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rawmidi driver dropping some sysex commands
by Kolten Pearson 17 Dec '21

17 Dec '21

I have been working on reverse engineering the sysex commands to control an Arturia MiniLab mkII controller, and have run into an issue where the rawmidi driver is dropping some of the commands. Here is a snippet of my code that illustrates the problem --- snd_rawmidi_t* midi_input; snd_rawmidi_t* midi_output; void midi_init() { int ret = snd_rawmidi_open(&midi_input, &midi_output, "hw:3,0,0", SND_RAWMIDI_NONBLOCK); assert (ret == 0 && "could not open midi device"); uint8_t pad_list[] = { MA_PAD_1, MPC_NO_COLOR, MA_PAD_2, MPC_RED, MA_PAD_3, MPC_GREEN, MA_PAD_4, MPC_YELLOW, MA_PAD_5, MPC_BLUE, MA_PAD_6, MPC_CYAN, MA_PAD_7, MPC_MAGENTA, MA_PAD_8, MPC_WHITE, MA_PAD_9, MPC_NO_COLOR, MA_PAD_10, MPC_RED, MA_PAD_11, MPC_GREEN, MA_PAD_12, MPC_YELLOW, MA_PAD_13, MPC_BLUE, MA_PAD_14, MPC_CYAN, MA_PAD_15, MPC_MAGENTA, MA_PAD_16, MPC_WHITE, }; for (int i = 0; i < 16; i++) { midi_set(MC_ON_OFF, pad_list[2*i], 0x08); //sets the pad to be a control instead of a note midi_set(MC_MIN_RANGE, pad_list[2*i], 0); midi_set(MC_MAX_RANGE, pad_list[2*i], 1); midi_set(MC_MODE_SET, pad_list[2*i], 1); //set it to gate, 0 is toggle midi_set(MC_PAD_COLOR, pad_list[2*i], pad_list[2*i+1]); } midi_set(MC_GLOBAL_SET, MA_PAD_BACKLIGHTS, 0x0f); //updates the pad backlights uint8_t knob_list[] = { MA_KNOB_1, MA_KNOB_2, MA_KNOB_3, MA_KNOB_4, MA_KNOB_5, MA_KNOB_6, MA_KNOB_7, MA_KNOB_8, MA_KNOB_9, MA_KNOB_10, MA_KNOB_11, MA_KNOB_12, MA_KNOB_13, MA_KNOB_14, MA_KNOB_15, MA_KNOB_16 }; for (int i = 0; i < 16; i++) { midi_set(MC_MODE_SET, knob_list[i], 0x02); //sets it to relative mode where 0-3 is right turn, 7f-7d is left turn midi_set(MC_CC_SET, knob_list[i], i); } } void midi_set(enum midi_set_command command, enum midi_set_address address, uint8_t data) { uint8_t raw_command[] = {0xf0, 0x00, 0x20, 0x6b, 0x7f, 0x42, 0x02, 0x00, command, address, data, 0xf7}; int count = snd_rawmidi_write(midi_output, raw_command, sizeof(raw_command)); assert(count == sizeof(raw_command) && "bad write"); //SDL_Delay(1); int ret = snd_rawmidi_drain(midi_output); assert(ret == 0 && "could not drain"); } --- When this code is run, the minilab will only respond to a nondeterministic number of calls to the midi_set function and ignore the rest. However if the SDL_Delay(1) line is uncommented, it will work correctly every time (This function call causes the process to sleep for at least 1 millisecond). I feel like the call to snd_rawmidi_drain should make the SDL_Delay call unnecessary, but that is not the case. Is there something I am not understanding about how the api works? Thanks for any help!

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ALSA 1.2.6.3 release
by Jaroslav Kysela 17 Dec '21

17 Dec '21

Hello all, new ALSA userspace packages were released. This is third bugfix release. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-ucm-conf Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.6.2_v1.2.6.3 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[PATCH v5 00/11] Support HDMI audio on NVIDIA Tegra20
by Dmitry Osipenko 17 Dec '21

17 Dec '21

This series revives Tegra20 S/PDIF driver which was upstreamed long time ago, but never was used. It also turns Tegra DRM HDMI driver into HDMI audio CODEC provider. Finally, HDMI audio is enabled in device-trees. For now the audio is enable only for Acer A500 tablet and Toshiba AC100 netbook because they're already supported by upstream, later on ASUS TF101 tablet will join them. (!) These patches are made on top of stable dmaengine branch from Vinod Koul. The following changes since commit fa55b7dcdc43c1aa1ba12bca9d2dd4318c2a0dbf: Linux 5.16-rc1 (2021-11-14 13:56:52 -0800) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/dmaengine.git tags/dmaengine_topic_slave_id_removal_5.17 for you to fetch changes up to 3c219644075795a99271d345efdfa8b256e55161: dmaengine: remove slave_id config field (2021-12-17 11:23:56 +0530) Changelog: v5: - Dropped all patches that were already applied by Thierry Reding. - Made minor changes that were suggested by Thierry Reding. - Added acks from Thierry Reding. - Rebased patches on top of stable dmaengine branch from Vinod Koul. v4: - Added patches that update multi_v7_defconfig with the enabled S/PDIF and APB DMA drivers. v3: - Renamed S/PDIF device-tree clocks as was suggested by Rob Herring. - Added r-bs and acks that were given by Rob Herring to v2. v2: - Corrected I2S yaml problem that was reported by the DT bot for v1 by removing the non-existent required clock-names property. - Removed assigned-clocks property from S/PDIF yaml since this property is now inherited from the clocks property. - Reordered the "tegra20: spdif: Set FIFO trigger level" patch, making it the first sound/soc patch in the series, like it was suggested by Mark Brown in the comment to v1. Also reworded commit message of this patch to *not* make it looks like it should be backported to stable kernels. Dmitry Osipenko (11): ASoC: dt-bindings: Add binding for Tegra20 S/PDIF ASoC: dt-bindings: tegra20-i2s: Convert to schema ASoC: dt-bindings: tegra20-i2s: Document new nvidia,fixed-parent-rate property ASoC: tegra20: spdif: Set FIFO trigger level ASoC: tegra20: spdif: Support device-tree ASoC: tegra20: spdif: Improve driver's code ASoC: tegra20: spdif: Use more resource-managed helpers ASoC: tegra20: spdif: Reset hardware ASoC: tegra20: spdif: Support system suspend ASoC: tegra20: spdif: Filter out unsupported rates ASoC: tegra20: i2s: Filter out unsupported rates .../bindings/sound/nvidia,tegra20-i2s.txt | 30 --- .../bindings/sound/nvidia,tegra20-i2s.yaml | 77 +++++++ .../bindings/sound/nvidia,tegra20-spdif.yaml | 85 ++++++++ sound/soc/tegra/tegra20_i2s.c | 49 +++++ sound/soc/tegra/tegra20_spdif.c | 198 +++++++++++++----- sound/soc/tegra/tegra20_spdif.h | 1 + sound/soc/tegra/tegra_pcm.c | 6 + sound/soc/tegra/tegra_pcm.h | 1 + 8 files changed, 359 insertions(+), 88 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml -- 2.33.1

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[PATCH 0/6] ASoC : soc-pcm: fix trigger race conditions with shared BE
by Pierre-Louis Bossart 17 Dec '21

17 Dec '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 follows the design suggested by Takashi Iwai. By default the protection relies on the 'pcm_mutex', except for the FE and BE triggers where the mutex cannot be used. In this case, the FE PCM lock is used instead. In the cases where a BE is added/removed, the pcm_mutex and FE PCM lock are both taken. In addition, the BE PCM lock is used to serialize access to a shared 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]. The tests were reproduced by Bard Liao for SoundWire devices. 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 level alone. Opens: 1) The issues reported by Nvidia on the RFCv3 may or may not be present. We'd need test results to make sure the locking update does not introduce a regression on Tegra. 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 RFCv3: Used two patches from Takashi. We now use the pcm_mutex, the FE stream lock when adding and deleting a BE, and the BE stream lock to handle concurrency between streams using the same BE. Added a patch to use GFP_ATOMIC for the DPCM structure. Fixed PAUSE_RELEASE transition (GitHub comment from Kai Vehmanen) 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 (4): ASoC: soc-pcm: use GFP_ATOMIC for dpcm structure ASoC: soc-pcm: align BE 'atomicity' with that of the FE ASoC: soc-pcm: test refcount before triggering ASoC: soc-pcm: fix BE handling of PAUSE_RELEASE Takashi Iwai (2): ASoC: soc-pcm: Fix and cleanup DPCM locking ASoC: soc-pcm: serialize BE triggers include/sound/soc-dpcm.h | 2 + include/sound/soc.h | 2 - sound/soc/soc-core.c | 1 - sound/soc/soc-pcm.c | 351 +++++++++++++++++++++++++++------------ 4 files changed, 246 insertions(+), 110 deletions(-) -- 2.25.1

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[PATCH] ASoC: max98357a: Add mixer control to mute/unmute speaker
by Ajit Kumar Pandey 17 Dec '21

17 Dec '21

Add "Playback Switch" mixer control to mute or unmute speaker playback from ucm conf depend on use cases. Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> --- sound/soc/codecs/max98357a.c | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 918812763884..9b2f16ab4498 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -73,6 +73,36 @@ static int max98357a_sdmode_event(struct snd_soc_dapm_widget *w, return 0; } +static int speaker_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct max98357a_priv *max98357a = snd_soc_component_get_drvdata(component); + + ucontrol->value.enumerated.item[0] = max98357a->sdmode_switch; + + return 0; +} + +static int speaker_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct max98357a_priv *max98357a = snd_soc_component_get_drvdata(component); + int mode = ucontrol->value.enumerated.item[0]; + + max98357a->sdmode_switch = mode; + gpiod_set_value_cansleep(max98357a->sdmode, mode); + dev_dbg(component->dev, "set sdmode to %d", mode); + + return 0; +} + +static const struct snd_kcontrol_new max98357a_snd_controls[] = { + SOC_SINGLE_BOOL_EXT("Playback Switch", 0, + speaker_mute_get, speaker_mute_put), +}; + static const struct snd_soc_dapm_widget max98357a_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Speaker"), SND_SOC_DAPM_OUT_DRV_E("SD_MODE", SND_SOC_NOPM, 0, 0, NULL, 0, @@ -86,6 +116,8 @@ static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { }; static const struct snd_soc_component_driver max98357a_component_driver = { + .controls = max98357a_snd_controls, + .num_controls = ARRAY_SIZE(max98357a_snd_controls), .dapm_widgets = max98357a_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98357a_dapm_widgets), .dapm_routes = max98357a_dapm_routes, -- 2.25.1

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[PATCH v5 00/10] Add support for CS35L41 in HDA systems
by Lucas Tanure 17 Dec '21

17 Dec '21

Add support for laptops that have CS35L41 connected to an HDA codec by I2S and direct I2C connection to the CPU. Laptops that use CS35L41 and are SPI will be added in the future, after the support for it is resolved at i2c-multi-instantiate driver. i2c-multi-instantiate thread: https://lkml.org/lkml/2021/12/10/557 Hardware: - Some laptops have two CS35L41 amplifiers, connected to Realtek ALC287 by an I2S bus and by and direct I2C to the CPU. - The ALC287 codec is connected to the CPU by an HDA bus. - The CS35L41 has a DSP which will require firmware to be loaded. Architecture: - Creation of a library of shared functions for CS35L41 ASoC and HDA - HDA codec driver (RealTek) and CS35L41 HDA driver are combined by using component binding, where it uses device names to find the components and bind to the master - The HDA CS35L41 driver applies pre-defined registers sequences for each action in playback for HDA Sound card Changes from V4: - Save index received from ACPI for latter use in DSP Support - Move CS35L41_DSP1_CCM_CORE_CTRL to regmap_write so doesn't affect DSP hibernation feature V4: https://lkml.org/lkml/2021/12/14/487 Changes from V3: - SPI bus driver starter added - Use separate modules approach instead of liking library - Add CSC3551 ACPI _HID for more I2C laptops - Removed regulators from HDA driver - Add note about Non-conforming _HID V3: https://lkml.org/lkml/2021/11/23/723 Changes from V2: - Not an RFC - Create a new HDA driver for CS35L41 instead of using the ASoC one - Use component binding and device names to find the CS35L41 driver - Create a shared library for ASoC and HDA CS35L41 drivers v2: https://lkml.org/lkml/2021/10/8/344 Lucas Tanure (9): ASoC: cs35l41: Convert tables to shared source code ASoC: cs35l41: Move cs35l41_otp_unpack to shared code ASoC: cs35l41: Move power initializations to reg_sequence ASoC: cs35l41: Create shared function for errata patches ASoC: cs35l41: Create shared function for setting channels ASoC: cs35l41: Create shared function for boost configuration hda: cs35l41: Add support for CS35L41 in HDA systems ACPI / scan: Create platform device for CLSA0100 and CSC3551 ACPI nodes ALSA: hda/realtek: Add support for Legion 7 16ACHg6 laptop Stefan Binding (1): ALSA: hda/realtek: Add CS35L41 support for Thinkpad laptops MAINTAINERS | 2 + drivers/acpi/scan.c | 3 + drivers/platform/x86/i2c-multi-instantiate.c | 11 + include/sound/cs35l41.h | 739 ++++++++++++++++++ sound/pci/hda/Kconfig | 27 + sound/pci/hda/Makefile | 10 + sound/pci/hda/cs35l41_hda.c | 527 +++++++++++++ sound/pci/hda/cs35l41_hda.h | 69 ++ sound/pci/hda/cs35l41_hda_i2c.c | 66 ++ sound/pci/hda/cs35l41_hda_spi.c | 63 ++ sound/pci/hda/hda_component.h | 20 + sound/pci/hda/patch_realtek.c | 144 ++++ sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/cs35l41-i2c.c | 1 - .../{cs35l41-tables.c => cs35l41-lib.c} | 355 ++++++++- sound/soc/codecs/cs35l41-spi.c | 1 - sound/soc/codecs/cs35l41.c | 360 +-------- sound/soc/codecs/cs35l41.h | 734 ----------------- 19 files changed, 2051 insertions(+), 1092 deletions(-) create mode 100644 sound/pci/hda/cs35l41_hda.c create mode 100644 sound/pci/hda/cs35l41_hda.h create mode 100644 sound/pci/hda/cs35l41_hda_i2c.c create mode 100644 sound/pci/hda/cs35l41_hda_spi.c create mode 100644 sound/pci/hda/hda_component.h rename sound/soc/codecs/{cs35l41-tables.c => cs35l41-lib.c} (71%) -- 2.34.1

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[PATCH] ASoC: wm8962: potential use of error pointer
by Jiasheng Jiang 17 Dec '21

17 Dec '21

The return value of devm_clk_get() needs to be checked. To avoid use of error pointer in case of the failure of alloc. Fixes: d74e9e7090ae ("ASoC: wm8962: Add device tree binding") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/codecs/wm8962.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 34080f497584..66693bf17763 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3540,6 +3540,8 @@ static int wm8962_set_pdata_from_of(struct i2c_client *i2c, } pdata->mclk = devm_clk_get(&i2c->dev, NULL); + if (IS_ERR(pdata->mclk)) + return -ENOMEM; return 0; } -- 2.25.1

3 2

[PATCH 1/2] ASoC: mediatek: mt8195: update control for RT5682 series
by Trevor Wu 17 Dec '21

17 Dec '21

Playback pop is observed and the root cause is the reference clock provided by MT8195 is diabled before RT5682 finishes the control flow. To ensure the reference clock supplied to RT5682 is disabled after RT5682 finishes all register controls. We replace BCLK with MCLK for RT5682 reference clock, and makes use of set_bias_level_post to handle MCLK which guarantees MCLK is off after all RT5682 register access. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- .../mt8195/mt8195-mt6359-rt1011-rt5682.c | 66 +++++++++++++++- .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 76 ++++++++++++++++--- 2 files changed, 128 insertions(+), 14 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c index 5cdbfaafd479..05359365f200 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c @@ -37,6 +37,7 @@ struct mt8195_mt6359_rt1011_rt5682_priv { struct snd_soc_jack headset_jack; struct snd_soc_jack dp_jack; struct snd_soc_jack hdmi_jack; + struct clk *i2so1_mclk; }; static const struct snd_soc_dapm_widget @@ -87,8 +88,8 @@ static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream, return ret; } - ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL1, RT5682_PLL1_S_BCLK1, - rate * 64, rate * 512); + ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL1, RT5682_PLL1_S_MCLK, + rate * 256, rate * 512); if (ret) { dev_err(card->dev, "failed to set pll\n"); return ret; @@ -101,7 +102,7 @@ static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream, return ret; } - return snd_soc_dai_set_sysclk(cpu_dai, 0, rate * 128, + return snd_soc_dai_set_sysclk(cpu_dai, 0, rate * 256, SND_SOC_CLOCK_OUT); } @@ -565,6 +566,51 @@ static const struct snd_soc_ops mt8195_capture_ops = { .startup = mt8195_capture_startup, }; +static int mt8195_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) +{ + struct snd_soc_component *component = dapm->component; + struct mt8195_mt6359_rt1011_rt5682_priv *priv = + snd_soc_card_get_drvdata(card); + int ret = 0; + + /* + * It's required to control mclk directly in the set_bias_level_post + * function for rt5682 and rt5682s codec, or the unexpected pop happens + * at the end of playback. + */ + if (!component || + (strcmp(component->name, RT5682_DEV0_NAME) && + strcmp(component->name, RT5682S_DEV0_NAME))) + return 0; + + if (IS_ERR(priv->i2so1_mclk)) + return 0; + + switch (level) { + case SND_SOC_BIAS_OFF: + if (!__clk_is_enabled(priv->i2so1_mclk)) + return 0; + + dev_dbg(card->dev, "Disable i2so1"); + clk_disable_unprepare(priv->i2so1_mclk); + break; + case SND_SOC_BIAS_ON: + dev_dbg(card->dev, "Enable i2so1"); + ret = clk_prepare_enable(priv->i2so1_mclk); + if (ret) { + dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); + return ret; + } + break; + default: + break; + } + + return ret; +} + + enum { DAI_LINK_DL2_FE, DAI_LINK_DL3_FE, @@ -1040,6 +1086,7 @@ static struct snd_soc_card mt8195_mt6359_rt1011_rt5682_soc_card = { .num_dapm_routes = ARRAY_SIZE(mt8195_mt6359_rt1011_rt5682_routes), .codec_conf = rt1011_amp_conf, .num_configs = ARRAY_SIZE(rt1011_amp_conf), + .set_bias_level_post = mt8195_set_bias_level_post, }; static int mt8195_mt6359_rt1011_rt5682_dev_probe(struct platform_device *pdev) @@ -1072,6 +1119,19 @@ static int mt8195_mt6359_rt1011_rt5682_dev_probe(struct platform_device *pdev) return -EINVAL; } + priv->i2so1_mclk = devm_clk_get(&pdev->dev, "i2so1_mclk"); + if (IS_ERR(priv->i2so1_mclk)) { + ret = PTR_ERR(priv->i2so1_mclk); + if (ret == -ENOENT) { + dev_dbg(&pdev->dev, + "Failed to get i2so1_mclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get i2so1_mclk, err:%d\n", ret); + return ret; + } + for_each_card_prelinks(card, i, dai_link) { if (!dai_link->platforms->name) dai_link->platforms->of_node = priv->platform_node; diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index fa50a31e9718..b2c3b57da9c4 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -50,6 +50,7 @@ struct mt8195_mt6359_rt1019_rt5682_priv { struct snd_soc_jack headset_jack; struct snd_soc_jack dp_jack; struct snd_soc_jack hdmi_jack; + struct clk *i2so1_mclk; }; static const struct snd_soc_dapm_widget @@ -96,8 +97,6 @@ static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int rate = params_rate(params); - unsigned int mclk_fs_ratio = 128; - unsigned int mclk_fs = rate * mclk_fs_ratio; int bitwidth; int ret; @@ -113,25 +112,22 @@ static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream, return ret; } - ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL1, - RT5682_PLL1_S_BCLK1, - params_rate(params) * 64, - params_rate(params) * 512); + ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL1, RT5682_PLL1_S_MCLK, + rate * 256, rate * 512); if (ret) { dev_err(card->dev, "failed to set pll\n"); return ret; } - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5682_SCLK_S_PLL1, - params_rate(params) * 512, - SND_SOC_CLOCK_IN); + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + rate * 512, SND_SOC_CLOCK_IN); if (ret) { dev_err(card->dev, "failed to set sysclk\n"); return ret; } - return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_fs, SND_SOC_CLOCK_OUT); + return snd_soc_dai_set_sysclk(cpu_dai, 0, rate * 256, + SND_SOC_CLOCK_OUT); } static const struct snd_soc_ops mt8195_rt5682_etdm_ops = { @@ -564,6 +560,50 @@ static const struct snd_soc_ops mt8195_capture_ops = { .startup = mt8195_capture_startup, }; +static int mt8195_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) +{ + struct snd_soc_component *component = dapm->component; + struct mt8195_mt6359_rt1019_rt5682_priv *priv = + snd_soc_card_get_drvdata(card); + int ret = 0; + + /* + * It's required to control mclk directly in the set_bias_level_post + * function for rt5682 and rt5682s codec, or the unexpected pop happens + * at the end of playback. + */ + if (!component || + (strcmp(component->name, RT5682_DEV0_NAME) && + strcmp(component->name, RT5682S_DEV0_NAME))) + return 0; + + if (IS_ERR(priv->i2so1_mclk)) + return 0; + + switch (level) { + case SND_SOC_BIAS_OFF: + if (!__clk_is_enabled(priv->i2so1_mclk)) + return 0; + + dev_dbg(card->dev, "Disable i2so1"); + clk_disable_unprepare(priv->i2so1_mclk); + break; + case SND_SOC_BIAS_ON: + dev_dbg(card->dev, "Enable i2so1"); + ret = clk_prepare_enable(priv->i2so1_mclk); + if (ret) { + dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); + return ret; + } + break; + default: + break; + } + + return ret; +} + enum { DAI_LINK_DL2_FE, DAI_LINK_DL3_FE, @@ -1203,6 +1243,7 @@ static struct snd_soc_card mt8195_mt6359_rt1019_rt5682_soc_card = { .num_dapm_widgets = ARRAY_SIZE(mt8195_mt6359_rt1019_rt5682_widgets), .dapm_routes = mt8195_mt6359_rt1019_rt5682_routes, .num_dapm_routes = ARRAY_SIZE(mt8195_mt6359_rt1019_rt5682_routes), + .set_bias_level_post = mt8195_set_bias_level_post, }; static int mt8195_dailink_parse_of(struct snd_soc_card *card, struct device_node *np, @@ -1285,6 +1326,19 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) return -EINVAL; } + priv->i2so1_mclk = devm_clk_get(&pdev->dev, "i2so1_mclk"); + if (IS_ERR(priv->i2so1_mclk)) { + ret = PTR_ERR(priv->i2so1_mclk); + if (ret == -ENOENT) { + dev_dbg(&pdev->dev, + "Failed to get i2so1_mclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get i2so1_mclk, err:%d\n", ret); + return ret; + } + /* dai link */ priv->adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0); -- 2.18.0

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Re: [PATCH v2 1/1] ASoC: rt5682s: Add dapm switch to mute/unmute HP playback output
by Ajit Kumar Pandey 17 Dec '21

17 Dec '21

On 12/9/2021 7:55 AM, Shuming [范書銘] wrote: > [CAUTION: External Email] > >> Subject: [PATCH v2 1/1] ASoC: rt5682s: Add dapm switch to mute/unmute HP >> playback output >> >> Add dapm switch in playback path to mute or unmute HP output data. >> We will set and reset MUTE_SFT bit in RT5682S_HP_CTRL_1 register based on >> switch value to mute or unmute respective channel. >> >> Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> >> --- >> sound/soc/codecs/rt5682s.c | 19 +++++++++++++++++-- >> 1 file changed, 17 insertions(+), 2 deletions(-) >> >> diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index >> d49a4f68566d..21ab5f7df422 100644 >> --- a/sound/soc/codecs/rt5682s.c >> +++ b/sound/soc/codecs/rt5682s.c >> @@ -1573,6 +1573,14 @@ static const char * const >> rt5682s_adcdat_pin_select[] = { >> "ADCDAT1", "ADCDAT2", >> }; >> >> +/* Out Switch */ >> +static const struct snd_kcontrol_new hpol_switch = >> + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682S_HP_CTRL_1, >> + RT5682S_L_MUTE_SFT, 1, 1); >> +static const struct snd_kcontrol_new hpor_switch = >> + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682S_HP_CTRL_1, >> + RT5682S_R_MUTE_SFT, 1, 1); >> + >> static SOC_VALUE_ENUM_SINGLE_DECL(rt5682s_adcdat_pin_enum, >> RT5682S_GPIO_CTRL_1, RT5682S_GP4_PIN_SFT, >> RT5682S_GP4_PIN_MASK, >> rt5682s_adcdat_pin_select, rt5682s_adcdat_pin_values); @@ -1746,6 >> +1754,11 @@ static const struct snd_soc_dapm_widget >> rt5682s_dapm_widgets[] = { >> SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, >> rt5682s_hp_amp_event, >> SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), >> >> + SND_SOC_DAPM_SWITCH("HPOL Playback", SND_SOC_NOPM, 0, 0, >> + &hpol_switch), >> + SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0, >> + &hpor_switch), >> + >> /* CLK DET */ >> SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682S_CLK_DET, >> RT5682S_SYS_CLK_DET_SFT, 0, NULL, 0), @@ -1895,8 +1908,10 @@ >> static const struct snd_soc_dapm_route rt5682s_dapm_routes[] = { >> {"HP Amp", NULL, "CLKDET SYS"}, >> {"HP Amp", NULL, "SAR"}, >> >> - {"HPOL", NULL, "HP Amp"}, >> - {"HPOR", NULL, "HP Amp"}, >> + {"HPOL Playback", "Switch", "HP Amp"}, >> + {"HPOR Playback", "Switch", "HP Amp"}, >> + {"HPOL", NULL, "HPOL Playback"}, >> + {"HPOR", NULL, "HPOR Playback"}, >> }; > > RT5682S enables the 1 bit control for HP output. > Therefore, the setting of RT5682S_HP_CTRL_1[15][7] will be not effective. > > Ok, it was actually effective and muted audio when we test in our platform, possible because i'm tearing down dapm path based on switch value. If we don't have register bits available to mute i'll resubmit the change with virtual switch widget to disable DAPM and stop audio.

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Question about soc_pcm_apply_msb()
by Kuninori Morimoto 17 Dec '21

17 Dec '21

Hi ALSA ML At soc_pcm_apply_msb(), (A) part finds max sig_bits for Codec, and (B) part finds max sig_bits for CPU and, set it to substream via soc_pcm_set_msb() at (X), (Y). static void soc_pcm_apply_msb() { ... ^ for_each_rtd_codec_dais(rtd, i, codec_dai) { (A) ... v bits = max(pcm_codec->sig_bits, bits); } ^ for_each_rtd_cpu_dais(rtd, i, cpu_dai) { (B) ... v cpu_bits = max(pcm_cpu->sig_bits, cpu_bits); } (X) soc_pcm_set_msb(substream, bits); (Y) soc_pcm_set_msb(substream, cpu_bits); } I wonder do we need both (X) (Y) ? I think we can merge (A) and (B) (= find Codec/CPU max sig_bits), and call soc_pcm_set_msb() once, but am I misunderstand ? We have many patch around here, and below are the main patches. The 1st patch has both (X)(Y) code. new 19bdcc7aeed4169820be6a683c422fc06d030136 ("ASoC: Add multiple CPU DAI support for PCM ops") 57be92066f68e63bd4a72a65d45c3407c0cb552a ("ASoC: soc-pcm: cleanup soc_pcm_apply_msb()") c8dd1fec47d0b1875f292c40bed381b343e38b40 ("ASoC: pcm: Refactor soc_pcm_apply_msb for multicodecs") 58ba9b25454fe9b6ded804f69cb7ed4500b685fc ("ASoC: Allow drivers to specify how many bits are significant on a DAI") old Thank you for your help !! Best regards --- Kuninori Morimoto

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