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[RESEND PATCH] ASoC: dt-bindings: qcom,lpass-rx-macro: correct minItems for clocks
by Krzysztof Kozlowski 30 Mar '23

30 Mar '23

The RX macro codec comes on some platforms in two variants - ADSP and ADSP bypassed - thus the clock-names varies from 3 to 5. The clocks must vary as well: sc7280-idp.dtb: codec@3200000: clocks: [[202, 8], [202, 7], [203]] is too short Fixes: 852fda58d99a ("ASoC: qcom: dt-bindings: Update bindings for clocks in lpass digital codes") Cc: <stable(a)vger.kernel.org> Acked-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml index f8972769cc6a..ec4b0ac8ad68 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml @@ -28,6 +28,7 @@ properties: const: 0 clocks: + minItems: 3 maxItems: 5 clock-names: -- 2.34.1

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[PATCH v4] ASoC: dt-bindings: alc5632: Convert to dtschema
by Saalim Quadri 30 Mar '23

30 Mar '23

Convert the ALC5632 audio codec bindings to DT schema. Signed-off-by: Saalim Quadri <danascape(a)gmail.com> --- Changes: V1 -> V2: Use correct node name V2 -> V3: Add correct maintainer name V3 -> V4: Drop alc5632 label name .../devicetree/bindings/sound/alc5632.txt | 43 ------------- .../bindings/sound/realtek,alc5632.yaml | 63 +++++++++++++++++++ 2 files changed, 63 insertions(+), 43 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/alc5632.txt create mode 100644 Documentation/devicetree/bindings/sound/realtek,alc5632.yaml diff --git a/Documentation/devicetree/bindings/sound/alc5632.txt b/Documentation/devicetree/bindings/sound/alc5632.txt deleted file mode 100644 index ffd886d110bd..000000000000 --- a/Documentation/devicetree/bindings/sound/alc5632.txt +++ /dev/null @@ -1,43 +0,0 @@ -ALC5632 audio CODEC - -This device supports I2C only. - -Required properties: - - - compatible : "realtek,alc5632" - - - reg : the I2C address of the device. - - - gpio-controller : Indicates this device is a GPIO controller. - - - #gpio-cells : Should be two. The first cell is the pin number and the - second cell is used to specify optional parameters (currently unused). - -Pins on the device (for linking into audio routes): - - * SPK_OUTP - * SPK_OUTN - * HP_OUT_L - * HP_OUT_R - * AUX_OUT_P - * AUX_OUT_N - * LINE_IN_L - * LINE_IN_R - * PHONE_P - * PHONE_N - * MIC1_P - * MIC1_N - * MIC2_P - * MIC2_N - * MICBIAS1 - * DMICDAT - -Example: - -alc5632: alc5632@1e { - compatible = "realtek,alc5632"; - reg = <0x1a>; - - gpio-controller; - #gpio-cells = <2>; -}; diff --git a/Documentation/devicetree/bindings/sound/realtek,alc5632.yaml b/Documentation/devicetree/bindings/sound/realtek,alc5632.yaml new file mode 100644 index 000000000000..fb05988ff7ea --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,alc5632.yaml @@ -0,0 +1,63 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,alc5632.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ALC5632 audio CODEC + +description: | + Pins on the device (for linking into audio routes): + * SPK_OUTP + * SPK_OUTN + * HP_OUT_L + * HP_OUT_R + * AUX_OUT_P + * AUX_OUT_N + * LINE_IN_L + * LINE_IN_R + * PHONE_P + * PHONE_N + * MIC1_P + * MIC1_N + * MIC2_P + * MIC2_N + * MICBIAS1 + * DMICDAT + +maintainers: + - Leon Romanovsky <leon(a)leon.nu> + +properties: + compatible: + const: realtek,alc5632 + + reg: + maxItems: 1 + + '#gpio-cells': + const: 2 + + gpio-controller: true + +required: + - compatible + - reg + - '#gpio-cells' + - gpio-controller + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "realtek,alc5632"; + reg = <0x1a>; + gpio-controller; + #gpio-cells = <2>; + }; + }; -- 2.34.1

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[PATCH] ASoC: dapm: Sort speakers after other outputs
by Mark Brown 30 Mar '23

30 Mar '23

Currently we sequence speakers with line and headphone outputs in DAPM. This works well when speakers are integrate into a CODEC but when there is an external speaker driver connected to a line or headphone output it can mean that the speaker driver ends up getting sequenced such that it picks up pops and clicks from the CODEC. Mask this by moving speakers after the other outputs in DAPM. We may want to consider doing this for headphones too but separate drivers are less common there and headphone drivers often also function as line outputs so the situation is less clear. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 74 ++++++++++++++++++++++++++-------------------------- 1 file changed, 37 insertions(+), 37 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5d9a671e50f1..34fdcb7ee079 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -99,52 +99,52 @@ static int dapm_up_seq[] = { [snd_soc_dapm_adc] = 11, [snd_soc_dapm_out_drv] = 12, [snd_soc_dapm_hp] = 12, - [snd_soc_dapm_spk] = 12, [snd_soc_dapm_line] = 12, [snd_soc_dapm_sink] = 12, - [snd_soc_dapm_kcontrol] = 13, - [snd_soc_dapm_post] = 14, + [snd_soc_dapm_spk] = 13, + [snd_soc_dapm_kcontrol] = 14, + [snd_soc_dapm_post] = 15, }; static int dapm_down_seq[] = { [snd_soc_dapm_pre] = 1, [snd_soc_dapm_kcontrol] = 2, [snd_soc_dapm_adc] = 3, - [snd_soc_dapm_hp] = 4, [snd_soc_dapm_spk] = 4, - [snd_soc_dapm_line] = 4, - [snd_soc_dapm_out_drv] = 4, - [snd_soc_dapm_sink] = 4, - [snd_soc_dapm_pga] = 5, - [snd_soc_dapm_buffer] = 5, - [snd_soc_dapm_scheduler] = 5, - [snd_soc_dapm_effect] = 5, - [snd_soc_dapm_src] = 5, - [snd_soc_dapm_asrc] = 5, - [snd_soc_dapm_encoder] = 5, - [snd_soc_dapm_decoder] = 5, - [snd_soc_dapm_switch] = 6, - [snd_soc_dapm_mixer_named_ctl] = 6, - [snd_soc_dapm_mixer] = 6, - [snd_soc_dapm_dac] = 7, - [snd_soc_dapm_mic] = 8, - [snd_soc_dapm_siggen] = 8, - [snd_soc_dapm_input] = 8, - [snd_soc_dapm_output] = 8, - [snd_soc_dapm_micbias] = 9, - [snd_soc_dapm_vmid] = 9, - [snd_soc_dapm_mux] = 10, - [snd_soc_dapm_demux] = 10, - [snd_soc_dapm_aif_in] = 11, - [snd_soc_dapm_aif_out] = 11, - [snd_soc_dapm_dai_in] = 11, - [snd_soc_dapm_dai_out] = 11, - [snd_soc_dapm_dai_link] = 12, - [snd_soc_dapm_supply] = 13, - [snd_soc_dapm_clock_supply] = 14, - [snd_soc_dapm_pinctrl] = 14, - [snd_soc_dapm_regulator_supply] = 14, - [snd_soc_dapm_post] = 15, + [snd_soc_dapm_hp] = 5, + [snd_soc_dapm_line] = 5, + [snd_soc_dapm_out_drv] = 5, + [snd_soc_dapm_sink] = 6, + [snd_soc_dapm_pga] = 6, + [snd_soc_dapm_buffer] = 6, + [snd_soc_dapm_scheduler] = 6, + [snd_soc_dapm_effect] = 6, + [snd_soc_dapm_src] = 6, + [snd_soc_dapm_asrc] = 6, + [snd_soc_dapm_encoder] = 6, + [snd_soc_dapm_decoder] = 6, + [snd_soc_dapm_switch] = 7, + [snd_soc_dapm_mixer_named_ctl] = 7, + [snd_soc_dapm_mixer] = 7, + [snd_soc_dapm_dac] = 8, + [snd_soc_dapm_mic] = 9, + [snd_soc_dapm_siggen] = 9, + [snd_soc_dapm_input] = 9, + [snd_soc_dapm_output] = 9, + [snd_soc_dapm_micbias] = 10, + [snd_soc_dapm_vmid] = 10, + [snd_soc_dapm_mux] = 11, + [snd_soc_dapm_demux] = 11, + [snd_soc_dapm_aif_in] = 12, + [snd_soc_dapm_aif_out] = 12, + [snd_soc_dapm_dai_in] = 12, + [snd_soc_dapm_dai_out] = 12, + [snd_soc_dapm_dai_link] = 13, + [snd_soc_dapm_supply] = 14, + [snd_soc_dapm_clock_supply] = 15, + [snd_soc_dapm_pinctrl] = 15, + [snd_soc_dapm_regulator_supply] = 15, + [snd_soc_dapm_post] = 16, }; static void dapm_assert_locked(struct snd_soc_dapm_context *dapm) --- base-commit: e8d018dd0257f744ca50a729e3d042cf2ec9da65 change-id: 20230324-asoc-dapm-spk-30a8bf9ff4f5 Best regards, -- Mark Brown <broonie(a)kernel.org>

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[PATCH 00/13] ASoC: SoundWire codecs: simplify/fix set_stream
by Bard Liao 30 Mar '23

30 Mar '23

Simplify set_stream in SoundWire codecs. Pierre-Louis Bossart (13): ASoC: codecs: cs42l42-sdw: clear stream ASoC: codecs: sdw-mockup: simplify set_stream ASoC: codecs: max98373-sdw: simplify set_stream ASoC: codecs: rt1308-sdw: simplify set_stream ASoC: codecs: rt1316-sdw: simplify set_stream ASoC: codecs: rt1318-sdw: simplify set_stream ASoC: codecs: rt700-sdw: simplify set_stream ASoC: codecs: rt711-sdw: simplify set_stream ASoC: codecs: rt711-sdca: simplify set_stream ASoC: codecs: rt712-sdca: simplify set_stream ASoC: codecs: rt715: simplify set_stream ASoC: codecs: rt715-sdca: simplify set_stream ASoC: codecs: rt5682-sdw: simplify set_stream sound/soc/codecs/cs42l42-sdw.c | 3 --- sound/soc/codecs/max98373-sdw.c | 34 +++++++-------------------------- sound/soc/codecs/rt1308-sdw.c | 30 +++++++---------------------- sound/soc/codecs/rt1308-sdw.h | 4 ---- sound/soc/codecs/rt1316-sdw.c | 30 +++++++---------------------- sound/soc/codecs/rt1316-sdw.h | 4 ---- sound/soc/codecs/rt1318-sdw.c | 30 +++++++---------------------- sound/soc/codecs/rt1318-sdw.h | 4 ---- sound/soc/codecs/rt5682-sdw.c | 34 +++++++-------------------------- sound/soc/codecs/rt700.c | 30 +++++++---------------------- sound/soc/codecs/rt700.h | 4 ---- sound/soc/codecs/rt711-sdca.c | 30 +++++++---------------------- sound/soc/codecs/rt711-sdca.h | 4 ---- sound/soc/codecs/rt711.c | 30 +++++++---------------------- sound/soc/codecs/rt711.h | 4 ---- sound/soc/codecs/rt712-sdca.c | 30 +++++++---------------------- sound/soc/codecs/rt712-sdca.h | 4 ---- sound/soc/codecs/rt715-sdca.c | 30 +++++++---------------------- sound/soc/codecs/rt715-sdca.h | 4 ---- sound/soc/codecs/rt715.c | 30 +++++++---------------------- sound/soc/codecs/rt715.h | 4 ---- sound/soc/codecs/sdw-mockup.c | 34 ++++++--------------------------- 22 files changed, 83 insertions(+), 328 deletions(-) -- 2.25.1

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No sound output of headphone on Senary SN6140
by GitHub issues - opened 30 Mar '23

30 Mar '23

alsa-project/alsa-ucm-conf issue #300 was opened from pyma1: Thinkpad L14 got a senary SN6140 codec, speaker, Dmic, headset mic works fine. Only no sound output from headphone. add snd-intel-dspcfg.dsp_driver=1 to kernel cmdline the issue is reproduced too. When not fully plugin headset, it didn't trigger headphone switch, sound output can be enabled in alsamixer. Disable pipewire or pulseaudio, the issue is the same as before. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/11108039… [alsaucm.txt](https://github.com/alsa-project/alsa-ucm-conf/files/11108047/a… Would be appreciated for your help. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/300 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Does this compile any code?
by GitHub issues - opened 29 Mar '23

29 Mar '23

alsa-project/alsa-lib issue #309 was opened from stefan11111: If I build the same kernel twice, I get this output when building it the second time: ``` # make CALL scripts/checksyscalls.sh DESCEND objtool Kernel: arch/x86/boot/bzImage is ready (#17) ``` If I build the kernel again, only changing SND_X86, I get this output: ``` # make SYNC include/config/auto.conf CALL scripts/checksyscalls.sh DESCEND objtool Kernel: arch/x86/boot/bzImage is ready (#17) ``` It doesn't look like that option compiles any code. Issue URL : https://github.com/alsa-project/alsa-lib/issues/309 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ASoC: soc-topology.c: dai_link->platform again
by Kuninori Morimoto 29 Mar '23

29 Mar '23

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> On SOF, many topology is assuming dai_link->platform exists, and is allowed to be overwritten on each link_load(). This patch restore the removed dai_link->platform for SOF, and add the comment. Fixes: e7098ba9b378 ("ASoC: soc-topology.c: remove unnecessary dai_link->platform") Reported-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/87jzz7jczp.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-topology.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index be9849749713..d0aca6b9058b 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1685,8 +1685,8 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, struct snd_soc_dai_link_component *dlc; int ret; - /* link + cpu + codec */ - link = devm_kzalloc(tplg->dev, sizeof(*link) + (2 * sizeof(*dlc)), GFP_KERNEL); + /* link + cpu + codec + platform */ + link = devm_kzalloc(tplg->dev, sizeof(*link) + (3 * sizeof(*dlc)), GFP_KERNEL); if (link == NULL) return -ENOMEM; @@ -1724,6 +1724,14 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, link->codecs->name = "snd-soc-dummy"; link->codecs->dai_name = "snd-soc-dummy-dai"; + /* + * Many topology is assuming link has Platform. + * This might be overwritten at soc_tplg_dai_link_load(). + */ + link->platforms = &dlc[2]; + link->platforms->name = "snd-soc-dummy"; + link->num_platforms = 1; + /* enable DPCM */ link->dynamic = 1; link->ignore_pmdown_time = 1; -- 2.25.1

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[PATCH for v6.3-rc] ASoC: SOF: avoid a NULL dereference with unsupported widgets
by Peter Ujfalusi 29 Mar '23

29 Mar '23

From: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> If an IPC4 topology contains an unsupported widget, its .module_info field won't be set, then sof_ipc4_route_setup() will cause a kernel Oops trying to dereference it. Add a check for such cases. Cc: stable(a)vger.kernel.org # 6.2 Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi Mark, This patch is generated on top of 6.3-rc4, it will have conflict with asoc-next because we have ChainDMA scheduled for 6.4 in there. I should have taken this patch a faster track, but missed it when arranging the patches, features. We noticed this when trying to use our development IPC4 topologies with mainline which does not yet able to handle the process module types (slated fro 6.4). IPC4 is still evolving so it is not rare that fw/tplg/kernel needs to be lock-stepped, but NULL pointer dereference should not happen. This is how the merge conflict resolution should end up between 6.3 and 6.4: int ret; /* no route set up if chain DMA is used */ if (src_pipeline->use_chain_dma || sink_pipeline->use_chain_dma) { if (!src_pipeline->use_chain_dma || !sink_pipeline->use_chain_dma) { dev_err(sdev->dev, "use_chain_dma must be set for both src %s and sink %s pipelines\n", src_widget->widget->name, sink_widget->widget->name); return -EINVAL; } return 0; } if (!src_fw_module || !sink_fw_module) { /* The NULL module will print as "(efault)" */ dev_err(sdev->dev, "source %s or sink %s widget weren't set up properly\n", src_fw_module->man4_module_entry.name, sink_fw_module->man4_module_entry.name); return -ENODEV; } sroute->src_queue_id = sof_ipc4_get_queue_id(src_widget, sink_widget, SOF_PIN_TYPE_SOURCE); Can you send this patch for 6.3 cycle? Thank you, Peter sound/soc/sof/ipc4-topology.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index a623707c8ffc..669b99a4f76e 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1805,6 +1805,14 @@ static int sof_ipc4_route_setup(struct snd_sof_dev *sdev, struct snd_sof_route * u32 header, extension; int ret; + if (!src_fw_module || !sink_fw_module) { + /* The NULL module will print as "(efault)" */ + dev_err(sdev->dev, "source %s or sink %s widget weren't set up properly\n", + src_fw_module->man4_module_entry.name, + sink_fw_module->man4_module_entry.name); + return -ENODEV; + } + sroute->src_queue_id = sof_ipc4_get_queue_id(src_widget, sink_widget, SOF_PIN_TYPE_SOURCE); if (sroute->src_queue_id < 0) { -- 2.40.0

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[PATCH] Asoc: wm_adsp: Add support for loading firmware with prefix name
by Jianhua Lu 29 Mar '23

29 Mar '23

For platform using fdt, system_name is NULL, it doesn't provide a way to load firmware with prefix name, so add it. Signed-off-by: Jianhua Lu <lujianhua000(a)gmail.com> --- sound/soc/codecs/wm_adsp.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 216120b68b64..17481e42d440 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -760,6 +760,10 @@ static int wm_adsp_request_firmware_file(struct wm_adsp *dsp, *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s.%s", dir, dsp->part, dsp->fwf_name, wm_adsp_fw[dsp->fw].file, system_name, filetype); + else if (asoc_component_prefix) + *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s.%s", dir, dsp->part, + dsp->fwf_name, wm_adsp_fw[dsp->fw].file, asoc_component_prefix, + filetype); else *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s.%s", dir, dsp->part, dsp->fwf_name, wm_adsp_fw[dsp->fw].file, filetype); @@ -831,6 +835,16 @@ static int wm_adsp_request_firmware_files(struct wm_adsp *dsp, NULL, "bin"); return 0; } + } else if (asoc_component_prefix) { + if (!wm_adsp_request_firmware_file(dsp, wmfw_firmware, wmfw_filename, + cirrus_dir, NULL, + asoc_component_prefix, "wmfw")) { + adsp_dbg(dsp, "Found '%s'\n", *wmfw_filename); + wm_adsp_request_firmware_file(dsp, coeff_firmware, coeff_filename, + cirrus_dir, NULL, + asoc_component_prefix, "bin"); + return 0; + } } if (!wm_adsp_request_firmware_file(dsp, wmfw_firmware, wmfw_filename, -- 2.39.2

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Re: [PATCH 3/4] ASoC: amd: acp: Add machine driver that enables sound for systems with a ES8336 codec
by Zhu Ning 29 Mar '23

29 Mar '23

> > The GPIO thing on Huawei Intel platform is too complicated, they use two GPIOs > > for headphone and speaker, and the headphone GPIO is inverted, which means low > > means on and high means off. Luckily there should be only one hardware config > > for the AMD acp3x platform GPIO and you may just choose the correct one. > > There being two different GPIOs sounds like it just allows the headphone > and speaker to be controlled separately - that seems more flexible, not > a problem? Yes it's called multi stream in Windows. However, extra GPIO causes more confusion in the driver. > > There is no such situation, and the system doesn't produce sound from speaker > > when headphones are plugged in. The user may manually open speaker using > > amixer sset 'Speaker' on or pavucontrol. > > Again, you're describing a specific configuration - someone might want > to do something different. Hi Marian Postevca you may want to separate the GPIO control by adding Headphone Power SND_SOC_DAPM_SUPPLY. You may also want to change the gpio handling function in the acp3x_es83xx_jack_events function. static const struct snd_soc_dapm_widget acp3x_es83xx_widgets[] = { SND_SOC_DAPM_SPK("Speaker", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_MIC("Internal Mic", NULL), + SND_SOC_DAPM_SUPPLY("Headphone Power", SND_SOC_NOPM, 0, 0, + acp3x_es83xx_headphone_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("Speaker Power", SND_SOC_NOPM, 0, 0, acp3x_es83xx_speaker_power_event, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), }; static const struct snd_soc_dapm_route acp3x_es83xx_audio_map[] = { {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, + {"Headphone", NULL, "Headphone Power"}, {"Speaker", NULL, "HPOL"}, {"Speaker", NULL, "HPOR"}, {"Speaker", NULL, "Speaker Power"}, }; +static int acp3x_es83xx_headphone_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct acp3x_es83xx_private *priv = get_mach_priv(w->dapm->card); + + dev_dbg(priv->codec_dev, "speaker power event: %d\n", event); + if (SND_SOC_DAPM_EVENT_ON(event)) + gpiod_set_value_cansleep(priv->gpio_headphone, true); + else + gpiod_set_value_cansleep(priv->gpio_headphone, false); + + return 0; +} + +static int acp3x_es83xx_speaker_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct acp3x_es83xx_private *priv = get_mach_priv(w->dapm->card); + + dev_dbg(priv->codec_dev, "speaker power event: %d\n", event); + if (SND_SOC_DAPM_EVENT_ON(event)) + gpiod_set_value_cansleep(priv->gpio_speakers, true); + else + gpiod_set_value_cansleep(priv->gpio_speakers, false); + + return 0; +}

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