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

[PATCH V4 0/3] ASoC: codecs: Add awinic AW88261 audio amplifier driver
by wangweidong.a＠awinic.com 07 Aug '23

07 Aug '23

From: Weidong Wang <wangweidong.a(a)awinic.com> The awinic AW88261 is an I2S/TDM input, high efficiency digital Smart K audio amplifier v3 -> v4: Modify the dev_err_probe usage Changed the use of sizeof in kzalloc Changed the function name Merge aw88261_device.c into aw88261.c Weidong Wang (3): ASoC: dt-bindings: Add schema for "awinic,aw88261" ASoC: codecs: Add code for bin parsing compatible with aw88261 ASoC: codecs: Add aw88261 amplifier driver .../bindings/sound/awinic,aw88395.yaml | 4 +- sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/aw88261.c | 1298 +++++++++++++++++ sound/soc/codecs/aw88261.h | 459 ++++++ sound/soc/codecs/aw88395/aw88395_lib.c | 193 ++- sound/soc/codecs/aw88395/aw88395_reg.h | 1 + 7 files changed, 1954 insertions(+), 18 deletions(-) create mode 100644 sound/soc/codecs/aw88261.c create mode 100644 sound/soc/codecs/aw88261.h base-commit: c1a515d3c0270628df8ae5f5118ba859b85464a2 -- 2.41.0

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[PATCH] ASoC: jz4740: Update to use maple tree register cache
by Mark Brown 07 Aug '23

07 Aug '23

The maple tree register cache is based on a much more modern data structure than the rbtree cache and makes optimisation choices which are probably more appropriate for modern systems than those made by the rbtree cache. In v6.5 it has also acquired the ability to generate multi-register writes in sync operations, bringing performance up to parity with the rbtree cache there. Update the jz4740 driver to use the more modern data structure. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/jz4740.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 7c25acf6ff0d..d1cea93bdb59 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -301,7 +301,7 @@ static const struct regmap_config jz4740_codec_regmap_config = { .reg_defaults = jz4740_codec_reg_defaults, .num_reg_defaults = ARRAY_SIZE(jz4740_codec_reg_defaults), - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; static int jz4740_codec_probe(struct platform_device *pdev) --- base-commit: 06c2afb862f9da8dc5efa4b6076a0e48c3fbaaa5 change-id: 20230630-asoc-jz-maple-aacbd3a12bb6 Best regards, -- Mark Brown <broonie(a)kernel.org>

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[PATCH 00/16] ASoC: SoundWire codecs: improve pm_runtime handling
by Pierre-Louis Bossart 07 Aug '23

07 Aug '23

This patchset improves the pm_runtime behavior in rare corner cases identified by the Intel CI in the last 6 months. a) in stress-tests, it's not uncommon to see the following type of warnings when the codec reports as ATTACHED "rt711 sdw:0:025d:0711:00: runtime PM trying to activate child device sdw:0:025d:0711:00 but parent (sdw-master-0) is not active" This warning was not correlated with any functional issue, but it exposed a design issue on when to enable pm_runtime. The recommended practice in the pm_runtime documentation is to keep the devices in 'suspended' mode and mark them as 'active' when they are really functional. b) enabling pm_runtime when the codec reports as ATTACHED also creates a problematic case when the ASoC pm_runtime_get_sync() will silently fail due to the -EACCESS error handling. This can happen when playback starts before the codec is enumerated. This patchset modifies the initial stages so that codecs are pm_runtime enabled in the .probe() callback, but become pm_runtime 'active' only when they report present. This is better aligned with the design of the pm_runtime helpers and improved CI results significantly. This patchset modifies all existing SoundWire codecs (except Qualcomm ones), but the pattern of changes is exactly the same in all patches. Pierre-Louis Bossart (16): ASoC: SoundWire codecs: return error status in probe ASoC: SoundWire codecs: make regmap cache-only in probe ASoC: rt5682-sdw: make regmap cache-only in probe ASoC: rt711: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt711-sdca: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt700: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1712-sdca: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt712-sdca-dmic: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt715: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt715-sdca: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1308-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1316-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1318-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt5682-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: max98363: enable pm_runtime in probe, keep status as 'suspended' ASoC: max98373-sdw: enable pm_runtime in probe, keep status as 'suspended' sound/soc/codecs/max98363.c | 41 +++++++++++++--------- sound/soc/codecs/max98373-sdw.c | 47 ++++++++++++++----------- sound/soc/codecs/rt1308-sdw.c | 54 +++++++++++++++------------- sound/soc/codecs/rt1316-sdw.c | 44 ++++++++++++++--------- sound/soc/codecs/rt1318-sdw.c | 43 ++++++++++++++--------- sound/soc/codecs/rt5682-sdw.c | 56 ++++++++++++++++++------------ sound/soc/codecs/rt5682.c | 3 ++ sound/soc/codecs/rt700-sdw.c | 7 ++-- sound/soc/codecs/rt700.c | 46 +++++++++++++++--------- sound/soc/codecs/rt711-sdca-sdw.c | 3 +- sound/soc/codecs/rt711-sdca.c | 48 +++++++++++++++++-------- sound/soc/codecs/rt711-sdw.c | 7 ++-- sound/soc/codecs/rt711.c | 47 ++++++++++++++++--------- sound/soc/codecs/rt712-sdca-dmic.c | 47 +++++++++++++++---------- sound/soc/codecs/rt712-sdca-sdw.c | 3 +- sound/soc/codecs/rt712-sdca.c | 45 ++++++++++++++++-------- sound/soc/codecs/rt715-sdca-sdw.c | 5 +-- sound/soc/codecs/rt715-sdca.c | 41 ++++++++++++++++------ sound/soc/codecs/rt715-sdw.c | 9 ++--- sound/soc/codecs/rt715.c | 41 +++++++++++++++------- 20 files changed, 392 insertions(+), 245 deletions(-) -- 2.39.2

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[PATCH v2 1/3] ASoC: dt-bindings: Convert maxim,max98925 to DT schema
by Krzysztof Kozlowski 07 Aug '23

07 Aug '23

Convert the Maxim Integrated MAX98925/MAX98926/MAX98927 speaker amplifier bindings to DT schema format. Changes during conversion: 1. Add "sound-dai-cells", already used by DTS. 2. Use "maxim,interleave-mode" instead previous "interleave-mode" and undocumented but used interleave_mode. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v2: 1. Use maxim,interleave-mode as bool. Ryan, As original author, I added you as bindings maintainer. Is that okay or maybe someone else from Maxim should take care about the bindings? --- .../devicetree/bindings/sound/max9892x.txt | 44 --------- .../bindings/sound/maxim,max98925.yaml | 98 +++++++++++++++++++ 2 files changed, 98 insertions(+), 44 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/max9892x.txt create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98925.yaml diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt deleted file mode 100644 index 98cb9ba5b328..000000000000 --- a/Documentation/devicetree/bindings/sound/max9892x.txt +++ /dev/null @@ -1,44 +0,0 @@ -Maxim Integrated MAX98925/MAX98926/MAX98927 Speaker Amplifier - -This device supports I2C. - -Required properties: - - - compatible : should be one of the following - - "maxim,max98925" - - "maxim,max98926" - - "maxim,max98927" - - - vmon-slot-no : slot number used to send voltage information - or in inteleave mode this will be used as - interleave slot. - MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 - MAX98927 slot range : 0 ~ 15, Default : 0 - - - imon-slot-no : slot number used to send current information - MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 - MAX98927 slot range : 0 ~ 15, Default : 0 - - - interleave-mode : When using two MAX9892X in a system it is - possible to create ADC data that that will - overflow the frame size. Digital Audio Interleave - mode provides a means to output VMON and IMON data - from two devices on a single DOUT line when running - smaller frames sizes such as 32 BCLKS per LRCLK or - 48 BCLKS per LRCLK. - Range : 0 (off), 1 (on), Default : 0 - - - reg : the I2C address of the device for I2C - -Optional properties: - - reset-gpios : GPIO to reset the device - -Example: - -codec: max98927@3a { - compatible = "maxim,max98927"; - vmon-slot-no = <0>; - imon-slot-no = <1>; - interleave-mode = <0>; - reg = <0x3a>; -}; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98925.yaml b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml new file mode 100644 index 000000000000..32fd86204a7a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98925.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98925/MAX98926/MAX98927 speaker amplifier + +maintainers: + - Ryan Lee <ryans.lee(a)maximintegrated.com> + +properties: + compatible: + enum: + - maxim,max98925 + - maxim,max98926 + - maxim,max98927 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send voltage information or in inteleave mode this + will be used as interleave slot. + + imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send current information. + + maxim,interleave-mode: + type: boolean + description: + When using two MAX9892X in a system it is possible to create ADC data + that will overflow the frame size. When enabled, the Digital Audio + Interleave mode provides a means to output VMON and IMON data from two + devices on a single DOUT line when running smaller frames sizes such as + 32 BCLKS per LRCLK or 48 BCLKS per LRCLK. + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - maxim,max98927 + then: + properties: + vmon-slot-no: + minimum: 0 + maximum: 15 + + imon-slot-no: + minimum: 0 + maximum: 15 + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + #include <dt-bindings/gpio/gpio.h> + audio-codec@3a { + compatible = "maxim,max98927"; + reg = <0x3a>; + #sound-dai-cells = <0>; + + pinctrl-0 = <&speaker_default>; + pinctrl-names = "default"; + + reset-gpios = <&tlmm 69 GPIO_ACTIVE_LOW>; + + vmon-slot-no = <1>; + imon-slot-no = <0>; + }; + }; -- 2.34.1

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[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

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[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

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[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

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[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

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[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

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[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

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