_ctl_led_dev, "ctl-led"); if (device_add(&snd_ctl_led_dev)) { diff --git a/sound/core/init.c b/sound/core/init.c index df0c22480375..baef2688d0cf 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -129,7 +129,7 @@ void snd_device_initialize(struct device *dev, struct snd_card *card) device_initialize(dev); if (card) dev->parent = &card->card_dev; - dev->class = sound_class; + dev->class = &sound_class; dev->release = default_release; } EXPORT_SYMBOL_GPL(snd_device_initialize); @@ -331,7 +331,7 @@ static int snd_card_init(struct snd_card *card, struct device *parent, device_initialize(&card->card_dev); card->card_dev.parent = parent; - card->card_dev.class = sound_class; + card->card_dev.class = &sound_class; card->card_dev.release = release_card_device; card->card_dev.groups = card->dev_groups; card->dev_groups[0] = &card_dev_attr_group; diff --git a/sound/sound_core.c b/sound/sound_core.c index 4f6911274d56..d81fed1c1226 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -23,9 +23,6 @@ static inline int init_oss_soundcore(void) { return 0; } static inline void cleanup_oss_soundcore(void) { } #endif -struct class *sound_class; -EXPORT_SYMBOL(sound_class); - MODULE_DESCRIPTION("Core sound module"); MODULE_AUTHOR("Alan Cox"); MODULE_LICENSE("GPL"); @@ -37,6 +34,12 @@ static char *sound_devnode(const struct device *dev, umode_t *mode) return kasprintf(GFP_KERNEL, "snd/%s", dev_name(dev)); } +const struct class sound_class = { + .name = "sound", + .devnode = sound_devnode, +}; +EXPORT_SYMBOL(sound_class); + static int __init init_soundcore(void) { int rc; @@ -45,21 +48,19 @@ static int __init init_soundcore(void) if (rc) return rc; - sound_class = class_create("sound"); - if (IS_ERR(sound_class)) { + rc = class_register(&sound_class); + if (rc) { cleanup_oss_soundcore(); - return PTR_ERR(sound_class); + return rc; } - sound_class->devnode = sound_devnode; - return 0; } static void __exit cleanup_soundcore(void) { cleanup_oss_soundcore(); - class_destroy(sound_class); + class_unregister(&sound_class); } subsys_initcall(init_soundcore); @@ -276,7 +277,7 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati } } - device_create(sound_class, dev, MKDEV(SOUND_MAJOR, s->unit_minor), + device_create(&sound_class, dev, MKDEV(SOUND_MAJOR, s->unit_minor), NULL, "%s", s->name+6); return s->unit_minor; @@ -302,7 +303,7 @@ static void sound_remove_unit(struct sound_unit **list, int unit) if (!preclaim_oss) __unregister_chrdev(SOUND_MAJOR, p->unit_minor, 1, p->name); - device_destroy(sound_class, MKDEV(SOUND_MAJOR, p->unit_minor)); + device_destroy(&sound_class, MKDEV(SOUND_MAJOR, p->unit_minor)); kfree(p); } } -- 2.41.0

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[PATCH v3] ALSA: compress: allow setting codec params after next track
by Srinivas Kandagatla 21 Jun '23

21 Jun '23

For gapless playback it is possible that each track can have different codec profile with same decoder, for example we have WMA album, we may have different tracks as WMA v9, WMA v10 and so on Or if DSP's like QDSP have abililty to switch decoders on single stream for each track, then this call could be used to set new codec parameters. Existing code does not allow to change this profile while doing gapless playback. Reuse existing SNDRV_COMPRESS_SET_PARAMS to set this new track params along some additional checks to enforce proper state machine. With this new changes now the user can call SNDRV_COMPRESS_SET_PARAMS anytime after setting next track and additional check in write should also ensure that params are set before writing new data. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v2: simplified the logic to allow set_params and udated the documentation accordingly Documentation/sound/designs/compress-offload.rst | 11 ++++++----- sound/core/compress_offload.c | 5 ++++- 2 files changed, 10 insertions(+), 6 deletions(-) diff --git a/Documentation/sound/designs/compress-offload.rst b/Documentation/sound/designs/compress-offload.rst index 935f325dbc77..655624f77092 100644 --- a/Documentation/sound/designs/compress-offload.rst +++ b/Documentation/sound/designs/compress-offload.rst @@ -268,11 +268,12 @@ with setting of meta_data and signalling for next track :: | | | V | +----------+ - | | | - | |NEXT_TRACK| - | | | - | +----------+ - | | + | compr_set_params() | | + | +-----------|NEXT_TRACK| + | | | | + | | +--+-------+ + | | | | + | +--------------+ | | | | | compr_partial_drain() | | diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 243acad89fd3..30f73097447b 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -589,7 +589,7 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) struct snd_compr_params *params; int retval; - if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) { + if (stream->runtime->state == SNDRV_PCM_STATE_OPEN || stream->next_track) { /* * we should allow parameter change only when stream has been * opened not in other cases @@ -612,6 +612,9 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) if (retval) goto out; + if (stream->next_track) + goto out; + stream->metadata_set = false; stream->next_track = false; -- 2.21.0

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Thinkpad Z16 Gen2 Mic mute LED not switching on when press F4
by GitHub issues - opened 21 Jun '23

21 Jun '23

alsa-project/alsa-ucm-conf issue #326 was opened from pma1: The mic and sound output works fine. alsamixer can switch on/off the F4/Mic-mute LED automatically. But when press Fn+F4, the mic-mute led is not working. /sys/class/sound/ctl-led/mic/card1# cat list 129 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/326 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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use-case: Add new LatencyOffsetMs field
by GitHub pull_request - opened 21 Jun '23

21 Jun '23

alsa-project/alsa-lib pull request #332 was opened from afq984: Some drivers do not report accurate latencies. Add a LatencyOffsetMs UCM Value allow adding a static offset to the reported latency. Request URL : https://github.com/alsa-project/alsa-lib/pull/332 Patch URL : https://github.com/alsa-project/alsa-lib/pull/332.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH V4 0/9] ASoC: amd: ps: add SoundWire support
by Vijendar Mukunda 21 Jun '23

21 Jun '23

This patch series add support for - Platform device creation for SoundWire Manager instances and PDM controller. - SoundWire DMA driver. - Interrupt handling for SoundWire manager related interrupts, SoundWire DMA interrupts and ACP error interrupts. - ACP PCI driver PM ops modification with respect to SoundWire Power modes. Changes since v3: - use pdev_config instead of pdev_mask in the code. - define platform device configuration macros rather than enums - add comments for MIPI DisCo version - refactor SoundWire DMA start/stop sequence using single function - refactor "acp_reset" flag related functionality. Changes since v2: - add comments in irq handler. - remove workqueue for SoundWire DMA interrupts and use thread implementation for DMA interrupt handling. - add error checks in sdw_amd_scan_controller() - remove passing "acp_lock" as platform resource for SoundWire DMA driver and PDM driver. - retrieve "acp_lock" reference from parent driver directly and use the reference in SoundWire DMA driver. - add handling for acp pci driver probe even when no ACP PDM or SoundWire manager platform devices created. - Fix indentation for acp_sdw_dma_count structure variables. - Use macro instead of hard coded values for FIFO offset and PTE offset. - Change pm_runtime enable sequence in SoundWire DMA driver probe function. - Refactor system level resume callback in SoundWire DMA Changes since v1: - update "soundwire" as "SoundWire" in code. - add comments for platform device mask and platform device count - remove unused variables in acp pci driver private data structure - refactor dma enable register structures in SoundWire DMA driver - add TODO comments in IRQ handler - update IRQ mask values using bit macros - Fix build error reported in Makefile - rename "sdw_dma_stream_instance" structure name as "acp_sdw_dma_stream" Vijendar Mukunda (9): ASoC: amd: ps: create platform devices based on acp config ASoC: amd: ps: handle SoundWire interrupts in acp pci driver ASoC: amd: ps: add SoundWire dma driver ASoC: amd: ps: add SoundWire dma driver dma ops ASoC: amd: ps: add support for SoundWire DMA interrupts ASoC: amd: ps: add pm ops support for SoundWire dma driver ASoC: amd: ps: enable SoundWire dma driver build ASoC: amd: update comments in Kconfig file ASoC: amd: ps: add acp_reset flag check in acp pci driver pm ops. sound/soc/amd/Kconfig | 3 +- sound/soc/amd/ps/Makefile | 2 + sound/soc/amd/ps/acp63.h | 172 ++++++++++- sound/soc/amd/ps/pci-ps.c | 419 +++++++++++++++++++++++-- sound/soc/amd/ps/ps-sdw-dma.c | 555 ++++++++++++++++++++++++++++++++++ 5 files changed, 1115 insertions(+), 36 deletions(-) create mode 100644 sound/soc/amd/ps/ps-sdw-dma.c -- 2.34.1

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[PATCH v3 0/9] add snd_soc_{of_}get_dlc()
by Kuninori Morimoto 21 Jun '23

21 Jun '23

Hi Mark Current soc-core.c has snd_soc_{of_}get_dai_name() to get DAI name for dlc (snd_soc_dai_link_component). It gets .dai_name, but we need .of_node too. Therefor user need to arrange. It will be more useful if it gets both .dai_name and .of_node. This patch adds snd_soc_{of_}get_dlc() for it, and convert to use it. v2 -> v3 - care loongson on [2/9] v1 -> v2 - add patch for loongson Link: https://lore.kernel.org/r/87cz1rdmp0.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87mt12hp75.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (9): ASoC: soc-core.c: add snd_soc_{of_}get_dlc() ASoC: soc-core.c: add index on snd_soc_of_get_dai_name() ASoC: fsl: use snd_soc_{of_}get_dlc() ASoC: qcom: use snd_soc_{of_}get_dlc() ASoC: meson: use snd_soc_{of_}get_dlc() ASoC: samsung: use snd_soc_{of_}get_dlc() ASoC: loongson: use snd_soc_{of_}get_dlc() ASoC: soc-core.c: use snd_soc_{of_}get_dlc() ASoC: simple-card.c: use snd_soc_{of_}get_dlc() include/sound/soc.h | 8 ++- sound/soc/fsl/imx-card.c | 14 +---- sound/soc/fsl/imx-rpmsg.c | 3 +- sound/soc/generic/simple-card-utils.c | 4 +- sound/soc/generic/simple-card.c | 4 +- sound/soc/loongson/loongson_card.c | 34 ++++------ sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/meson/gx-card.c | 3 +- sound/soc/meson/meson-card-utils.c | 16 ++--- sound/soc/meson/meson-card.h | 3 +- sound/soc/qcom/common.c | 13 +--- sound/soc/samsung/odroid.c | 16 +---- sound/soc/soc-core.c | 77 ++++++++++++----------- 14 files changed, 79 insertions(+), 121 deletions(-) -- 2.25.1

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[PATCH v2 0/9] add snd_soc_{of_}get_dlc()
by Kuninori Morimoto 21 Jun '23

21 Jun '23

Hi Mark Current soc-core.c has snd_soc_{of_}get_dai_name() to get DAI name for dlc (snd_soc_dai_link_component). It gets .dai_name, but we need .of_node too. Therefor user need to arrange. It will be more useful if it gets both .dai_name and .of_node. This patch adds snd_soc_{of_}get_dlc() for it, and convert to use it. v1 -> v2 - care loongson Link: https://lore.kernel.org/r/87mt12hp75.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (9): ASoC: soc-core.c: add snd_soc_{of_}get_dlc() ASoC: soc-core.c: add index on snd_soc_of_get_dai_name() ASoC: fsl: use snd_soc_{of_}get_dlc() ASoC: qcom: use snd_soc_{of_}get_dlc() ASoC: meson: use snd_soc_{of_}get_dlc() ASoC: samsung: use snd_soc_{of_}get_dlc() ASoC: loongson: use snd_soc_{of_}get_dlc() ASoC: soc-core.c: use snd_soc_{of_}get_dlc() ASoC: simple-card.c: use snd_soc_{of_}get_dlc() include/sound/soc.h | 8 ++- sound/soc/fsl/imx-card.c | 14 +---- sound/soc/fsl/imx-rpmsg.c | 3 +- sound/soc/generic/simple-card-utils.c | 4 +- sound/soc/generic/simple-card.c | 4 +- sound/soc/loongson/loongson_card.c | 34 ++++------ sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/meson/gx-card.c | 3 +- sound/soc/meson/meson-card-utils.c | 16 ++--- sound/soc/meson/meson-card.h | 3 +- sound/soc/qcom/common.c | 13 +--- sound/soc/samsung/odroid.c | 16 +---- sound/soc/soc-core.c | 77 ++++++++++++----------- 14 files changed, 79 insertions(+), 121 deletions(-) -- 2.25.1

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