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

[PATCH] soundwire: qcom: fix storing port config out-of-bounds
by Krzysztof Kozlowski 21 Jun '23

21 Jun '23

The 'qcom_swrm_ctrl->pconfig' has size of QCOM_SDW_MAX_PORTS (14), however we index it starting from 1, not 0, to match real port numbers. This can lead to writing port config past 'pconfig' bounds and overwriting next member of 'qcom_swrm_ctrl' struct. Reported also by smatch: drivers/soundwire/qcom.c:1269 qcom_swrm_get_port_config() error: buffer overflow 'ctrl->pconfig' 14 <= 14 Fixes: 9916c02ccd74 ("soundwire: qcom: cleanup internal port config indexing") Cc: <stable(a)vger.kernel.org> Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <error27(a)gmail.com> Link: https://lore.kernel.org/r/202305201301.sCJ8UDKV-lkp@intel.com/ Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/soundwire/qcom.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 7cb1b7eba814..88a772075907 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -202,7 +202,8 @@ struct qcom_swrm_ctrl { u32 intr_mask; u8 rcmd_id; u8 wcmd_id; - struct qcom_swrm_port_config pconfig[QCOM_SDW_MAX_PORTS]; + /* Port numbers are 1 - 14 */ + struct qcom_swrm_port_config pconfig[QCOM_SDW_MAX_PORTS + 1]; struct sdw_stream_runtime *sruntime[SWRM_MAX_DAIS]; enum sdw_slave_status status[SDW_MAX_DEVICES + 1]; int (*reg_read)(struct qcom_swrm_ctrl *ctrl, int reg, u32 *val); -- 2.34.1

4 5

[PATCH 0/3] ALSA: ump: Preliminary changes for gadget driver
by Takashi Iwai 21 Jun '23

21 Jun '23

Hi, this is a small patch set to change the UMP core for the upcoming gadget driver support. Basically exporting a couple of helper functions and adding a flag to suppress the internal UMP handling. No functional changes by those alone. Takashi === Takashi Iwai (3): ALSA: ump: Add helper to change MIDI protocol ALSA: ump: Add no_process_stream flag ALSA: ump: Export snd_ump_receive_ump_val() include/sound/ump.h | 4 ++++ sound/core/ump.c | 46 +++++++++++++++++++++++++++++++++++---------- 2 files changed, 40 insertions(+), 10 deletions(-) -- 2.35.3

1 3

[PATCH 1/2] soundwire: bus: Prevent lockdep asserts when stream has multiple buses
by Richard Fitzgerald 21 Jun '23

21 Jun '23

Give the bus_lock and msg_lock of each bus a different unique key so that it is possible to acquire the locks of multiple buses without lockdep asserting a possible deadlock. Using mutex_init() to initialize a mutex gives all those mutexes the same lock class. Lockdep checking treats it as an error to attempt to take a mutex while already holding a mutex of the same class. This causes a lockdep assert when sdw_acquire_bus_lock() attempts to lock multiple buses, and when do_bank_switch() takes multiple msg_lock. [ 138.697350] WARNING: possible recursive locking detected [ 138.697366] 6.3.0-test #1 Tainted: G E [ 138.697380] -------------------------------------------- [ 138.697394] play/903 is trying to acquire lock: [ 138.697409] ffff99b8c41aa8c8 (&bus->bus_lock){+.+.}-{3:3}, at: sdw_prepare_stream+0x52/0x2e0 [ 138.697443] but task is already holding lock: [ 138.697468] ffff99b8c41af8c8 (&bus->bus_lock){+.+.}-{3:3}, at: sdw_prepare_stream+0x52/0x2e0 [ 138.697493] other info that might help us debug this: [ 138.697521] Possible unsafe locking scenario: [ 138.697540] CPU0 [ 138.697550] ---- [ 138.697559] lock(&bus->bus_lock); [ 138.697570] lock(&bus->bus_lock); [ 138.697581] *** DEADLOCK *** Giving each mutex a unique key allows multiple to be held without triggering a lockdep assert. But note that it does not allow them to be taken in one order then a different order. If two mutexes are taken in the order A, B then they must always be taken in that order otherwise they could deadlock. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/soundwire/bus.c | 15 +++++++++++++-- include/linux/soundwire/sdw.h | 3 +++ 2 files changed, 16 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index b44f8d0affa6..dba920ec88f6 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -69,8 +69,17 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, return -EINVAL; } - mutex_init(&bus->msg_lock); - mutex_init(&bus->bus_lock); + /* + * Give each bus_lock and msg_lock a unique key so that lockdep won't + * trigger a deadlock warning when the locks of several buses are + * grabbed during configuration of a multi-bus stream. + */ + lockdep_register_key(&bus->msg_lock_key); + __mutex_init(&bus->msg_lock, "msg_lock", &bus->msg_lock_key); + + lockdep_register_key(&bus->bus_lock_key); + __mutex_init(&bus->bus_lock, "bus_lock", &bus->bus_lock_key); + INIT_LIST_HEAD(&bus->slaves); INIT_LIST_HEAD(&bus->m_rt_list); @@ -181,6 +190,8 @@ void sdw_bus_master_delete(struct sdw_bus *bus) sdw_master_device_del(bus); sdw_bus_debugfs_exit(bus); + lockdep_unregister_key(&bus->bus_lock_key); + lockdep_unregister_key(&bus->msg_lock_key); ida_free(&sdw_bus_ida, bus->id); } EXPORT_SYMBOL(sdw_bus_master_delete); diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index c076a3f879b3..f523ceabd059 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -5,6 +5,7 @@ #define __SOUNDWIRE_H #include <linux/bug.h> +#include <linux/lockdep_types.h> #include <linux/mod_devicetable.h> #include <linux/bitfield.h> @@ -907,7 +908,9 @@ struct sdw_bus { struct list_head slaves; DECLARE_BITMAP(assigned, SDW_MAX_DEVICES); struct mutex bus_lock; + struct lock_class_key bus_lock_key; struct mutex msg_lock; + struct lock_class_key msg_lock_key; int (*compute_params)(struct sdw_bus *bus); const struct sdw_master_ops *ops; const struct sdw_master_port_ops *port_ops; -- 2.30.2

3 4

[PATCH] soundwire: intel_ace2x: fix SND_SOC_SOF_HDA_MLINK dependency
by Arnd Bergmann 21 Jun '23

21 Jun '23

From: Arnd Bergmann <arnd(a)arndb.de> The ace2x driver can be build with or without mlink support, but when SND_SOC_SOF_HDA_MLINK is set to =m and soundwire is built-in, it fails with a link error: ld.lld: error: undefined symbol: hdac_bus_eml_sdw_wait_syncpu_unlocked >>> referenced by intel_ace2x.c >>> drivers/soundwire/intel_ace2x.o:(intel_link_power_up) in archive vmlinux.a ld.lld: error: undefined symbol: hdac_bus_eml_sdw_sync_arm_unlocked >>> referenced by intel_ace2x.c >>> drivers/soundwire/intel_ace2x.o:(intel_sync_arm) in archive vmlinux.a Add a Kconfig dependency that prevents that broken configuration but still allows soundwire to be a loadable module instead. Fixes: 4d1e2464a1104 ("soundwire: intel_ace2x: add sync_arm/sync_go helpers") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- drivers/soundwire/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig index fa71c9a36df7a..4d8f3b7024ae5 100644 --- a/drivers/soundwire/Kconfig +++ b/drivers/soundwire/Kconfig @@ -37,6 +37,7 @@ config SOUNDWIRE_INTEL select SOUNDWIRE_GENERIC_ALLOCATION select AUXILIARY_BUS depends on ACPI && SND_SOC + depends on SND_SOC_SOF_HDA_MLINK || !SND_SOC_SOF_HDA_MLINK help SoundWire Intel Master driver. If you have an Intel platform which has a SoundWire Master then -- 2.39.2

2 1

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

1 0

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

1 0

[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

1 3

[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

2 1

[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

3 2

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

1 0