April 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH v2 RESEND 0/2] Make headphone work on Huawei Matebook D15
by Mauro Carvalho Chehab 05 Apr '22

05 Apr '22

(resent, as the From/SoB at the patches were using my past e-mail. No changes here, except by having Hans ack and have the From/SoB fixed) At Huawei Matebook D15 two different GPIOs are used to control the output: - gpio0 controls the speaker output; - gpio1 controls the headphone output. Changing both at the same time cause spurious events that are mis-interpreted as input events, causing troubles on apps. So, a delay is needed before turning on such gpios. With this patch, plugging a headphone causes a jack event to trigger the speaker supply, powering down the speaker and powering up the headphone output. Removing the headphone also triggers the power supply, powering up the speaker and powering down the headphone. Mauro Carvalho Chehab (2): ASoC: Intel: sof_es8336: support a separate gpio to control headphone ASoC: Intel: sof_es8336: Huawei Matebook D15 uses a headphone gpio sound/soc/intel/boards/sof_es8336.c | 68 ++++++++++++++++++++++++----- 1 file changed, 58 insertions(+), 10 deletions(-) -- 2.35.1

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[PATCH] ASoC: codecs: add support for ES8326 dts
by Zhu Ning 05 Apr '22

05 Apr '22

From: yangxiaohua <yangxiaohua(a)everest-semi.com> The ES8326 codec is not compatible with ES8316 and requires a dedicated driver. This is a separate patch for ES8326 dts Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> --- .../bindings/sound/everest,es8326.yaml | 109 ++++++++++++++++++ 1 file changed, 109 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/everest,es8326.yaml diff --git a/Documentation/devicetree/bindings/sound/everest,es8326.yaml b/Documentation/devicetree/bindings/sound/everest,es8326.yaml new file mode 100755 index 000000000000..8cae9c328dbc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8326.yaml @@ -0,0 +1,109 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8326.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8326 audio CODEC + +maintainers: + - David Yang <yangxiaohua(a)everest-semi.com> + +properties: + compatible: + const: everest,es8326 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + + mclk-rate: + description: + clock rate of mclk + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + + everest,jack-pol: + description: + just the value of reg 57. Bit(3) decides whether the jack polarity is inverted. + Bit(2) decides whether the bottom on the headset is inverted. + Bit(1)/(0) decides the mic properity to be OMTP/CTIA or auto. + $ref: /schemas/types.yaml#/definitions/uint8-array + minimum: 0 + maximum: 0x0f + default: 0x0f + + everest,mic1-src: + description: + the value of reg 2A when headset plugged. + $ref: /schemas/types.yaml#/definitions/uint8-array + minimum: 0x00 + maximum: 0x77 + default: 0x22 + + everest,mic2-src: + description: + the value of reg 2A when headset unplugged. + $ref: /schemas/types.yaml#/definitions/uint8-array + minimum: 0x00 + maximum: 0x77 + default: 0x44 + + everest,amic1-src: + description: + the value of reg 23 when headset plugged. + $ref: /schemas/types.yaml#/definitions/uint8-array + minimum: 0x19 + maximum: 0x29 + default: 0x19 + + everest,amic2-src: + description: + the value of reg 23 when headset unplugged. + Set to 0x19 if dmic used. Set to 0x29 if amic used. + $ref: /schemas/types.yaml#/definitions/uint8-array + minimum: 0x19 + maximum: 0x29 + default: 0x19 + + everest,jack-detect-inverted: + description: + Defined to invert the jack detection. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - reg + - "#sound-dai-cells" + + +additionalProperties: false + +examples: + - | + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + es8326: codec@19 { + compatible = "everest,es8326"; + reg = <0x19>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + mic1-src = [22]; + mic2-src = [44]; + amic1-src = [19]; + amic2-src = [19]; + jack-pol = [0e]; + }; + }; -- 2.17.1

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[PATCH] ALSA: emu10k1: Stop using iommu_present()
by Robin Murphy 05 Apr '22

05 Apr '22

iommu_get_domain_for_dev() is already perfectly happy to return NULL if the given device has no IOMMU. Drop the unnecessary check. Signed-off-by: Robin Murphy <robin.murphy(a)arm.com> --- sound/pci/emu10k1/emu10k1_main.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 86cc1ca025e4..5ffab343b89c 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1751,9 +1751,6 @@ static void snd_emu10k1_detect_iommu(struct snd_emu10k1 *emu) emu->iommu_workaround = false; - if (!iommu_present(emu->card->dev->bus)) - return; - domain = iommu_get_domain_for_dev(emu->card->dev); if (domain && domain->type == IOMMU_DOMAIN_IDENTITY) return; -- 2.28.0.dirty

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[PATCH] ASoC: Intel: catpt: Drop redundant enum constant
by Cezary Rojewski 05 Apr '22

05 Apr '22

CATPT_SSP_IFACE_LAST is being used only to calculate CATPT_SSP_COUNT. Make CATPT_SSP_COUNT part of the enum directly and remove the redundant constant. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/catpt/messages.h | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/catpt/messages.h b/sound/soc/intel/catpt/messages.h index 978a20b3f471..c17e948d9f52 100644 --- a/sound/soc/intel/catpt/messages.h +++ b/sound/soc/intel/catpt/messages.h @@ -219,11 +219,9 @@ int catpt_ipc_free_stream(struct catpt_dev *cdev, u8 stream_hw_id); enum catpt_ssp_iface { CATPT_SSP_IFACE_0 = 0, CATPT_SSP_IFACE_1 = 1, - CATPT_SSP_IFACE_LAST = CATPT_SSP_IFACE_1, + CATPT_SSP_COUNT, }; -#define CATPT_SSP_COUNT (CATPT_SSP_IFACE_LAST + 1) - enum catpt_mclk_frequency { CATPT_MCLK_OFF = 0, CATPT_MCLK_FREQ_6_MHZ = 1, -- 2.25.1

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[v2] ASoC: qcom: Add driver support for ALC5682I-VS
by Judy Hsiao 05 Apr '22

05 Apr '22

Add Machine driver support for ALC5682I-VS codec. Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- Changes from v1[1]: - Fix nits and use SND_SOC_DAIFMT_CBC_CFS. - Fix return value when snd_soc_dai_set_sysclk failed. [1]: https://patchwork.kernel.org/project/alsa-devel/patch/20220330084519.386385… sound/soc/qcom/Kconfig | 2 + sound/soc/qcom/sc7280.c | 107 ++++++++++++++++++++++++++++++++++++++++ 2 files changed, 109 insertions(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 28d0dfb4033c..750653404ba3 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -197,6 +197,8 @@ config SND_SOC_SC7280 select SND_SOC_LPASS_MACRO_COMMON imply SND_SOC_LPASS_RX_MACRO imply SND_SOC_LPASS_TX_MACRO + select SND_SOC_RT5682_I2C + select SND_SOC_RT5682S help Add support for audio on Qualcomm Technologies Inc. SC7280 SoC-based systems. diff --git a/sound/soc/qcom/sc7280.c b/sound/soc/qcom/sc7280.c index bd0bf9c8cb28..8c0bbae490dd 100644 --- a/sound/soc/qcom/sc7280.c +++ b/sound/soc/qcom/sc7280.c @@ -12,14 +12,21 @@ #include <sound/jack.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/rt5682s.h> #include <linux/soundwire/sdw.h> +#include "../codecs/rt5682.h" +#include "../codecs/rt5682s.h" #include "common.h" #include "lpass.h" +#define DEFAULT_MCLK_RATE 19200000 +#define RT5682_PLL1_FREQ (48000 * 512) + struct sc7280_snd_data { struct snd_soc_card card; struct sdw_stream_runtime *sruntime[LPASS_MAX_PORTS]; + u32 pri_mi2s_clk_count; struct snd_soc_jack hs_jack; struct snd_soc_jack hdmi_jack; bool jack_setup; @@ -69,6 +76,7 @@ static int sc7280_headset_init(struct snd_soc_pcm_runtime *rtd) pdata->jack_setup = true; } switch (cpu_dai->id) { + case MI2S_PRIMARY: case LPASS_CDC_DMA_RX0: case LPASS_CDC_DMA_TX3: for_each_rtd_codec_dais(rtd, i, codec_dai) { @@ -110,11 +118,51 @@ static int sc7280_hdmi_init(struct snd_soc_pcm_runtime *rtd) return snd_soc_component_set_jack(component, &pdata->hdmi_jack, NULL); } +static int sc7280_rt5682_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_card *card = rtd->card; + struct sc7280_snd_data *data = snd_soc_card_get_drvdata(card); + int ret; + + if (++data->pri_mi2s_clk_count == 1) { + snd_soc_dai_set_sysclk(cpu_dai, + LPASS_MCLK0, + DEFAULT_MCLK_RATE, + SNDRV_PCM_STREAM_PLAYBACK); + } + snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_CBC_CFC | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_I2S); + + ret = snd_soc_dai_set_pll(codec_dai, RT5682S_PLL1, RT5682S_PLL_S_BCLK1, + 1536000, RT5682_PLL1_FREQ); + if (ret) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682S_SCLK_S_PLL1, + RT5682_PLL1_FREQ, + SND_SOC_CLOCK_IN); + + if (ret) { + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", + ret); + return ret; + } + + return 0; +} + static int sc7280_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); switch (cpu_dai->id) { + case MI2S_PRIMARY: case LPASS_CDC_DMA_TX3: return sc7280_headset_init(rtd); case LPASS_CDC_DMA_RX0: @@ -227,10 +275,66 @@ static int sc7280_snd_hw_free(struct snd_pcm_substream *substream) return 0; } +static void sc7280_snd_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct sc7280_snd_data *data = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + + switch (cpu_dai->id) { + case MI2S_PRIMARY: + if (--data->pri_mi2s_clk_count == 0) { + snd_soc_dai_set_sysclk(cpu_dai, + LPASS_MCLK0, + 0, + SNDRV_PCM_STREAM_PLAYBACK); + } + break; + case MI2S_SECONDARY: + break; + case LPASS_DP_RX: + break; + default: + dev_err(rtd->dev, "%s: invalid dai id 0x%x\n", __func__, + cpu_dai->id); + break; + } +} + +static int sc7280_snd_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + int ret = 0; + + switch (cpu_dai->id) { + case MI2S_PRIMARY: + ret = sc7280_rt5682_init(rtd); + break; + case MI2S_SECONDARY: + break; + case LPASS_DP_RX: + break; + default: + dev_err(rtd->dev, "%s: invalid dai id 0x%x\n", __func__, + cpu_dai->id); + return -EINVAL; + } + return ret; +} + static const struct snd_soc_ops sc7280_ops = { + .startup = sc7280_snd_startup, .hw_params = sc7280_snd_hw_params, .hw_free = sc7280_snd_hw_free, .prepare = sc7280_snd_prepare, + .shutdown = sc7280_snd_shutdown, +}; + +static const struct snd_soc_dapm_widget sc7280_snd_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), }; static int sc7280_snd_platform_probe(struct platform_device *pdev) @@ -252,6 +356,9 @@ static int sc7280_snd_platform_probe(struct platform_device *pdev) card->driver_name = "SC7280"; card->dev = dev; + card->dapm_widgets = sc7280_snd_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sc7280_snd_widgets); + ret = qcom_snd_parse_of(card); if (ret) return ret; -- 2.35.1.1021.g381101b075-goog

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[PATCH] ASoC: topology: Correct error handling in soc_tplg_dapm_widget_create()
by Peter Ujfalusi 05 Apr '22

05 Apr '22

Academic correction of error handling: In case the allocation of kc or kcontrol_type fails the correct label to jump is hdr_err since the template.sname has been also allocated at this point. Fixes: d29d41e28eea6 ("ASoC: topology: Add support for multiple kcontrol types to a widget") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/soc-topology.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 72e50df7052c..3bb90a819650 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1436,12 +1436,12 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, template.num_kcontrols = le32_to_cpu(w->num_kcontrols); kc = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(*kc), GFP_KERNEL); if (!kc) - goto err; + goto hdr_err; kcontrol_type = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(unsigned int), GFP_KERNEL); if (!kcontrol_type) - goto err; + goto hdr_err; for (i = 0; i < le32_to_cpu(w->num_kcontrols); i++) { control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; -- 2.35.1

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[PATCH] ASoC: SOF: ipc3-topology: Set scontrol->priv to NULL after freeing it
by Peter Ujfalusi 05 Apr '22

05 Apr '22

Since the scontrol->priv is freed up during load operation it should be set to NULL to be safe against double freeing attempt. Fixes: b5cee8feb1d48 ("ASoC: SOF: topology: Make control parsing IPC agnostic") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/ipc3-topology.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 2f8450a8c0a1..2e1c76ed087d 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -1592,6 +1592,7 @@ static int sof_ipc3_control_load_bytes(struct snd_sof_dev *sdev, struct snd_sof_ if (scontrol->priv_size > 0) { memcpy(cdata->data, scontrol->priv, scontrol->priv_size); kfree(scontrol->priv); + scontrol->priv = NULL; if (cdata->data->magic != SOF_ABI_MAGIC) { dev_err(sdev->dev, "Wrong ABI magic 0x%08x.\n", cdata->data->magic); -- 2.35.1

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[PATCH] ASoC: SOF: topology: Fix memory leak of scontrol->name
by Peter Ujfalusi 05 Apr '22

05 Apr '22

The scontrol->name is allocated with kstrdup, it must be freed before the scontrol is freed to avoid leaking memory. The constant leaking happens via sof_widget_unload() path on every module removal. Fixes: b5cee8feb1d48 ("ASoC: SOF: topology: Make control parsing IPC agnostic") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/topology.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 9b11e9795a7a..75d78f9178a3 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -941,11 +941,13 @@ static int sof_control_load(struct snd_soc_component *scomp, int index, default: dev_warn(scomp->dev, "control type not supported %d:%d:%d\n", hdr->ops.get, hdr->ops.put, hdr->ops.info); + kfree(scontrol->name); kfree(scontrol); return 0; } if (ret < 0) { + kfree(scontrol->name); kfree(scontrol); return ret; } @@ -1380,6 +1382,7 @@ static int sof_widget_unload(struct snd_soc_component *scomp, } kfree(scontrol->ipc_control_data); list_del(&scontrol->list); + kfree(scontrol->name); kfree(scontrol); } -- 2.35.1

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[PATCH 00/11] ASoC: SOF: Miscellaneous fixes for IPC and trace
by Ranjani Sridharan 05 Apr '22

05 Apr '22

This series includes some miscellaenous fixes pertaining to IPC, DMA buffer and DMA trace. Libin Yang (2): ASoC: SOF: extend the interface to stop DMA trace ASoC: SOF: disable dma trace in s0ix Peter Ujfalusi (8): ASoC: SOF: Intel: hda-loader: Use local snd_dma_buffer ASoC: SOF: Remove dmab and dmab_bdl from snd_sof_dev struct ASoC: SOF: ipc: Add max_payload_size field in struct snd_sof_ipc ASoC: SOF: ipc: Use ipc->max_payload_size for message/reply size checking ASoC: SOF: Drop support for mapped control data ASoC: SOF: ipc: Use msg->reply_data directly in snd_sof_ipc_get_reply() ASoC: SOF: Drop 'header' parameter from tx_message() API ASoC: SOF: Remove header from struct snd_sof_ipc_msg tangmeng (1): ASoC: SOF: Remove redundant return statements sound/soc/sof/compress.c | 8 ++-- sound/soc/sof/debug.c | 2 +- sound/soc/sof/intel/hda-dsp.c | 8 ++-- sound/soc/sof/intel/hda-loader.c | 12 ++--- sound/soc/sof/intel/tgl.c | 6 +-- sound/soc/sof/ipc.c | 76 ++++++++++++-------------------- sound/soc/sof/ipc3-pcm.c | 8 ++-- sound/soc/sof/ipc3-topology.c | 25 +++++------ sound/soc/sof/ipc3.c | 4 +- sound/soc/sof/pcm.c | 5 +-- sound/soc/sof/pm.c | 20 ++++++--- sound/soc/sof/sof-audio.c | 3 -- sound/soc/sof/sof-audio.h | 1 - sound/soc/sof/sof-client.c | 2 +- sound/soc/sof/sof-priv.h | 26 ++++++----- sound/soc/sof/trace.c | 61 ++++++++++++++++--------- 16 files changed, 131 insertions(+), 136 deletions(-) -- 2.25.1

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[PATCH v1 0/7] ASoC: wm8731: Overhaul of the driver
by Mark Brown 05 Apr '22

05 Apr '22

This series contains a bunch of fairly minor changes which overhaul and modernise the WM8731 driver, there should be no impact on the driver's functionality. Mark Brown (7): ASoC: wm8731: Update to modern DAI terminology ASoC: wm8731: Factor component init out of bus code ASoC: wm8731: Move regulator request into wm8731_init() ASoC: wm8731: Factor our MCLK and mutex initialisation ASoC: wm8731: Factor out the I2C and SPI bus code into separate modules ASoC: wm8731: Convert DT bindings to YAML format ARM: configs: Update multi_v5_defconfig for WM8731 bus refactoring .../devicetree/bindings/sound/wlf,wm8731.yaml | 97 +++++++ .../devicetree/bindings/sound/wm8731.txt | 27 -- arch/arm/configs/multi_v5_defconfig | 3 +- sound/soc/atmel/Kconfig | 4 +- sound/soc/au1x/Kconfig | 2 +- sound/soc/codecs/Kconfig | 18 +- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wm8731-i2c.c | 69 +++++ sound/soc/codecs/wm8731-spi.c | 59 ++++ sound/soc/codecs/wm8731.c | 266 +++--------------- sound/soc/codecs/wm8731.h | 27 ++ sound/soc/pxa/Kconfig | 4 +- 12 files changed, 323 insertions(+), 257 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8731.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8731.txt create mode 100644 sound/soc/codecs/wm8731-i2c.c create mode 100644 sound/soc/codecs/wm8731-spi.c base-commit: 0b88a659002151e354272a13ae29d8b795ef1b46 -- 2.30.2

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