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

[PATCH v4 0/2] Add reset-gpios handling for max98927
by Alejandro Tafalla 07 Oct '21

07 Oct '21

The max98927 codec on some devices (i.e. Xiaomi Mi A2 Lite phone) requires hardware-resetting the codec by driving a reset-gpio. This series adds support for it through an optional reset-gpios property. v4: * Correctly assert/deassert the GPIO states * Wait for the i2c port to be ready after reset * Reset device when removed v3: * Fix indentation on the dev_err_probe line v2: * Use dev_err_probe instead of dev_err Alejandro Tafalla (2): ASoC: max98927: Handle reset gpio when probing i2c dt-bindings: sound: max98927: Add reset-gpios optional property .../devicetree/bindings/sound/max9892x.txt | 3 +++ sound/soc/codecs/max98927.c | 25 +++++++++++++++++++ sound/soc/codecs/max98927.h | 1 + 3 files changed, 29 insertions(+) -- 2.33.0

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[PATCH] ASoC: SOF: topology: do not power down primary core during topology removal
by Peter Ujfalusi 07 Oct '21

07 Oct '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> When removing the topology components, do not power down the primary core. Doing so will result in an IPC timeout when the SOF PCI device runtime suspends. Fixes: 0dcdf84289fb ("ASoC: SOF: add a "core" parameter to widget loading functions") Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/topology.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 534f004f6162..1723dd11acd7 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2514,6 +2514,15 @@ static int sof_widget_unload(struct snd_soc_component *scomp, /* power down the pipeline schedule core */ pipeline = swidget->private; + + /* + * Runtime PM should still function normally if topology loading fails and + * it's components are unloaded. Do not power down the primary core so that the + * CTX_SAVE IPC can succeed during runtime suspend. + */ + if (pipeline->core == SOF_DSP_PRIMARY_CORE) + break; + ret = snd_sof_dsp_core_power_down(sdev, 1 << pipeline->core); if (ret < 0) dev_err(scomp->dev, "error: powering down pipeline schedule core %d\n", -- 2.33.0

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[PATCH] ASoC: DAPM: Fix missing kctl change notifications
by Takashi Iwai 07 Oct '21

07 Oct '21

The put callback of a kcontrol is supposed to return 1 when the value is changed, and this will be notified to user-space. However, some DAPM kcontrols always return 0 (except for errors), hence the user-space misses the update of a control value. This patch corrects the behavior by properly returning 1 when the value gets updated. Reported-and-tested-by: Hans de Goede <hdegoede(a)redhat.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-dapm.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7b67f1e19ae9..59d07648a7e7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2561,6 +2561,7 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, const char *pin, int status) { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + int ret = 0; dapm_assert_locked(dapm); @@ -2573,13 +2574,14 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(w, "pin configuration"); dapm_widget_invalidate_input_paths(w); dapm_widget_invalidate_output_paths(w); + ret = 1; } w->connected = status; if (status == 0) w->force = 0; - return 0; + return ret; } /** @@ -3583,14 +3585,15 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; + int ret; if (ucontrol->value.integer.value[0]) - snd_soc_dapm_enable_pin(&card->dapm, pin); + ret = snd_soc_dapm_enable_pin(&card->dapm, pin); else - snd_soc_dapm_disable_pin(&card->dapm, pin); + ret = snd_soc_dapm_disable_pin(&card->dapm, pin); snd_soc_dapm_sync(&card->dapm); - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); @@ -4023,7 +4026,7 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, rtd->params_select = ucontrol->value.enumerated.item[0]; - return 0; + return 1; } static void -- 2.26.2

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[PATCH v1 1/4] ASoC: Intel: bytcht_es8316: Get platform data via dev_get_platdata()
by Andy Shevchenko 07 Oct '21

07 Oct '21

Access to platform data via dev_get_platdata() getter to make code cleaner. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/boards/bytcht_es8316.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 1bb9b8e7bcc7..4360519fbb0c 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -456,12 +456,12 @@ static const struct dmi_system_id byt_cht_es8316_quirk_table[] = { static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; static const char * const mic_name[] = { "in1", "in2" }; + struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); struct property_entry props[MAX_NO_PROPS] = {}; struct byt_cht_es8316_private *priv; const struct dmi_system_id *dmi_id; - struct device *dev = &pdev->dev; - struct snd_soc_acpi_mach *mach; struct fwnode_handle *fwnode; const char *platform_name; struct acpi_device *adev; @@ -476,7 +476,6 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; - mach = dev->platform_data; /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_cht_es8316_dais); i++) { if (!strcmp(byt_cht_es8316_dais[i].codecs->name, -- 2.33.0

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[PATCH] ASoC: soc-pcm: restore mixer functionality
by Pierre-Louis Bossart 07 Oct '21

07 Oct '21

Recent changes in soc-pcm completely broke basic support for mixers on Intel systems: the filters on BE states prevent the connection of a second mixer input while the back-end is already active. Rather than reverting the changes, which would be problematic for Tegra systems, this patch suggests an additional filter which will only apply to Tegra systems. This is a temporary solution which will have to be revisited - additional issues have been reported with DPCM. Fixes: 0c25db3f7621 ('ASoC: soc-pcm: Don't reconnect an already active BE') Suggested-by: Sameer Pujar <spujar(a)nvidia.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/soc-pcm.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b5825d8d6caa..0aa0ae9703d1 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1395,7 +1395,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; - if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_NEW) && + /* + * Filter for systems with 'component_chaining' enabled. + * This helps to avoid unnecessary re-configuration of an + * already active BE on such systems. + */ + if (fe->card->component_chaining && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_NEW) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_CLOSE)) continue; -- 2.25.1

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[PATCH] ASoC: codec: wcd938x: Add irq config support
by Srinivasa Rao Mandadapu 07 Oct '21

07 Oct '21

This patch fixes compilation error in wcd98x codec driver. Fixes: 045442228868 ("ASoC: codecs: wcd938x: add audio routing and Kconfig") Signed-off-by: Venkata Prasad Potturu <potturu(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82ee233..216cea0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1583,6 +1583,7 @@ config SND_SOC_WCD938X_SDW tristate "WCD9380/WCD9385 Codec - SDW" select SND_SOC_WCD938X select SND_SOC_WCD_MBHC + select REGMAP_IRQ depends on SOUNDWIRE select REGMAP_SOUNDWIRE help -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH v2 0/3] Multiple headphone codec driver support
by Brent Lu 07 Oct '21

07 Oct '21

Support multiple headphone drivers in same machine driver. In this case, both rt5682 and rt5682s are supported and enumerated by different ACPI HID "10EC5682" and "RTL5682". V2 Changes: - remove useless 'NULL', 'false' in if-condition - can use 'comp_ids' field alone to enumerate driver - add comma to the end of entry in structure initialization - keep the table of byt/cht/cml/icl untouched Brent Lu (3): ASoC: soc-acpi: add comp_ids field for machine driver matching ASoC: Intel: sof_rt5682: detect codec variant in probe function ASoC: Intel: sof_rt5682: use comp_ids to enumerate rt5682s include/sound/soc-acpi.h | 2 ++ sound/soc/intel/boards/sof_rt5682.c | 34 +++--------------- .../intel/common/soc-acpi-intel-adl-match.c | 11 ++++-- .../intel/common/soc-acpi-intel-jsl-match.c | 35 +++++-------------- .../intel/common/soc-acpi-intel-tgl-match.c | 11 ++++-- sound/soc/soc-acpi.c | 24 +++++++++++-- 6 files changed, 52 insertions(+), 65 deletions(-) -- 2.25.1

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[PATCH v2 00/13] Extend AHUB audio support for Tegra210 and later
by Sameer Pujar 07 Oct '21

07 Oct '21

Earlier as part of series [0], support for ADMAIF and I/O modules (such as I2S, DMIC and DSPK) was added. This series aims at exposing some of the AHUB internal modules (listed below), which can be used for audio pre or post processing. * SFC (Sampling Frequency Converter) * MVC (Master Volume Control) * AMX (Audio Multiplexer) * ADX (Audio Demultiplexer) * Mixer These modules can be plugged into audio paths and relevant processing can be done. The MUX routes are extended to allow add or remove above modules in the path via mixer controls. This is similar to how specific ADMAIF channels are connected to relevant I/O module instances at the moment. Some of these modules can alter PCM parameters. Consider example of resampler (44.1 -> 48 kHz) in the path. aplay(44.1 kHz) -> ADMAIF -> SFC -> (48 kHz) I2S -> (48kHz) Codec The modules following SFC should be using converted sample rate and DAIs need to be configured accordingly. The audio-graph driver provides a mechanism to fixup the new parameters which can be specified in DT for a given DAI. Then core uses these new values via fixup callback and then pass it to respective DAIs hw_param() callback. The "convert-rate", described in [1], property can be used when there is rate conversion in the audio path. Similarly "convert-channels" can be used when there is channel conversion in the path. There is no "convert-xxx" property for sample size conversions. It can be added if necessary. [0] https://www.lkml.org/lkml/2020/7/21/1357 [1] Documentation/devicetree/bindings/sound/audio-graph-port.yaml Changelog ========= v1 -> v2 -------- * Put comments for soft reset application in the drivers. * Split out mute/volume control logic in put() calls of MVC driver and use separate callbacks for the respective kcontrols. * Update kcontrol put() callback in MVC driver to return 1 whenever there is change. Similar change is done in other drivers too. * Use name-prefix.yaml reference for the driver documentation now. * Add sound-name-prefix pattern for MIXER driver and use prefix accordingly in DT. Sameer Pujar (13): ASoC: soc-pcm: Don't reconnect an already active BE ASoC: simple-card-utils: Increase maximum DAI links limit to 512 ASoC: audio-graph: Fixup CPU endpoint hw_params in a BE<->BE link ASoC: dt-bindings: tegra: Few more Tegra210 AHUB modules ASoC: tegra: Add routes for few AHUB modules ASoC: tegra: Add Tegra210 based MVC driver ASoC: tegra: Add Tegra210 based SFC driver ASoC: tegra: Add Tegra210 based AMX driver ASoC: tegra: Add Tegra210 based ADX driver ASoC: tegra: Add Tegra210 based Mixer driver arm64: defconfig: Enable few Tegra210 based AHUB drivers arm64: tegra: Add few AHUB devices for Tegra210 and later arm64: tegra: Extend APE audio support on Jetson platforms .../bindings/sound/nvidia,tegra210-adx.yaml | 76 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 20 + .../bindings/sound/nvidia,tegra210-amx.yaml | 76 + .../bindings/sound/nvidia,tegra210-mixer.yaml | 74 + .../bindings/sound/nvidia,tegra210-mvc.yaml | 76 + .../bindings/sound/nvidia,tegra210-sfc.yaml | 73 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 1554 ++++++++- arch/arm64/boot/dts/nvidia/tegra186.dtsi | 120 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 1493 +++++++- .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 1520 ++++++++- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 116 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 876 +++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 876 +++++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 77 + arch/arm64/configs/defconfig | 5 + include/sound/simple_card_utils.h | 2 +- sound/soc/generic/audio-graph-card.c | 4 +- sound/soc/soc-pcm.c | 4 + sound/soc/tegra/Kconfig | 48 + sound/soc/tegra/Makefile | 10 + sound/soc/tegra/tegra210_adx.c | 531 +++ sound/soc/tegra/tegra210_adx.h | 72 + sound/soc/tegra/tegra210_ahub.c | 511 ++- sound/soc/tegra/tegra210_amx.c | 600 ++++ sound/soc/tegra/tegra210_amx.h | 93 + sound/soc/tegra/tegra210_mixer.c | 674 ++++ sound/soc/tegra/tegra210_mixer.h | 100 + sound/soc/tegra/tegra210_mvc.c | 645 ++++ sound/soc/tegra/tegra210_mvc.h | 117 + sound/soc/tegra/tegra210_sfc.c | 3549 ++++++++++++++++++++ sound/soc/tegra/tegra210_sfc.h | 78 + 31 files changed, 13647 insertions(+), 423 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-amx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml create mode 100644 sound/soc/tegra/tegra210_adx.c create mode 100644 sound/soc/tegra/tegra210_adx.h create mode 100644 sound/soc/tegra/tegra210_amx.c create mode 100644 sound/soc/tegra/tegra210_amx.h create mode 100644 sound/soc/tegra/tegra210_mixer.c create mode 100644 sound/soc/tegra/tegra210_mixer.h create mode 100644 sound/soc/tegra/tegra210_mvc.c create mode 100644 sound/soc/tegra/tegra210_mvc.h create mode 100644 sound/soc/tegra/tegra210_sfc.c create mode 100644 sound/soc/tegra/tegra210_sfc.h -- 2.7.4

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[PATCH 0/3] Multiple headphone codec driver support
by Brent Lu 07 Oct '21

07 Oct '21

Support multiple headphone drivers in same machine driver. In this case, both rt5682 and rt5682s are supported and enumerated by different ACPI HID "10EC5682" and "RTL5682". Brent Lu (3): ASoC: soc-acpi: add alternative id field for machine driver matching ASoC: Intel: sof_rt5682: detect codec variant in probe function ASoC: Intel: sof_rt5682: use id_alt to enumerate rt5682s include/sound/soc-acpi.h | 2 ++ sound/soc/intel/boards/sof_rt5682.c | 34 +++---------------- .../intel/common/soc-acpi-intel-adl-match.c | 8 +++++ .../intel/common/soc-acpi-intel-byt-match.c | 6 ++++ .../intel/common/soc-acpi-intel-cht-match.c | 6 ++++ .../intel/common/soc-acpi-intel-cml-match.c | 8 +++++ .../intel/common/soc-acpi-intel-icl-match.c | 6 ++++ .../intel/common/soc-acpi-intel-jsl-match.c | 32 +++++------------ .../intel/common/soc-acpi-intel-tgl-match.c | 8 +++++ sound/soc/soc-acpi.c | 21 +++++++++++- 10 files changed, 76 insertions(+), 55 deletions(-) -- 2.25.1

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DAPM PIN switches do not update in alsamixer when changed through UCM profile
by Hans de Goede 07 Oct '21

07 Oct '21

Hi All, I notice that DAPM PIN switches, such as e.g. the "Headphone" SOC_DAPM_PIN_SWITCH defined in: sound/soc/intel/boards/cht_bsw_nau8824.c: static const struct snd_kcontrol_new cht_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Ext Spk"), }; Do not get updated to reflect state-changes when the output is switched between e.g. Headphone / "Ext Spk" by pulseaudio/pipewire through the UCM profile mechanism. If I exit alsa-mixer after changing the output and start it again then the control does show the expect value. So it seems that we are failing to send a change event about this somewhere? Regards, Hans

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