October 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH v5 00/16] ASoC: Add Audio Graph Card2 support
by Kuninori Morimoto 15 Oct '21

15 Oct '21

Hi Mark We already have Audio-Graph-Card which is Of-Graph base general sound card driver. Basically it supports basic CPU-Codec connection, and is also supporting DPCM connection. Because it was forcibly expanded to DPCM, DT parsing is very limited and very difficult to add new features on it, for example Multi-CPU/Codec support, Codec2Codec support, etc. This patch adds more flexible new Audio-Graph-Card2 driver for it. Audio-Graph-Card and Audio-Graph-Card2 are similar, but don't have full compatibility. The reason why I need Audio-Graph-Card2 instead of updating Audio-Graph-Card is that it is very difficult to keep compatibility. Audio-Graph-Card2 supports Normal/DPCM/Codec2Codec Connection wich Single/Multi DAIs. And it is possible to Customizing. This patch-set adds Audio-Graph-Card2 driver and its custom driver sample, and DT settings sample which can be used for testing. To enable testing/debuging, this patch-set also adds Test-Component driver. We already have Dummy Component and/or Dummy DAI on soc-utils, but 1) we can't use it from DT, 2) it do nothing. Added new Test-Component can be used from DT, and it can indicate called function name. We can use it to trace callback order, understanding ALSA SoC behavior, etc, etc... Sample DT settings of Audio Graph Card2 is using Test-Component as CPU/Codec DAI. You can easily try to use/test it if you added below line to your DT file. Your .config needs to have below CONFIGs to use/test it. It will probe sample Sound Card which has Normal/DPCM/Multi/Codec2Codec connections. #include "../../../../../sound/soc/generic/audio-graph-card2-custom-sample.dtsi" CONFIG_SND_AUDIO_GRAPH_CARD2 CONFIG_SND_AUDIO_GRAPH_CARD2_CUSTOM_SAMPLE CONFIG_SND_TEST_COMPONENT Because Audio Graph Card2 is still under experimental stage, it will indicate such warning when probing, and the DT might be updated/exchanged. It can use Codec2Codec, but it will start automatically when probed, and can't stop it so far. It should be updated. Link: https://lore.kernel.org/r/87k0xszlep.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871r8u4s6q.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87a6mhwyqn.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87tuitusy4.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87a6jn56x0.wl-kuninori.morimoto.gx@renesas.com v1 -> v2 - don't use "port" base for_each loop v2 -> v3 - Rename audio-graph-card2 to rich-graph-card - Rename DSP to DPCM not to confuse - Normal/DPCM/Codec2Codec can use Single/Multi DAIs. - use dpcm/multi/codec2codec node instead of using extra compatible - Sample DTSI patch is separated to Single/Multi. v3 -> v4 - Rename rich-graph-card to audio-graph-card2 - fixup custom sample driver's connection bug - test-component compatible uses "verbose" instead of "vv" v4 -> v5 - tidyup git-log comment at - tidyup Custom Sample comment Kuninori Morimoto (16): ASoC: test-component: add Test Component YAML bindings ASoC: test-component: add Test Component for Sound debug/test ASoC: simple-card-utils: add asoc_graph_is_ports0() ASoC: simple-card-utils: add codec2codec support ASoC: add Audio Graph Card2 driver ASoC: audio-graph-card2: add Multi CPU/Codec support ASoC: audio-graph-card2: add DPCM support ASoC: audio-graph-card2: add Codec2Codec support ASoC: add Audio Graph Card2 Yaml Document ASoC: add Audio Graph Card2 Custom Sample ASoC: audio-graph-card2-custom-sample.dtsi: add Sample DT for Normal (Single) ASoC: audio-graph-card2-custom-sample.dtsi: add Sample DT for Normal (Nulti) ASoC: audio-graph-card2-custom-sample.dtsi: add DPCM sample (Single) ASoC: audio-graph-card2-custom-sample.dtsi: add DPCM sample (Multi) ASoC: audio-graph-card2-custom-sample.dtsi: add Codec2Codec sample (Single) ASoC: audio-graph-card2-custom-sample.dtsi: add Codec2Codec sample (Multi) .../bindings/sound/audio-graph-card2.yaml | 57 + .../bindings/sound/test-component.yaml | 33 + include/sound/graph_card.h | 21 + include/sound/simple_card_utils.h | 4 + sound/soc/generic/Kconfig | 20 + sound/soc/generic/Makefile | 6 + .../generic/audio-graph-card2-custom-sample.c | 183 +++ .../audio-graph-card2-custom-sample.dtsi | 227 +++ sound/soc/generic/audio-graph-card2.c | 1281 +++++++++++++++++ sound/soc/generic/simple-card-utils.c | 46 +- sound/soc/generic/test-component.c | 659 +++++++++ 11 files changed, 2536 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card2.yaml create mode 100644 Documentation/devicetree/bindings/sound/test-component.yaml create mode 100644 sound/soc/generic/audio-graph-card2-custom-sample.c create mode 100644 sound/soc/generic/audio-graph-card2-custom-sample.dtsi create mode 100644 sound/soc/generic/audio-graph-card2.c create mode 100644 sound/soc/generic/test-component.c -- 2.25.1

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[PATCH V2 1/3] ASoc: amd: create platform device for VG machine driver
by Vijendar Mukunda 15 Oct '21

15 Oct '21

Create platform device for Vangogh machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- Changes since v1: - changed machine driver platform device name as "acp5x_mach" sound/soc/amd/vangogh/acp5x.h | 2 +- sound/soc/amd/vangogh/pci-acp5x.c | 3 +++ 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/vangogh/acp5x.h b/sound/soc/amd/vangogh/acp5x.h index a808635f9740..fe5e1fa98974 100644 --- a/sound/soc/amd/vangogh/acp5x.h +++ b/sound/soc/amd/vangogh/acp5x.h @@ -23,7 +23,7 @@ #define ACP_ERR_INTR_MASK 0x20000000 #define ACP_EXT_INTR_STAT_CLEAR_MASK 0xFFFFFFFF -#define ACP5x_DEVS 3 +#define ACP5x_DEVS 4 #define ACP5x_REG_START 0x1240000 #define ACP5x_REG_END 0x1250200 #define ACP5x_I2STDM_REG_START 0x1242400 diff --git a/sound/soc/amd/vangogh/pci-acp5x.c b/sound/soc/amd/vangogh/pci-acp5x.c index a57b762d9f2e..2b6b9edc36e2 100644 --- a/sound/soc/amd/vangogh/pci-acp5x.c +++ b/sound/soc/amd/vangogh/pci-acp5x.c @@ -213,6 +213,9 @@ static int snd_acp5x_probe(struct pci_dev *pci, pdevinfo[2].num_res = 1; pdevinfo[2].res = &adata->res[2]; + pdevinfo[3].name = "acp5x_mach"; + pdevinfo[3].id = 0; + pdevinfo[3].parent = &pci->dev; for (i = 0; i < ACP5x_DEVS; i++) { adata->pdev[i] = platform_device_register_full(&pdevinfo[i]); -- 2.25.1

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[PATCH 00/16] ASoC: cs42l42: Collection of bugfixes
by Richard Fitzgerald 15 Oct '21

15 Oct '21

This patch set contains various bugfixes for the cs42l42 codec driver. Patches marked "Fixes" will apply cleanly to the patch that first introduced the bug. Patches NOT marked "Fixes" will not apply cleanly to the point the bug was first introduced and/or the bug is not having sufficient impact to risk churning older code versions. Richard Fitzgerald (16): ASoC: cs42l42: Don't reconfigure the PLL while it is running ASoC: cs42l42: Always configure both ASP TX channels ASoC: cs42l42: Correct some register default values ASoC: cs42l42: Don't set defaults for volatile registers ASoC: cs42l42: Defer probe if request_threaded_irq() returns EPROBE_DEFER ASoC: cs42l42: Reset GPIO is mandatory ASoC: cs42l42: Correct power-up sequence to match datasheet ASoC: cs42l42: Reset and power-down on driver remove() ASoC: cs42l42: Prevent NULL pointer deref in interrupt handler ASoC: cs42l42: Don't claim to support 192k ASoC: cs42l42: Use PLL for SCLK > 12.288MHz ASoC: cs42l42: Allow time for HP/ADC to power-up after enable ASoC: cs42l42: Set correct SRC MCLK ASoC: cs42l42: Mark OSC_SWITCH_STATUS register volatile ASoC: cs42l42: Fix WARN in remove() if running without an interrupt ASoC: cs42l42: Always enable TS_PLUG and TS_UNPLUG interrupts sound/soc/codecs/cs42l42.c | 282 +++++++++++++++++++++++++-------------------- sound/soc/codecs/cs42l42.h | 9 +- 2 files changed, 160 insertions(+), 131 deletions(-) -- 2.11.0

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[PATCH v4 00/16] ASoC: Add Audio Graph Card2 support
by Kuninori Morimoto 15 Oct '21

15 Oct '21

Hi Mark We already have Audio-Graph-Card which is Of-Graph base general sound card driver. Basically it supports basic CPU-Codec connection, and is also supporting DPCM connection. Because it was forcibly expanded to DPCM, DT parsing is very limited and very difficult to add new features on it, for example Multi-CPU/Codec support, Codec2Codec support, etc. This patch adds more flexible new Audio-Graph-Card2 driver for it. Audio-Graph-Card and Audio-Graph-Card2 are similar, but don't have full compatibility. The reason why I need Audio-Graph-Card2 instead of updating Audio-Graph-Card is that it is very difficult to keep compatibility. Audio-Graph-Card2 supports Normal/DPCM/Codec2Codec Connection wich Single/Multi DAIs. And it is possible to Customizing. This patch-set adds Audio-Graph-Card2 driver and its custom driver sample, and DT settings sample which can be used for testing. To enable testing/debuging, this patch-set also adds Test-Component driver. We already have Dummy Component and/or Dummy DAI on soc-utils, but 1) we can't use it from DT, 2) it do nothing. Added new Test-Component can be used from DT, and it can indicate called function name. We can use it to trace callback order, understanding ALSA SoC behavior, etc, etc... Sample DT settings of Audio Graph Card2 is using Test-Component as CPU/Codec DAI. You can easily try to use/test it if you added below line to your DT file. Your .config needs to have below CONFIGs to use/test it. It will probe sample Sound Card which has Normal/DPCM/Multi/Codec2Codec connections. #include "../../../../../sound/soc/generic/audio-graph-card2-custom-sample.dtsi" CONFIG_SND_AUDIO_GRAPH_CARD2 CONFIG_SND_AUDIO_GRAPH_CARD2_CUSTOM_SAMPLE CONFIG_SND_TEST_COMPONENT Because Audio Graph Card2 is still under experimental stage, it will indicate such warning when probing, and the DT might be updated/exchanged. It can use Codec2Codec, but it will start automatically when probed, and can't stop it so far. It should be updated. Link: https://lore.kernel.org/r/87k0xszlep.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871r8u4s6q.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87a6mhwyqn.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87tuitusy4.wl-kuninori.morimoto.gx@renesas.com v1 -> v2 - don't use "port" base for_each loop v2 -> v3 - Rename audio-graph-card2 to rich-graph-card - Rename DSP to DPCM not to confuse - Normal/DPCM/Codec2Codec can use Single/Multi DAIs. - use dpcm/multi/codec2codec node instead of using extra compatible - Sample DTSI patch is separated to Single/Multi. v3 -> v4 - Rename rich-graph-card to audio-graph-card2 - fixup custom sample driver's connection bug - test-component compatible uses "verbose" instead of "vv" Kuninori Morimoto (16): ASoC: test-component: add Test Component YAML bindings ASoC: test-component: add Test Component for Sound debug/test ASoC: simple-card-utils: add asoc_graph_is_ports0() ASoC: simple-card-utils: add codec2codec support ASoC: add Audio Graph Card2 driver ASoC: audio-graph-card2: add Multi CPU/Codec support ASoC: audio-graph-card2: add DPCM support ASoC: audio-graph-card2: add Codec2Codec support ASoC: add Audio Graph Card2 Yaml Document ASoC: add Audio Graph Card2 Custom Sample ASoC: audio-graph-card2-custom-sample.dtsi: add Sample DT for Normal (Single) ASoC: audio-graph-card2-custom-sample.dtsi: add Sample DT for Normal (Nulti) ASoC: audio-graph-card2-custom-sample.dtsi: add DPCM sample (Single) ASoC: audio-graph-card2-custom-sample.dtsi: add DPCM sample (Multi) ASoC: audio-graph-card2-custom-sample.dtsi: add Codec2Codec sample (Single) ASoC: audio-graph-card2-custom-sample.dtsi: add Codec2Codec sample (Multi) .../bindings/sound/audio-graph-card2.yaml | 57 + .../bindings/sound/test-component.yaml | 33 + include/sound/graph_card.h | 21 + include/sound/simple_card_utils.h | 4 + sound/soc/generic/Kconfig | 20 + sound/soc/generic/Makefile | 6 + .../generic/audio-graph-card2-custom-sample.c | 174 +++ .../audio-graph-card2-custom-sample.dtsi | 227 +++ sound/soc/generic/audio-graph-card2.c | 1281 +++++++++++++++++ sound/soc/generic/simple-card-utils.c | 46 +- sound/soc/generic/test-component.c | 659 +++++++++ 11 files changed, 2527 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card2.yaml create mode 100644 Documentation/devicetree/bindings/sound/test-component.yaml create mode 100644 sound/soc/generic/audio-graph-card2-custom-sample.c create mode 100644 sound/soc/generic/audio-graph-card2-custom-sample.dtsi create mode 100644 sound/soc/generic/audio-graph-card2.c create mode 100644 sound/soc/generic/test-component.c -- 2.25.1

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[PATCH] ASoC: soc-component: improve error reporting for register access
by Srinivas Kandagatla 15 Oct '21

15 Oct '21

Currently errors on register read/write/update are reported with an error code and the corresponding function but does not provide any details on the which register number did it actually fail. register number can give better clue and it should be easy to locate the code and fix. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Personally I found this patch very useful while developing and debugging. Ex: below error "ASoC: error at soc_component_read_no_lock on soc@0:codec: -16" did not provide much info except that it failed to read, but after this patch error message looks like: "ASoC: error at soc_component_read_no_lock on soc@0:codec for register: [0x00003125] -16" which was easy to locate and fix. sound/soc/soc-component.c | 26 ++++++++++++++++---------- 1 file changed, 16 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index a08a897c5230..c76ff9c59dfb 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -13,9 +13,10 @@ #include <sound/soc.h> #include <linux/bitops.h> -#define soc_component_ret(dai, ret) _soc_component_ret(dai, __func__, ret) +#define soc_component_ret(dai, ret) _soc_component_ret(dai, __func__, ret, -1) +#define soc_component_ret_reg_rw(dai, ret, reg) _soc_component_ret(dai, __func__, ret, reg) static inline int _soc_component_ret(struct snd_soc_component *component, - const char *func, int ret) + const char *func, int ret, int reg) { /* Positive/Zero values are not errors */ if (ret >= 0) @@ -27,9 +28,14 @@ static inline int _soc_component_ret(struct snd_soc_component *component, case -ENOTSUPP: break; default: - dev_err(component->dev, - "ASoC: error at %s on %s: %d\n", - func, component->name, ret); + if (reg == -1) + dev_err(component->dev, + "ASoC: error at %s on %s: %d\n", + func, component->name, ret); + else + dev_err(component->dev, + "ASoC: error at %s on %s for register: [0x%08x] %d\n", + func, component->name, reg, ret); } return ret; @@ -687,7 +693,7 @@ static unsigned int soc_component_read_no_lock( ret = -EIO; if (ret < 0) - return soc_component_ret(component, ret); + return soc_component_ret_reg_rw(component, ret, reg); return val; } @@ -723,7 +729,7 @@ static int soc_component_write_no_lock( else if (component->driver->write) ret = component->driver->write(component, reg, val); - return soc_component_ret(component, ret); + return soc_component_ret_reg_rw(component, ret, reg); } /** @@ -765,7 +771,7 @@ static int snd_soc_component_update_bits_legacy( mutex_unlock(&component->io_mutex); - return soc_component_ret(component, ret); + return soc_component_ret_reg_rw(component, ret, reg); } /** @@ -793,7 +799,7 @@ int snd_soc_component_update_bits(struct snd_soc_component *component, mask, val, &change); if (ret < 0) - return soc_component_ret(component, ret); + return soc_component_ret_reg_rw(component, ret, reg); return change; } EXPORT_SYMBOL_GPL(snd_soc_component_update_bits); @@ -829,7 +835,7 @@ int snd_soc_component_update_bits_async(struct snd_soc_component *component, mask, val, &change); if (ret < 0) - return soc_component_ret(component, ret); + return soc_component_ret_reg_rw(component, ret, reg); return change; } EXPORT_SYMBOL_GPL(snd_soc_component_update_bits_async); -- 2.21.0

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[RFC PATCH v3 00/13] ASoC : soc-pcm: fix trigger race conditions with shared BE
by Pierre-Louis Bossart 15 Oct '21

15 Oct '21

We've been adding a 'deep buffer' PCM device to several SOF topologies in order to reduce power consumption. The typical use-case would be music playback over a headset: this additional PCM device provides more buffering and longer latencies, leaving the rest of the system sleep for longer periods. Notifications and 'regular' low-latency audio playback would still use the 'normal' PCM device and be mixed with the 'deep buffer' before rendering on the headphone endpoint. The tentative direction would be to expose this alternate device to PulseAudio/PipeWire/CRAS via the UCM SectionModifier definitions. That seemed a straightforward topology change until our automated validation stress tests started reporting issues on SoundWire platforms, when e.g. two START triggers might be send and conversely the STOP trigger is never sent. The SoundWire stream state management flagged inconsistent states when the two 'normal' and 'deep buffer' devices are used concurrently with rapid play/stop/pause monkey testing. Looking at the soc-pcm.c code, it seems that the BE state management needs a lot of love. a) there is no consistent protection for the BE state. In some parts of the code, the state updates are protected by a spinlock but in the trigger they are not. When we open/play/close the two PCM devices in stress tests, we end-up testing a state that is being modified. That can't be good. b) there is a conceptual deadlock: on stop we check the FE states to see if a shared BE can be stopped, but since we trigger the BE first the FE states have not been modified yet, so the TRIGGER_STOP is never sent. This patchset suggests the removal of the dedicated 'dpcm_lock', and the use of the FE PCM lock before walking through the BE list, a mutual exclusion between triggers using the BE PCM lock, and the use of a refcount to decide when to trigger the BE. With these patches I am able to run our entire validation suite without any issues with this new 'deep buffer' topology, and no regressions on existing solutions [1] One might ask 'how come we didn't see this earlier'? The answer is probably that the .trigger callbacks in most implementations seems to perform DAPM operations, and sending the triggers multiple times is not an issue. In the case of SoundWire, we do use the .trigger callback to reconfigure the bus using the 'bank switch' mechanism. It could be acceptable to tolerate a trigger multiple times, but the deadlock on stop cannot be fixed at the SoundWire layer alone. I chose to send this patchset as an RFCv3 to gather more feedback and make use others know about DPCM issues. We're going to spend more time on this but if others can provide feedback/test results it would be greatly appreciated. The change in the locking model could be problematic on other platforms so we do want more time to comment/test before even considering a merge. Opens: 1) is this the right solution? The DPCM code is far from simple, has notions such as SND_SOC_DPCM_UPDATE_NO and 'trigger_pending' that I have no background on. It's not clear if these cases are still needed with the locking changes. 2) There are other reports of kernel oopses [2] that seem related to the lack of protection. I'd be good to confirm if this patchset solve these problems as well. [1] https://github.com/thesofproject/linux/pull/3146 [2] https://lore.kernel.org/alsa-devel/002f01d7b4f5$c030f4a0$4092dde0$@samsung.… changes since RFCv2: Removal of dpcm_lock to use FE PCM locks (credits to Takashi Iwai for the suggestion). The FE PCM lock is now used before each use of for_each_dpcm_be() - with the exception of the trigger where the lock is already taken. This change is also applied in drivers which make use of this loop (compress, SH, FSL). Addition of BE PCM lock to deal with mutual exclusion between triggers for the same BE. Alignment of the BE atomicity on the FE on connections, this is required to avoid sleeping in atomic context. Additional cleanups (indentation, static functions) changes since RFC v1: Removed unused function Removed exported symbols only used in soc-pcm.c, used static instead Use a mutex instead of a spinlock Protect all for_each_dpcm_be() loops Fix bugs introduced in the refcount Pierre-Louis Bossart (13): ASoC: soc-pcm: remove snd_soc_dpcm_fe_can_update() ASoC: soc-pcm: don't export local functions, use static ASoC: soc-pcm: use proper indentation on 'continue' ASoC: soc-pcm: introduce snd_soc_dpcm_fe_lock_irq/unlock_irq() ASoC: soc-pcm: align BE 'atomicity' with that of the FE ASoC: soc-pcm: remove dpcm spin_lock, use PCM stream lock ASoC: soc-pcm: protect for_each_dpcm_be() loops ASoC: soc-compress: protect for_each_dpcm_be() loops ASoC: sh: rcar: protect for_each_dpcm_be() loops ASoC: fsl: asrc_dma: protect for_each_dpcm_be() loops ASoC: soc-pcm: serialize BE triggers ASoC: soc-pcm: test refcount before triggering ASoC: soc-pcm: fix BE handling of PAUSE_RELEASE include/sound/soc-dpcm.h | 20 +-- include/sound/soc.h | 2 - sound/soc/fsl/fsl_asrc_dma.c | 2 + sound/soc/sh/rcar/core.c | 2 + sound/soc/soc-compress.c | 4 + sound/soc/soc-core.c | 1 - sound/soc/soc-pcm.c | 253 ++++++++++++++++++++++++----------- 7 files changed, 187 insertions(+), 97 deletions(-) -- 2.25.1

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[GIT PULL] ASoC fixes for v5.15-rc5
by Mark Brown 15 Oct '21

15 Oct '21

The following changes since commit cfacfefd382af3b42905108b54f02820dca225c4: ASoC: SOF: trace: Omit error print when waking up trace sleepers (2021-09-17 13:16:36 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.15-rc5 for you to fetch changes up to 6b9b546dc00797c74bef491668ce5431ff54e1e2: ASoC: wm8960: Fix clock configuration on slave mode (2021-10-13 16:25:33 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.15 A colletion of smallish mostly driver specific fixes, the biggest thing here is fixing some of the core code to generate change notifications properly when writing to controls which will fix issues with UIs not showing the correct values. There's one build fix here with a slightly misleading changelog saying it's adding IRQ config support, it's adding a missing select of the regmap-irq code rather than adding a feature. ---------------------------------------------------------------- Andy Shevchenko (4): ASoC: Intel: bytcht_es8316: Get platform data via dev_get_platdata() ASoC: Intel: bytcht_es8316: Use temporary variable for struct device ASoC: Intel: bytcht_es8316: Switch to use gpiod_get_optional() ASoC: Intel: bytcht_es8316: Utilize dev_err_probe() to avoid log saturation Hans de Goede (1): ASoC: nau8824: Fix headphone vs headset, button-press detection no longer working Mark Brown (2): ASoC: pcm179x: Add missing entries SPI to device ID table ASoC: cs4341: Add SPI device ID table Peter Rosin (1): ASoC: pcm512x: Mend accesses to the I2S_1 and I2S_2 registers Shengjiu Wang (2): ASoC: fsl_xcvr: Fix channel swap issue with ARC ASoC: wm8960: Fix clock configuration on slave mode Srinivasa Rao Mandadapu (2): ASoC: wcd938x: Fix jack detection issue ASoC: codec: wcd938x: Add irq config support Stefan Binding (1): ASoC: cs42l42: Ensure 0dB full scale volume is used for headsets Takashi Iwai (1): ASoC: DAPM: Fix missing kctl change notifications Yang Yingliang (1): ASoC: soc-core: fix null-ptr-deref in snd_soc_del_component_unlocked() sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/cs42l42.c | 16 +++------------ sound/soc/codecs/cs4341.c | 7 +++++++ sound/soc/codecs/nau8824.c | 4 ++-- sound/soc/codecs/pcm179x-spi.c | 1 + sound/soc/codecs/pcm512x.c | 2 ++ sound/soc/codecs/wcd938x.c | 6 +++--- sound/soc/codecs/wm8960.c | 13 +++++++++--- sound/soc/fsl/fsl_xcvr.c | 17 +++++++++++----- sound/soc/intel/boards/bytcht_es8316.c | 37 +++++++++++----------------------- sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 13 +++++++----- 12 files changed, 62 insertions(+), 56 deletions(-)

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[PATCH v2] Fixes HP Spectre x360 15-eb1xxx speakers
by Davide Baldo 15 Oct '21

15 Oct '21

In laptop 'HP Spectre x360 Convertible 15-eb1xxx/8811' both front and rear speakers are silent, this patch fixes that by overriding the pin layout and by initializing the amplifier which needs a GPIO pin to be set to 1 then 0, similar to the existing HP Spectre x360 14 model. In order to have volume control, both front and rear speakers were forced to use the DAC1. This patch also correctly map the mute LED but since there is no microphone on/off switch exposed by the alsa subsystem it never turns on by itself. There are still known audio issues in this laptop: headset microphone doesn't work, the button to mute/unmute microphone is not yet mapped, the LED of the mute/unmute speakers doesn't seems to be exposed via GPIO and never turns on. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213953 Signed-off-by: Davide Baldo <davide(a)baldo.me> --- sound/pci/hda/patch_realtek.c | 46 +++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0b9230a274b0..dda979eaeb44 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6414,6 +6414,44 @@ static void alc_fixup_no_int_mic(struct hda_codec *codec, } } +/* GPIO1 = amplifier on/off + * GPIO3 = mic mute LED + */ +static void alc285_fixup_hp_spectre_x360_eb1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t conn[] = { 0x02 }; + + struct alc_spec *spec = codec->spec; + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170110 }, /* front/high speakers */ + { 0x17, 0x90170130 }, /* back/bass speakers */ + { } + }; + + //enable micmute led + alc_fixup_hp_gpio_led(codec, action, 0x00, 0x04); + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->micmute_led_polarity = 1; + /* needed for amp of back speakers */ + spec->gpio_mask |= 0x01; + spec->gpio_dir |= 0x01; + snd_hda_apply_pincfgs(codec, pincfgs); + /* share DAC to have unified volume control */ + snd_hda_override_conn_list(codec, 0x14, ARRAY_SIZE(conn), conn); + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + break; + case HDA_FIXUP_ACT_INIT: + /* need to toggle GPIO to enable the amp of back speakers */ + alc_update_gpio_data(codec, 0x01, true); + msleep(100); + alc_update_gpio_data(codec, 0x01, false); + break; + } +} + static void alc285_fixup_hp_spectre_x360(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6548,6 +6586,7 @@ enum { ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, ALC280_FIXUP_HP_9480M, ALC245_FIXUP_HP_X360_AMP, + ALC285_FIXUP_HP_SPECTRE_X360_EB1, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13, @@ -8240,6 +8279,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_spectre_x360, }, + [ALC285_FIXUP_HP_SPECTRE_X360_EB1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_spectre_x360_eb1 + }, [ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_ideapad_s740_coef, @@ -8567,6 +8610,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), + SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), @@ -8987,6 +9032,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, + {.id = ALC285_FIXUP_HP_SPECTRE_X360_EB1, .name = "alc285-hp-spectre-x360-eb1"}, {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"}, -- 2.32.0

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[PATCH] sound/usb: Provide quirk for Sennheiser GSP670 Headset
by Brendan Grieve 15 Oct '21

15 Oct '21

As per discussion at: https://github.com/szszoke/sennheiser-gsp670-pulseaudio-profile/issues/13 The GSP670 has 2 playback and 1 recording device that by default are detected in an incompatible order for alsa. This may have been done to make it compatible for the console by the manufacturer and only affects the latest firmware which uses its own ID. This quirk will resolve this by reordering the channels. Signed-off-by: Brendan Grieve <brendan(a)grieve.com.au> --- sound/usb/quirks-table.h | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index de18fff69280..2af8c68fac27 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -4012,6 +4012,38 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Sennheiser GSP670 + * Change order of interfaces loaded + */ + USB_DEVICE(0x1395, 0x0300), + .bInterfaceClass = USB_CLASS_PER_INTERFACE, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + // Communication + { + .ifnum = 3, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + // Recording + { + .ifnum = 4, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + // Main + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC #undef USB_AUDIO_DEVICE -- 2.25.1

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[PATCH] ASoC: rsnd: Add of_node_put() before return
by Wan Jiabing 15 Oct '21

15 Oct '21

Fix following coccicheck warning: ./sound/soc/sh/rcar/core.c:1223:1-23: WARNING: Function for_each_child_of_node should have of_node_put() before return Early exits from for_each_child_of_node should decrement the node reference counter. Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/soc/sh/rcar/core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 978bd0406729..6a8fe0da7670 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1225,6 +1225,7 @@ int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name if (i < 0) { dev_err(dev, "strange node numbering (%s)", of_node_full_name(node)); + of_node_put(np); return 0; } i++; -- 2.20.1

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