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

[PATCH v2] ASoC: rt5665: add missed regulator_bulk_disable
by Zhang Shurong 07 Aug '23

07 Aug '23

The driver forgets to call regulator_bulk_disable() Add the missed call to fix it. Fixes: 33ada14a26c8 ("ASoC: add rt5665 codec driver") Signed-off-by: Zhang Shurong <zhang_shurong(a)foxmail.com> --- Changes in v2: - move regulator_bulk_disable() in rt5665_remove() sound/soc/codecs/rt5665.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 83c367af91da..525713c33d71 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4472,6 +4472,8 @@ static void rt5665_remove(struct snd_soc_component *component) struct rt5665_priv *rt5665 = snd_soc_component_get_drvdata(component); regmap_write(rt5665->regmap, RT5665_RESET, 0); + + regulator_bulk_disable(ARRAY_SIZE(rt5665->supplies), rt5665->supplies); } #ifdef CONFIG_PM -- 2.30.2

2 1

[PATCH 00/24] ALSA: Generic PCM copy ops using sockptr_t
by Takashi Iwai 07 Aug '23

07 Aug '23

Hi, this is a patch set to clean up the PCM copy ops using sockptr_t as a "universal" pointer, inspired by the recent patch from Andy Shevchenko: https://lore.kernel.org/r/20230721100146.67293-1-andriy.shevchenko@linux.in… Even though it sounds a bit weird, sockptr_t is a generic type that is used already in wide ranges, and it can fit our purpose, too. With sockptr_t, the former split of copy_user and copy_kernel PCM ops can be unified again gracefully. The patch set introduces the new PCM ops, converting users, and drops the old PCM ops. Most of conversions are straightforward, simply replacing copy_*_user() with copy_*_sockptr() variants. Note that the conversion in ASoC will fix a potential problem of ASoC PCM that has been for long time. Since ASoC component takes care of only copy_user, the conversion form/to kernel space might have been missing. With this patch set, both cases are handled with sockptr_t by a single callback. The patches are lightly tested (with a faked PCM copy implementation on HD-audio), while most of patches are only compile-tested. Takashi === Cc: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Cc: Andrey Utkin <andrey_utkin(a)fastmail.com> Cc: Anton Sviridenko <anton(a)corp.bluecherry.net> Cc: Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> Cc: Banajit Goswami <bgoswami(a)quicinc.com> Cc: Bluecherry Maintainers <maintainers(a)bluecherrydvr.com> Cc: Claudiu Beznea <claudiu.beznea(a)microchip.com> Cc: Ismael Luceno <ismael(a)iodev.co.uk> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Mauro Carvalho Chehab <mchehab(a)kernel.org> Cc: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Cc: Olivier Moysan <olivier.moysan(a)foss.st.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: linux-media(a)vger.kernel.org Cc: xen-devel(a)lists.xenproject.org === Takashi Iwai (24): ALSA: pcm: Add copy ops with universal sockptr_t ALSA: core: Add memory copy helpers between sockptr and iomem ALSA: dummy: Convert to generic PCM copy ops ALSA: gus: Convert to generic PCM copy ops ALSA: emu8000: Convert to generic PCM copy ops ALSA: es1938: Convert to generic PCM copy ops ALSA: korg1212: Convert to generic PCM copy ops ALSA: nm256: Convert to generic PCM copy ops ALSA: rme32: Convert to generic PCM copy ops ALSA: rme96: Convert to generic PCM copy ops ALSA: hdsp: Convert to generic PCM copy ops ALSA: rme9652: Convert to generic PCM copy ops ALSA: sh: Convert to generic PCM copy ops ALSA: xen: Convert to generic PCM copy ops ALSA: pcmtest: Update comment about PCM copy ops media: solo6x10: Convert to generic PCM copy ops ASoC: component: Add generic PCM copy ops ASoC: mediatek: Convert to generic PCM copy ops ASoC: qcom: Convert to generic PCM copy ops ASoC: dmaengine: Convert to generic PCM copy ops ASoC: dmaengine: Use sockptr_t for process callback, too ALSA: doc: Update description for the new PCM copy ops ASoC: pcm: Drop obsoleted PCM copy_user ops ALSA: pcm: Drop obsoleted PCM copy_user and copy_kernel ops .../kernel-api/writing-an-alsa-driver.rst | 59 +++++--------- drivers/media/pci/solo6x10/solo6x10-g723.c | 41 ++-------- include/sound/dmaengine_pcm.h | 2 +- include/sound/pcm.h | 12 +-- include/sound/soc-component.h | 14 ++-- sound/core/memory.c | 39 +++++++++ sound/core/pcm_lib.c | 81 +++++++++---------- sound/core/pcm_native.c | 2 +- sound/drivers/dummy.c | 12 +-- sound/drivers/pcmtest.c | 2 +- sound/isa/gus/gus_pcm.c | 23 +----- sound/isa/sb/emu8000_pcm.c | 79 +++++------------- sound/pci/es1938.c | 31 ++----- sound/pci/korg1212/korg1212.c | 46 +++-------- sound/pci/nm256/nm256.c | 42 ++-------- sound/pci/rme32.c | 50 +++--------- sound/pci/rme96.c | 48 +++-------- sound/pci/rme9652/hdsp.c | 42 ++-------- sound/pci/rme9652/rme9652.c | 46 ++--------- sound/sh/sh_dac_audio.c | 25 +----- sound/soc/atmel/mchp-pdmc.c | 2 +- sound/soc/mediatek/common/mtk-btcvsd.c | 22 ++--- sound/soc/qcom/lpass-platform.c | 12 +-- sound/soc/soc-component.c | 10 +-- sound/soc/soc-generic-dmaengine-pcm.c | 18 ++--- sound/soc/soc-pcm.c | 4 +- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/xen/xen_snd_front_alsa.c | 55 +++---------- 28 files changed, 251 insertions(+), 570 deletions(-) -- 2.35.3

3 33

[PATCH v2] ASoC: lower "no backend DAIs enabled for ... Port" log severity
by Hans de Goede 07 Aug '23

07 Aug '23

If SNDRV_PCM_IOCTL_PREPARE is called when the mixer settings linking frontend and backend have not been setup yet this results in e.g. the following errors getting logged: [ 43.244549] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 43.244744] Baytrail Audio Port: ASoC: error at dpcm_fe_dai_prepare on Baytrail Audio Port: -22 pipewire triggers this leading to 96 lines getting logged after the user has logged into a GNOME session. Change the actual "no backend DAIs enabled for ... Port" error to dev_err_once() to avoid it getting repeated 48 times. While at it also improve the error by hinting the user how to fix this. To not make developing new UCM profiles harder, also log the error at dev_dbg() level all the time (vs once). So that e.g. dyndbg can be used to (re)enable the messages. Also changes _soc_pcm_ret() to not log for -EINVAL errors, to fix the other error getting logged 48 times. Userspace passing wrong parameters should not lead to dmesg messages. Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3407 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Changes in v2: - Switch the logging to a combination of dev_err_once() with improved log-message + dev_dbg() with shorter log message --- sound/soc/soc-pcm.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8896227e4fb7..3aa6b988cb4b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -38,6 +38,7 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, switch (ret) { case -EPROBE_DEFER: case -ENOTSUPP: + case -EINVAL: break; default: dev_err(rtd->dev, @@ -2466,8 +2467,11 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) /* there is no point preparing this FE if there are no BEs */ if (list_empty(&fe->dpcm[stream].be_clients)) { - dev_err(fe->dev, "ASoC: no backend DAIs enabled for %s\n", - fe->dai_link->name); + /* dev_err_once() for visibility, dev_dbg() for debugging UCM profiles */ + dev_err_once(fe->dev, "ASoC: no backend DAIs enabled for %s, possibly missing ALSA mixer-based routing or UCM profile\n", + fe->dai_link->name); + dev_dbg(fe->dev, "ASoC: no backend DAIs enabled for %s\n", + fe->dai_link->name); ret = -EINVAL; goto out; } -- 2.41.0

3 2

[PATCH] sound: core: fix device ownership model in card and pcm
by cujomalainey＠chromium.org 07 Aug '23

07 Aug '23

From: Curtis Malainey <cujomalainey(a)chromium.org> The current implementation of how devices are released is valid for production use cases (root control of memory is handled by card_dev, all other devices are no-ops). This model does not work though in a kernel hacking environment where KASAN and delayed release on kobj is enabled. If the card_dev device is released before any of the child objects a use-after-free bug is caught by KASAN as the delayed release still has a reference to the devices that were freed by the card_dev release. Also both snd_card and snd_pcm both own two devices internally, so even if they released independently, the shared struct would result in another use after free. Solution is to move the child devices into their own memory so they can be handled independently and released on their own schedule. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Doug Anderson <dianders(a)chromium.org> --- include/sound/core.h | 2 +- include/sound/pcm.h | 2 +- sound/core/control.c | 21 +++++++++++++++------ sound/core/control_led.c | 4 ++-- sound/core/pcm.c | 30 ++++++++++++++++++++---------- sound/usb/media.c | 4 ++-- 6 files changed, 41 insertions(+), 22 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index f6e0dd648b80c..a08ab8c8cfb6d 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -96,7 +96,7 @@ struct snd_card { private data */ struct list_head devices; /* devices */ - struct device ctl_dev; /* control device */ + struct device *ctl_dev; /* control device */ unsigned int last_numid; /* last used numeric ID */ struct rw_semaphore controls_rwsem; /* controls lock (list and values) */ rwlock_t ctl_files_rwlock; /* ctl_files list lock */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 19f564606ac42..0243a13e9ac47 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -510,7 +510,7 @@ struct snd_pcm_str { #endif #endif struct snd_kcontrol *chmap_kctl; /* channel-mapping controls */ - struct device dev; + struct device *dev; }; struct snd_pcm { diff --git a/sound/core/control.c b/sound/core/control.c index 8386b53acdcd4..8bbaf3dffce62 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -2315,7 +2315,7 @@ static int snd_ctl_dev_register(struct snd_device *device) int err; err = snd_register_device(SNDRV_DEVICE_TYPE_CONTROL, card, -1, - &snd_ctl_f_ops, card, &card->ctl_dev); + &snd_ctl_f_ops, card, card->ctl_dev); if (err < 0) return err; down_read(&card->controls_rwsem); @@ -2351,7 +2351,7 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) up_read(&snd_ctl_layer_rwsem); up_read(&card->controls_rwsem); - return snd_unregister_device(&card->ctl_dev); + return snd_unregister_device(card->ctl_dev); } /* @@ -2373,10 +2373,15 @@ static int snd_ctl_dev_free(struct snd_device *device) xa_destroy(&card->ctl_hash); #endif up_write(&card->controls_rwsem); - put_device(&card->ctl_dev); + put_device(card->ctl_dev); return 0; } +static void snd_ctl_dev_release(struct device *dev) +{ + kfree(dev); +} + /* * create control core: * called from init.c @@ -2394,13 +2399,17 @@ int snd_ctl_create(struct snd_card *card) return -ENXIO; if (snd_BUG_ON(card->number < 0 || card->number >= SNDRV_CARDS)) return -ENXIO; + card->ctl_dev = kzalloc(sizeof(*card->ctl_dev), GFP_KERNEL); + if (!card->ctl_dev) + return -ENOMEM; - snd_device_initialize(&card->ctl_dev, card); - dev_set_name(&card->ctl_dev, "controlC%d", card->number); + snd_device_initialize(card->ctl_dev, card); + card->ctl_dev->release = snd_ctl_dev_release; + dev_set_name(card->ctl_dev, "controlC%d", card->number); err = snd_device_new(card, SNDRV_DEV_CONTROL, card, &ops); if (err < 0) - put_device(&card->ctl_dev); + put_device(card->ctl_dev); return err; } diff --git a/sound/core/control_led.c b/sound/core/control_led.c index ee77547bf8dcb..760e46cf25cc8 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -688,7 +688,7 @@ static void snd_ctl_led_sysfs_add(struct snd_card *card) goto cerr; led->cards[card->number] = led_card; snprintf(link_name, sizeof(link_name), "led-%s", led->name); - WARN(sysfs_create_link(&card->ctl_dev.kobj, &led_card->dev.kobj, link_name), + WARN(sysfs_create_link(&card->ctl_dev->kobj, &led_card->dev.kobj, link_name), "can't create symlink to controlC%i device\n", card->number); WARN(sysfs_create_link(&led_card->dev.kobj, &card->card_dev.kobj, "card"), "can't create symlink to card%i\n", card->number); @@ -714,7 +714,7 @@ static void snd_ctl_led_sysfs_remove(struct snd_card *card) if (!led_card) continue; snprintf(link_name, sizeof(link_name), "led-%s", led->name); - sysfs_remove_link(&card->ctl_dev.kobj, link_name); + sysfs_remove_link(&card->ctl_dev->kobj, link_name); sysfs_remove_link(&led_card->dev.kobj, "card"); device_unregister(&led_card->dev); led->cards[card->number] = NULL; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 9d95e37311230..9026ccc56dbe7 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -604,7 +604,7 @@ static const struct attribute_group *pcm_dev_attr_groups[]; #ifdef CONFIG_PM_SLEEP static int do_pcm_suspend(struct device *dev) { - struct snd_pcm_str *pstr = container_of(dev, struct snd_pcm_str, dev); + struct snd_pcm_str *pstr = dev_get_drvdata(dev); if (!pstr->pcm->no_device_suspend) snd_pcm_suspend_all(pstr->pcm); @@ -622,6 +622,11 @@ static const struct device_type pcm_dev_type = { .pm = &pcm_dev_pm_ops, }; +static void snd_pcm_dev_release(struct device *dev) +{ + kfree(dev); +} + /** * snd_pcm_new_stream - create a new PCM stream * @pcm: the pcm instance @@ -641,6 +646,10 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) struct snd_pcm_str *pstr = &pcm->streams[stream]; struct snd_pcm_substream *substream, *prev; + pstr->dev = kzalloc(sizeof(*pstr->dev), GFP_KERNEL); + if (!pstr->dev) + return -ENOMEM; + dev_set_drvdata(pstr->dev, pstr); #if IS_ENABLED(CONFIG_SND_PCM_OSS) mutex_init(&pstr->oss.setup_mutex); #endif @@ -650,10 +659,11 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) if (!substream_count) return 0; - snd_device_initialize(&pstr->dev, pcm->card); - pstr->dev.groups = pcm_dev_attr_groups; - pstr->dev.type = &pcm_dev_type; - dev_set_name(&pstr->dev, "pcmC%iD%i%c", pcm->card->number, pcm->device, + snd_device_initialize(pstr->dev, pcm->card); + pstr->dev->release = snd_pcm_dev_release; + pstr->dev->groups = pcm_dev_attr_groups; + pstr->dev->type = &pcm_dev_type; + dev_set_name(pstr->dev, "pcmC%iD%i%c", pcm->card->number, pcm->device, stream == SNDRV_PCM_STREAM_PLAYBACK ? 'p' : 'c'); if (!pcm->internal) { @@ -699,7 +709,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) prev = substream; } return 0; -} +} EXPORT_SYMBOL(snd_pcm_new_stream); static int _snd_pcm_new(struct snd_card *card, const char *id, int device, @@ -847,7 +857,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) #endif free_chmap(pstr); if (pstr->substream_count) - put_device(&pstr->dev); + put_device(pstr->dev); } #if IS_ENABLED(CONFIG_SND_PCM_OSS) @@ -1017,7 +1027,7 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) static ssize_t pcm_class_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_pcm_str *pstr = container_of(dev, struct snd_pcm_str, dev); + struct snd_pcm_str *pstr = dev_get_drvdata(dev); struct snd_pcm *pcm = pstr->pcm; const char *str; static const char *strs[SNDRV_PCM_CLASS_LAST + 1] = { @@ -1078,7 +1088,7 @@ static int snd_pcm_dev_register(struct snd_device *device) /* register pcm */ err = snd_register_device(devtype, pcm->card, pcm->device, &snd_pcm_f_ops[cidx], pcm, - &pcm->streams[cidx].dev); + pcm->streams[cidx].dev); if (err < 0) { list_del_init(&pcm->list); goto unlock; @@ -1125,7 +1135,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) pcm_call_notify(pcm, n_disconnect); for (cidx = 0; cidx < 2; cidx++) { - snd_unregister_device(&pcm->streams[cidx].dev); + snd_unregister_device(pcm->streams[cidx].dev); free_chmap(&pcm->streams[cidx]); } mutex_unlock(&pcm->open_mutex); diff --git a/sound/usb/media.c b/sound/usb/media.c index 840f42cb9272c..d48db6f3ae659 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -35,7 +35,7 @@ int snd_media_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, { struct media_device *mdev; struct media_ctl *mctl; - struct device *pcm_dev = &pcm->streams[stream].dev; + struct device *pcm_dev = pcm->streams[stream].dev; u32 intf_type; int ret = 0; u16 mixer_pad; @@ -163,7 +163,7 @@ void snd_media_stop_pipeline(struct snd_usb_substream *subs) static int snd_media_mixer_init(struct snd_usb_audio *chip) { - struct device *ctl_dev = &chip->card->ctl_dev; + struct device *ctl_dev = chip->card->ctl_dev; struct media_intf_devnode *ctl_intf; struct usb_mixer_interface *mixer; struct media_device *mdev = chip->media_dev; -- 2.41.0.585.gd2178a4bd4-goog

6 16

[PATCH v1 0/5] Add I2S support for the StarFive JH7110 SoC
by Xingyu Wu 07 Aug '23

07 Aug '23

This patch series adds I2S support for the StarFive JH7110 RISC-V SoC based on Designware I2S controller. There has three I2S channels (RX/TX0/TX1) on the JH7110 SoC, one of which is for record(RX) and two for playback(TX). The first patch adds the ops to get data from platform bus in the I2S driver. The second patch adds support for the StarFive JH7110 SoC in the Designware I2S bindings. The third patch adds support for the StarFive JH7110 SoC in the Designware I2S driver. The fourth patch fixes the name of I2STX1 pinmux. The last patch adds device node of I2S RX/TX0/TX1 in JH7110 dts. This patch series is based on Linux-next which is merge clock, syscon and dma nodes for the StarFive JH7110 SoC. And these should be applied after the following patchset: https://lore.kernel.org/all/20230724055440.100947-1-xingyu.wu@starfivetech.… The series has been tested and works normally on the VisionFive 2 board by plugging an audio expansion board. Xingyu Wu (5): ASoC: dwc: Use ops to get platform data ASoC: dt-bindings: snps,designware-i2s: Add StarFive JH7110 SoC support ASoC: dwc: i2s: Add StarFive JH7110 SoC support riscv: dts: starfive: pinfunc: Fix the pins name of I2STX1 riscv: dts: starfive: Add the nodes and pins of I2Srx/I2Stx0/I2Stx1 .../bindings/sound/snps,designware-i2s.yaml | 101 +++++- arch/riscv/boot/dts/starfive/jh7110-pinfunc.h | 4 +- .../jh7110-starfive-visionfive-2.dtsi | 58 ++++ arch/riscv/boot/dts/starfive/jh7110.dtsi | 65 ++++ include/sound/designware_i2s.h | 3 + sound/soc/dwc/dwc-i2s.c | 318 ++++++++++++++++-- sound/soc/dwc/local.h | 1 + 7 files changed, 523 insertions(+), 27 deletions(-) base-commit: 0fee53365d051781156b75c6f6e6015757e71fd8 prerequisite-patch-id: 0159f09bb0a1ff711a00ae17ef5b12662c9c7d3d -- 2.25.1

3 13

[PATCH v2 0/3] ASoC: mt8188-mt6359: add SOF support
by Trevor Wu 07 Aug '23

07 Aug '23

This series introduces dynamic pinctrl and adds support for the SOF on the mt8188-mt6359 machine driver. Changes since v1: - minor changes to incorporate the suggestions from the reviewer. Trevor Wu (3): ASoC: mediatek: mt8188-mt6359: support dynamic pinctrl ASoC: mediatek: common: revise SOF common code ASoC: mediatek: mt8188-mt6359: add SOF support .../soc/mediatek/common/mtk-dsp-sof-common.c | 106 ++++++-- .../soc/mediatek/common/mtk-dsp-sof-common.h | 8 + sound/soc/mediatek/mt8188/mt8188-mt6359.c | 239 +++++++++++++++++- 3 files changed, 325 insertions(+), 28 deletions(-) -- 2.18.0

3 6

[PATCH v5] um/drivers: fix hostaudio build errors
by Randy Dunlap 06 Aug '23

06 Aug '23

Use "select" to ensure that the required kconfig symbols are set as expected. Drop HOSTAUDIO since it is now equivalent to UML_SOUND. Set CONFIG_SOUND=m in ARCH=um defconfig files to maintain the status quo of the default configs. Allow SOUND with UML regardless of HAS_IOMEM. Otherwise there is a kconfig warning for unmet dependencies. (This was not an issue when SOUND was defined in arch/um/drivers/Kconfig. I have done 50 randconfig builds and didn't find any issues.) This fixes build errors when CONFIG_SOUND is not set: ld: arch/um/drivers/hostaudio_kern.o: in function `hostaudio_cleanup_module': hostaudio_kern.c:(.exit.text+0xa): undefined reference to `unregister_sound_mixer' ld: hostaudio_kern.c:(.exit.text+0x15): undefined reference to `unregister_sound_dsp' ld: arch/um/drivers/hostaudio_kern.o: in function `hostaudio_init_module': hostaudio_kern.c:(.init.text+0x19): undefined reference to `register_sound_dsp' ld: hostaudio_kern.c:(.init.text+0x31): undefined reference to `register_sound_mixer' ld: hostaudio_kern.c:(.init.text+0x49): undefined reference to `unregister_sound_dsp' and this kconfig warning: WARNING: unmet direct dependencies detected for SOUND Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Fixes: d886e87cb82b ("sound: make OSS sound core optional") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Closes: lore.kernel.org/r/202307141416.vxuRVpFv-lkp@intel.com Cc: Richard Weinberger <richard(a)nod.at> Cc: Anton Ivanov <anton.ivanov(a)cambridgegreys.com> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: linux-um(a)lists.infradead.org Cc: Tejun Heo <tj(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Masahiro Yamada <masahiroy(a)kernel.org> Cc: Nathan Chancellor <nathan(a)kernel.org> Cc: Nick Desaulniers <ndesaulniers(a)google.com> Cc: Nicolas Schier <nicolas(a)fjasle.eu> Cc: linux-kbuild(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- v2: don't delete the HOSTAUDIO Kconfig entry (Masahiro) v3: drop HOSTAUDIO and use CONFIG_UML_SOUND for it in Makefile (Takashi); add SOUND depends on "|| UML" to HAS_IOMEM v4: use depends on instead of select for SOUND (Masahiro); use Closes: instead of Link: v5: update ARCH=um defconfig files (Masahiro) arch/um/configs/i386_defconfig | 1 + arch/um/configs/x86_64_defconfig | 1 + arch/um/drivers/Kconfig | 16 +++------------- arch/um/drivers/Makefile | 2 +- sound/Kconfig | 2 +- 5 files changed, 7 insertions(+), 15 deletions(-) diff -- a/arch/um/drivers/Kconfig b/arch/um/drivers/Kconfig --- a/arch/um/drivers/Kconfig +++ b/arch/um/drivers/Kconfig @@ -111,24 +111,14 @@ config SSL_CHAN config UML_SOUND tristate "Sound support" + depends on SOUND + select SOUND_OSS_CORE help This option enables UML sound support. If enabled, it will pull in - soundcore and the UML hostaudio relay, which acts as a intermediary + the UML hostaudio relay, which acts as a intermediary between the host's dsp and mixer devices and the UML sound system. It is safe to say 'Y' here. -config SOUND - tristate - default UML_SOUND - -config SOUND_OSS_CORE - bool - default UML_SOUND - -config HOSTAUDIO - tristate - default UML_SOUND - endmenu menu "UML Network Devices" diff -- a/sound/Kconfig b/sound/Kconfig --- a/sound/Kconfig +++ b/sound/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only menuconfig SOUND tristate "Sound card support" - depends on HAS_IOMEM + depends on HAS_IOMEM || UML help If you have a sound card in your computer, i.e. if it can say more than an occasional beep, say Y. diff -- a/arch/um/drivers/Makefile b/arch/um/drivers/Makefile --- a/arch/um/drivers/Makefile +++ b/arch/um/drivers/Makefile @@ -54,7 +54,7 @@ obj-$(CONFIG_UML_NET) += net.o obj-$(CONFIG_MCONSOLE) += mconsole.o obj-$(CONFIG_MMAPPER) += mmapper_kern.o obj-$(CONFIG_BLK_DEV_UBD) += ubd.o -obj-$(CONFIG_HOSTAUDIO) += hostaudio.o +obj-$(CONFIG_UML_SOUND) += hostaudio.o obj-$(CONFIG_NULL_CHAN) += null.o obj-$(CONFIG_PORT_CHAN) += port.o obj-$(CONFIG_PTY_CHAN) += pty.o diff -- a/arch/um/configs/i386_defconfig b/arch/um/configs/i386_defconfig --- a/arch/um/configs/i386_defconfig +++ b/arch/um/configs/i386_defconfig @@ -34,6 +34,7 @@ CONFIG_TTY_CHAN=y CONFIG_XTERM_CHAN=y CONFIG_CON_CHAN="pts" CONFIG_SSL_CHAN="pts" +CONFIG_SOUND=m CONFIG_UML_SOUND=m CONFIG_DEVTMPFS=y CONFIG_DEVTMPFS_MOUNT=y diff -- a/arch/um/configs/x86_64_defconfig b/arch/um/configs/x86_64_defconfig --- a/arch/um/configs/x86_64_defconfig +++ b/arch/um/configs/x86_64_defconfig @@ -32,6 +32,7 @@ CONFIG_TTY_CHAN=y CONFIG_XTERM_CHAN=y CONFIG_CON_CHAN="pts" CONFIG_SSL_CHAN="pts" +CONFIG_SOUND=m CONFIG_UML_SOUND=m CONFIG_DEVTMPFS=y CONFIG_DEVTMPFS_MOUNT=y

2 1

change order in which kernel initializes devices
by Fourhundred Thecat 06 Aug '23

06 Aug '23

Hello, I am using monolithic kernel without loadable module support. Everything is compiled in statically. Now I have snd-aloop (CONFIG_SND_ALOOP=y) and when i boot my kernel, it is recognized as primary device. My real hw card is as second device. Is there any way I can change the order, in which kernel initializes the sound devices, so that my real (hw) sound card is recognized first ? If I was using modules, I could simply load snd-aloop later. I am compiling my own kernel, so I would be happy to patch the code if necessary, if somebody could point me into the right section. Here is output of aplay -l, in case it helps: **** List of PLAYBACK Hardware Devices **** card 0: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 1: PCH [HDA Intel PCH], device 0: Generic Analog [Generic Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 PS: why are there 2 loopback devices, with 8 subdevices? When I am alredy at it, can I change tat in the kernel, so that I only have 1 loopback device for simplicity ? thank you,

1 0

[PATCH 1/2] ASoC: dt-bindings: wlf,wm8960: Describe the power supplies
by Fabio Estevam 06 Aug '23

06 Aug '23

From: Fabio Estevam <festevam(a)denx.de> WM8960 has the following power supplies: - AVDD - DBVDD - DCVDD - SPKVDD1 - SPKVDD1 Add them to bindings to improve the hardware description. Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- .../devicetree/bindings/sound/wlf,wm8960.yaml | 20 +++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml index ee8eba7f0104..82519f6720ef 100644 --- a/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml +++ b/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml @@ -26,6 +26,21 @@ properties: '#sound-dai-cells': const: 0 + AVDD-supply: + description: Analogue supply. + + DBVDD-supply: + description: Digital Buffer Supply. + + DCVDD-supply: + description: Digital Core Supply. + + SPKVDD1-supply: + description: Supply for speaker drivers 1. + + SPKVDD2-supply: + description: Supply for speaker drivers 2. + wlf,capless: type: boolean description: @@ -84,5 +99,10 @@ examples: wlf,hp-cfg = <3 2 3>; wlf,gpio-cfg = <1 3>; wlf,shared-lrclk; + DCVDD-supply = <&reg_audio>; + DBVDD-supply = <&reg_audio>; + AVDD-supply = <&reg_audio>; + SPKVDD1-supply = <&reg_audio>; + SPKVDD2-supply = <&reg_audio>; }; }; -- 2.34.1

5 6

[PATCH] ASoC: max98363: don't return on success reading revision ID
by Bard Liao 04 Aug '23

04 Aug '23

max98363_io_init needs to keep going when we read revision ID successfully. Fixes: 18c0af945fa3 ("ASoC: max98363: add soundwire amplifier driver") Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/codecs/max98363.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98363.c b/sound/soc/codecs/max98363.c index b5c69bba0e48..2dfaf4fcfbd3 100644 --- a/sound/soc/codecs/max98363.c +++ b/sound/soc/codecs/max98363.c @@ -185,10 +185,10 @@ static int max98363_io_init(struct sdw_slave *slave) pm_runtime_get_noresume(dev); ret = regmap_read(max98363->regmap, MAX98363_R21FF_REV_ID, &reg); - if (!ret) { + if (!ret) dev_info(dev, "Revision ID: %X\n", reg); - return ret; - } + else + goto out; if (max98363->first_hw_init) { regcache_cache_bypass(max98363->regmap, false); @@ -198,10 +198,11 @@ static int max98363_io_init(struct sdw_slave *slave) max98363->first_hw_init = true; max98363->hw_init = true; +out: pm_runtime_mark_last_busy(dev); pm_runtime_put_autosuspend(dev); - return 0; + return ret; } #define MAX98363_RATES SNDRV_PCM_RATE_8000_192000 -- 2.25.1

2 1