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[PATCH AUTOSEL 5.15 05/41] ALSA: usb-audio: US16x08: Move overflow check before array access
by Sasha Levin 28 Jun '22

28 Jun '22

From: Daniil Dementev <d.dementev(a)ispras.ru> [ Upstream commit 3ddbe35d9a2ebd4924d458e0246b4ba6c13bb456 ] Buffer overflow could occur in the loop "while", due to accessing an array element before checking the index. Found by Linux Verification Center (linuxtesting.org) with SVACE. Signed-off-by: Daniil Dementev <d.dementev(a)ispras.ru> Reviewed-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> Link: https://lore.kernel.org/r/20220610165732.2904-1-d.dementev@ispras.ru Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_us16x08.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index b7b6f3834ed5..6eb7d93b358d 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -637,10 +637,10 @@ static int snd_get_meter_comp_index(struct snd_us16x08_meter_store *store) } } else { /* skip channels with no compressor active */ - while (!store->comp_store->val[ + while (store->comp_index <= SND_US16X08_MAX_CHANNELS + && !store->comp_store->val[ COMP_STORE_IDX(SND_US16X08_ID_COMP_SWITCH)] - [store->comp_index - 1] - && store->comp_index <= SND_US16X08_MAX_CHANNELS) { + [store->comp_index - 1]) { store->comp_index++; } ret = store->comp_index++; -- 2.35.1

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[PATCH AUTOSEL 5.18 19/53] ALSA: x86: intel_hdmi_audio: use pm_runtime_resume_and_get()
by Sasha Levin 28 Jun '22

28 Jun '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit bb30b453fedac277d66220431fd7063d9ddc10d8 ] The current code does not check for errors and does not release the reference on errors. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20220616222910.136854-3-pierre-louis.bossart@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/x86/intel_hdmi_audio.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 3b04c70a73e3..ab95fb34a635 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1068,7 +1068,9 @@ static int had_pcm_open(struct snd_pcm_substream *substream) intelhaddata = snd_pcm_substream_chip(substream); runtime = substream->runtime; - pm_runtime_get_sync(intelhaddata->dev); + retval = pm_runtime_resume_and_get(intelhaddata->dev); + if (retval < 0) + return retval; /* set the runtime hw parameter with local snd_pcm_hardware struct */ runtime->hw = had_pcm_hardware; @@ -1536,8 +1538,12 @@ static void had_audio_wq(struct work_struct *work) container_of(work, struct snd_intelhad, hdmi_audio_wq); struct intel_hdmi_lpe_audio_pdata *pdata = ctx->dev->platform_data; struct intel_hdmi_lpe_audio_port_pdata *ppdata = &pdata->port[ctx->port]; + int ret; + + ret = pm_runtime_resume_and_get(ctx->dev); + if (ret < 0) + return; - pm_runtime_get_sync(ctx->dev); mutex_lock(&ctx->mutex); if (ppdata->pipe < 0) { dev_dbg(ctx->dev, "%s: Event: HAD_NOTIFY_HOT_UNPLUG : port = %d\n", -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 18/53] ALSA: x86: intel_hdmi_audio: enable pm_runtime and set autosuspend delay
by Sasha Levin 28 Jun '22

28 Jun '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit e87c65aeb46ca4f5b7dc08531200bcb8a426c62e ] The existing code uses pm_runtime_get_sync/put_autosuspend, but pm_runtime was not explicitly enabled. The autosuspend delay was not set either, the value is set to 5s since HDMI is rather painful to resume. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20220616222910.136854-2-pierre-louis.bossart@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/x86/intel_hdmi_audio.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 0d828e35b401..3b04c70a73e3 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -33,6 +33,8 @@ #include <drm/intel_lpe_audio.h> #include "intel_hdmi_audio.h" +#define INTEL_HDMI_AUDIO_SUSPEND_DELAY_MS 5000 + #define for_each_pipe(card_ctx, pipe) \ for ((pipe) = 0; (pipe) < (card_ctx)->num_pipes; (pipe)++) #define for_each_port(card_ctx, port) \ @@ -1802,8 +1804,11 @@ static int __hdmi_lpe_audio_probe(struct platform_device *pdev) pdata->notify_audio_lpe = notify_audio_lpe; spin_unlock_irq(&pdata->lpe_audio_slock); + pm_runtime_set_autosuspend_delay(&pdev->dev, INTEL_HDMI_AUDIO_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_enable(&pdev->dev); pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_idle(&pdev->dev); dev_dbg(&pdev->dev, "%s: handle pending notification\n", __func__); for_each_port(card_ctx, port) { -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 06/53] ALSA: usb-audio: US16x08: Move overflow check before array access
by Sasha Levin 28 Jun '22

28 Jun '22

From: Daniil Dementev <d.dementev(a)ispras.ru> [ Upstream commit 3ddbe35d9a2ebd4924d458e0246b4ba6c13bb456 ] Buffer overflow could occur in the loop "while", due to accessing an array element before checking the index. Found by Linux Verification Center (linuxtesting.org) with SVACE. Signed-off-by: Daniil Dementev <d.dementev(a)ispras.ru> Reviewed-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> Link: https://lore.kernel.org/r/20220610165732.2904-1-d.dementev@ispras.ru Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_us16x08.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index b7b6f3834ed5..6eb7d93b358d 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -637,10 +637,10 @@ static int snd_get_meter_comp_index(struct snd_us16x08_meter_store *store) } } else { /* skip channels with no compressor active */ - while (!store->comp_store->val[ + while (store->comp_index <= SND_US16X08_MAX_CHANNELS + && !store->comp_store->val[ COMP_STORE_IDX(SND_US16X08_ID_COMP_SWITCH)] - [store->comp_index - 1] - && store->comp_index <= SND_US16X08_MAX_CHANNELS) { + [store->comp_index - 1]) { store->comp_index++; } ret = store->comp_index++; -- 2.35.1

1 0

[PATCH v2] ASoC: samsung: s3c24xx-i2s: Fix typo in DAIFMT handling
by Charles Keepax 28 Jun '22

28 Jun '22

The conversion of the set_fmt callback to direct clock specification included a small typo, correct the affected code. Fixes: 0b491c7c1b25 ("ASoC: samsung: Update to use set_fmt_new callback") Reported-by: kernel test robot <lkp(a)intel.com> Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Changes since v1: - Correct fixes tag. sound/soc/samsung/s3c24xx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 4082ad7cbcc11..c1a314b86b155 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -170,7 +170,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, pr_debug("hw_params r: IISMOD: %x \n", iismod); switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_BC_CFC: + case SND_SOC_DAIFMT_BC_FC: iismod |= S3C2410_IISMOD_SLAVE; break; case SND_SOC_DAIFMT_BP_FP: -- 2.30.2

2 1

[PATCH v3 0/2] Read _SUB from ACPI to be able to identify firmware
by Stefan Binding 28 Jun '22

28 Jun '22

CS35L41 has a DSP which is able to run firmware, as well as a tuning file. Different systems may want to use different firmwares and tuning files, and some firmwares/tunings may not be compatible with other systems. To allow a system to select the correct fimware/tuning, we can read an _SUB from the ACPI. This _SUB can then be used to uniquely identify the system in the firmware/tuning file name. Add a helper function which reads the _SUB, so this can be used by other parts in the future. Add support inside the CS35L41 ASoC driver to read this _SUB, and save it appropriately. Changes since v2: - Fix error in function prototype Changes since v1: - Add length validation for SSID String - Rename API - Allocate memory inside API - Use ACPI_HANDLE macro instead of ACPI_COMPANION - Improve error handling Stefan Binding (2): ACPI: utils: Add api to read _SUB from ACPI ASoC: cs35l41: Read System Name from ACPI _SUB to identify firmware drivers/acpi/utils.c | 38 ++++++++++++++++++++++++++++++++++++++ include/linux/acpi.h | 6 ++++++ sound/soc/codecs/cs35l41.c | 24 ++++++++++++++++++++++++ 3 files changed, 68 insertions(+) -- 2.25.1

2 4

[PATCH v3 0/2] Fix apq8016 compat string
by Bryan O'Donoghue 28 Jun '22

28 Jun '22

V3: - Marks qcom,lpass-cpu-apq8016 as deprecated instead of removing - Bjorn V2: - Adds Reviewed-by: - Srini - Adds Fixes - Srini V1: Reusing the apq8016 on msm8939 I found running checkpatch that the compat string for the LPASS was throwing a warning. This is easily fixed by alinging the YAML, DTS and driver to the documented compat string - compatible = "qcom,lpass-cpu-apq8016"; + compatible = "qcom,apq8016-lpass-cpu"; Bryan O'Donoghue (2): ASoC: qcom: lpass: Fix apq8016 compat string to match yaml arm64: dts: qcom: Fix apq8016 compat string to match yaml arch/arm64/boot/dts/qcom/msm8916.dtsi | 2 +- sound/soc/qcom/lpass-apq8016.c | 1 + sound/soc/qcom/lpass-cpu.c | 5 +++++ 3 files changed, 7 insertions(+), 1 deletion(-) -- 2.35.1

3 5

[PATCH 0/8] ASoC: max98396: Some assorted fixes and additions
by Daniel Mack 28 Jun '22

28 Jun '22

This is a series of some patches that I collected while using the max98396 driver is a TDM mode setup. They correct BSEL and PCM mode configs, add support for power supplies and add some bits to the documentation. The code is tested in TDM-16 and TDM-8 mode with 32 channel width. Thanks, Daniel Daniel Mack (8): ASoC: dt-bindings: max98396: add voltage supplies ASoC: dt-bindings: max98396: Add #sound-dai-cells ASoC: dt-bindings: max98396: Document adi,bypass-slot-no ASoC: max98396: add voltage regulators ASoC: max98396: Improve some error prints ASoC: max98396: Fix register access for PCM format settings ASoC: max98396: Implement DSP speaker monitor ASoC: max98396: Fix TDM mode BSEL settings .../bindings/sound/adi,max98396.yaml | 34 ++- sound/soc/codecs/max98396.c | 265 ++++++++++++++---- sound/soc/codecs/max98396.h | 8 + 3 files changed, 257 insertions(+), 50 deletions(-) -- 2.36.1

3 16

[PATCH] ASoC: samsung: s3c24xx-i2s: Fix typo in DAIFMT handling
by Charles Keepax 27 Jun '22

27 Jun '22

The conversion of the set_fmt callback to direct clock specification included a small typo, correct the affected code. Fixes: 91c49199e6d6 ("ASoC: samsung: Update to use set_fmt_new callback") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/samsung/s3c24xx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 4082ad7cbcc11..c1a314b86b155 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -170,7 +170,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, pr_debug("hw_params r: IISMOD: %x \n", iismod); switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_BC_CFC: + case SND_SOC_DAIFMT_BC_FC: iismod |= S3C2410_IISMOD_SLAVE; break; case SND_SOC_DAIFMT_BP_FP: -- 2.30.2

3 9

RT711/RT711-SDCA: use codec switch/volume
by GitHub pull_request - opened 27 Jun '22

27 Jun '22

alsa-project/alsa-ucm-conf pull request #180 was opened from plbossart: This PR uses the settings provided by @shumingfan in https://github.com/alsa-project/alsa-ucm-conf/issues/179 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/180 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/180.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: tlv320adcx140: Fix tx_mask check
by Sascha Hauer 27 Jun '22

27 Jun '22

The tx_mask check doesn't reflect what the driver and the chip support. The check currently checks for exactly two slots being enabled. The tlv320adcx140 supports anything between one and eight channels, so relax the check accordingly. The tlv320adcx140 supports arbitrary tx_mask settings, but the driver currently only supports adjacent slots beginning with the first slot, so extend the check to check that the first slot is being used and that there are no holes in the tx_mask. Leave a comment to make it's the driver that limits the tx_mask settings, not the chip itself. While at it remove the set-but-unused struct adcx140p_priv::tdm_delay field. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/codecs/tlv320adcx140.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index b55f0b836932b..0b729658fde80 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -33,7 +33,6 @@ struct adcx140_priv { bool micbias_vg; unsigned int dai_fmt; - unsigned int tdm_delay; unsigned int slot_width; }; @@ -792,12 +791,13 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); - unsigned int lsb; - /* TDM based on DSP mode requires slots to be adjacent */ - lsb = __ffs(tx_mask); - if ((lsb + 1) != __fls(tx_mask)) { - dev_err(component->dev, "Invalid mask, slots must be adjacent\n"); + /* + * The chip itself supports arbitrary masks, but the driver currently + * only supports adjacent slots beginning at the first slot. + */ + if (tx_mask != GENMASK(__fls(tx_mask), 0)) { + dev_err(component->dev, "Only lower adjacent slots are supported\n"); return -EINVAL; } @@ -812,7 +812,6 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, return -EINVAL; } - adcx140->tdm_delay = lsb; adcx140->slot_width = slot_width; return 0; -- 2.30.2

2 1

[PATCH v4] ASoC: ti: omap-mcbsp: duplicate sysfs error
by David Owens 27 Jun '22

27 Jun '22

From: David Owens <daowens01(a)gmail.com> Convert to managed versions of sysfs and clk allocation to simplify unbinding and error handling in probe. Managed sysfs node creation specifically addresses the following error seen the second time probe is attempted after sdma_pcm_platform_register() previously requsted probe deferral: sysfs: cannot create duplicate filename '/devices/platform/68000000.ocp/49022000.mcbsp/max_tx_thres' Signed-off-by: David Owens <dowens(a)precisionplanting.com> --- Changes in v4: * Reverted change that added use ATTRIBUTE_GROUP and left the attribute structs as-is to fix compilation error and lessen the area of impact. Changes in v3: * Whitespace changes only to allow clean apply Changes in v2: * Improved error handling --- sound/soc/ti/omap-mcbsp-priv.h | 2 -- sound/soc/ti/omap-mcbsp-st.c | 14 ++------------ sound/soc/ti/omap-mcbsp.c | 19 ++----------------- 3 files changed, 4 insertions(+), 31 deletions(-) diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h index 7865cda4bf0a..da519ea1f303 100644 --- a/sound/soc/ti/omap-mcbsp-priv.h +++ b/sound/soc/ti/omap-mcbsp-priv.h @@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, /* Sidetone specific API */ int omap_mcbsp_st_init(struct platform_device *pdev); -void omap_mcbsp_st_cleanup(struct platform_device *pdev); - int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp); int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp); diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c index 0bc7d26c660a..7e8179cae92e 100644 --- a/sound/soc/ti/omap-mcbsp-st.c +++ b/sound/soc/ti/omap-mcbsp-st.c @@ -347,7 +347,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data) return -ENOMEM; - st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); + st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick"); if (IS_ERR(st_data->mcbsp_iclk)) { dev_warn(mcbsp->dev, "Failed to get ick, sidetone might be broken\n"); @@ -359,7 +359,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data->io_base_st) return -ENOMEM; - ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); + ret = devm_device_add_group(mcbsp->dev, &sidetone_attr_group); if (ret) return ret; @@ -368,16 +368,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev) return 0; } -void omap_mcbsp_st_cleanup(struct platform_device *pdev) -{ - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - - if (mcbsp->st_data) { - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); - clk_put(mcbsp->st_data->mcbsp_iclk); - } -} - static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 58d8e200a7b9..9fb7cf0c9f88 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -702,8 +702,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; - ret = sysfs_create_group(&mcbsp->dev->kobj, - &additional_attr_group); + ret = devm_device_add_group(mcbsp->dev, &additional_attr_group); if (ret) { dev_err(mcbsp->dev, "Unable to create additional controls\n"); @@ -711,16 +710,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) } } - ret = omap_mcbsp_st_init(pdev); - if (ret) - goto err_st; - - return 0; - -err_st: - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - return ret; + return omap_mcbsp_st_init(pdev); } /* @@ -1431,11 +1421,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req)) cpu_latency_qos_remove_request(&mcbsp->pm_qos_req); - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - - omap_mcbsp_st_cleanup(pdev); - return 0; } -- 2.34.1

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[PATCH v3] ASoC: ti: omap-mcbsp: duplicate sysfs error
by David Owens 27 Jun '22

27 Jun '22

Convert to managed versions of sysfs and clk allocation to simplify unbinding and error handling in probe. Managed sysfs node creation specifically addresses the following error seen the second time probe is attempted after sdma_pcm_platform_register() previously requested probe deferral: sysfs: cannot create duplicate filename '/devices/platform/68000000.ocp/49022000.mcbsp/max_tx_thres' Signed-off-by: David Owens <dowens(a)precisionplanting.com> --- Changes in v3: * Whitespace changes only to allow clean applly Changes in v2: * Improved error handling --- sound/soc/ti/omap-mcbsp-priv.h | 2 -- sound/soc/ti/omap-mcbsp-st.c | 21 ++++----------------- sound/soc/ti/omap-mcbsp.c | 26 ++++---------------------- 3 files changed, 8 insertions(+), 41 deletions(-) diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h index 7865cda4bf0a..da519ea1f303 100644 --- a/sound/soc/ti/omap-mcbsp-priv.h +++ b/sound/soc/ti/omap-mcbsp-priv.h @@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, /* Sidetone specific API */ int omap_mcbsp_st_init(struct platform_device *pdev); -void omap_mcbsp_st_cleanup(struct platform_device *pdev); - int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp); int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp); diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c index 0bc7d26c660a..402a57a502e6 100644 --- a/sound/soc/ti/omap-mcbsp-st.c +++ b/sound/soc/ti/omap-mcbsp-st.c @@ -292,14 +292,11 @@ static ssize_t st_taps_store(struct device *dev, static DEVICE_ATTR_RW(st_taps); -static const struct attribute *sidetone_attrs[] = { +static struct attribute *sidetone_attrs[] = { &dev_attr_st_taps.attr, NULL, }; - -static const struct attribute_group sidetone_attr_group = { - .attrs = (struct attribute **)sidetone_attrs, -}; +ATTRIBUTE_GROUPS(sidetone); int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp) { @@ -347,7 +344,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data) return -ENOMEM; - st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); + st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick"); if (IS_ERR(st_data->mcbsp_iclk)) { dev_warn(mcbsp->dev, "Failed to get ick, sidetone might be broken\n"); @@ -359,7 +356,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data->io_base_st) return -ENOMEM; - ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); + ret = devm_device_add_group(mcbsp->dev, &sidetone_group); if (ret) return ret; @@ -368,16 +365,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev) return 0; } -void omap_mcbsp_st_cleanup(struct platform_device *pdev) -{ - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - - if (mcbsp->st_data) { - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); - clk_put(mcbsp->st_data->mcbsp_iclk); - } -} - static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 4479d74f0a45..395493a2d965 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -595,16 +595,13 @@ static ssize_t dma_op_mode_store(struct device *dev, static DEVICE_ATTR_RW(dma_op_mode); -static const struct attribute *additional_attrs[] = { +static struct attribute *additional_attrs[] = { &dev_attr_max_tx_thres.attr, &dev_attr_max_rx_thres.attr, &dev_attr_dma_op_mode.attr, NULL, }; - -static const struct attribute_group additional_attr_group = { - .attrs = (struct attribute **)additional_attrs, -}; +ATTRIBUTE_GROUPS(additional); /* * McBSP1 and McBSP3 are directly mapped on 1610 and 1510. @@ -702,8 +699,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; - ret = sysfs_create_group(&mcbsp->dev->kobj, - &additional_attr_group); + ret = devm_device_add_group(mcbsp->dev, &additional_group); if (ret) { dev_err(mcbsp->dev, "Unable to create additional controls\n"); @@ -711,16 +707,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) } } - ret = omap_mcbsp_st_init(pdev); - if (ret) - goto err_st; - - return 0; - -err_st: - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - return ret; + return omap_mcbsp_st_init(pdev); } /* @@ -1431,11 +1418,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req)) cpu_latency_qos_remove_request(&mcbsp->pm_qos_req); - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - - omap_mcbsp_st_cleanup(pdev); - return 0; } -- 2.34.1

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Re: [PATCH v2] ASoC: ti: omap-mcbsp: duplicate sysfs error
by Mark Brown 27 Jun '22

27 Jun '22

On Wed, 15 Jun 2022 10:09:55 -0500, David Owens wrote: > Convert to managed versions of sysfs and clk allocation to simplify > unbinding and error handling in probe. Managed sysfs node > creation specifically addresses the following error seen the second time > probe is attempted after sdma_pcm_platform_register() previously requsted > probe deferral: > > sysfs: cannot create duplicate filename '/devices/platform/68000000.ocp/49022000.mcbsp/max_tx_thres' > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: ti: omap-mcbsp: duplicate sysfs error commit: 6d31e225742a1955db8a0c6f6f52beb748a2895e All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

[PATCH] drm: amd: amdgpu: fix checkpatch warnings
by Vijendar Mukunda 27 Jun '22

27 Jun '22

From: vijendar <vijendar.mukunda(a)amd.com> Fixed below checkpatch warnings and errors drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c:131: CHECK: Comparison to NULL could be written "apd" drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c:150: CHECK: Comparison to NULL could be written "apd" drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c:196: CHECK: Prefer kernel type 'u64' over 'uint64_t' drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c:224: CHECK: Please don't use multiple blank lines drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c:226: CHECK: Comparison to NULL could be written "!adev->acp.acp_genpd" drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c:233: CHECK: Please don't use multiple blank lines drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c:239: CHECK: Alignment should match open parenthesis drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c:241: CHECK: Comparison to NULL could be written "!adev->acp.acp_cell" drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c:247: CHECK: Comparison to NULL could be written "!adev->acp.acp_res" drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c:253: CHECK: Comparison to NULL could be written "!i2s_pdata" drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c:350: CHECK: Alignment should match open parenthesis drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c:550: ERROR: that open brace { should be on the previous line Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 27 +++++++++---------------- 1 file changed, 10 insertions(+), 17 deletions(-) diff --git a/drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c b/drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c index cc9c9f8b23b2..ba1605ff521f 100644 --- a/drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c +++ b/drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c @@ -128,7 +128,7 @@ static int acp_poweroff(struct generic_pm_domain *genpd) struct amdgpu_device *adev; apd = container_of(genpd, struct acp_pm_domain, gpd); - if (apd != NULL) { + if (apd) { adev = apd->adev; /* call smu to POWER GATE ACP block * smu will @@ -147,7 +147,7 @@ static int acp_poweron(struct generic_pm_domain *genpd) struct amdgpu_device *adev; apd = container_of(genpd, struct acp_pm_domain, gpd); - if (apd != NULL) { + if (apd) { adev = apd->adev; /* call smu to UNGATE ACP block * smu will @@ -193,7 +193,7 @@ static int acp_genpd_remove_device(struct device *dev, void *data) static int acp_hw_init(void *handle) { int r; - uint64_t acp_base; + u64 acp_base; u32 val = 0; u32 count = 0; struct i2s_platform_data *i2s_pdata = NULL; @@ -220,37 +220,32 @@ static int acp_hw_init(void *handle) return -EINVAL; acp_base = adev->rmmio_base; - - adev->acp.acp_genpd = kzalloc(sizeof(struct acp_pm_domain), GFP_KERNEL); - if (adev->acp.acp_genpd == NULL) + if (!adev->acp.acp_genpd) return -ENOMEM; adev->acp.acp_genpd->gpd.name = "ACP_AUDIO"; adev->acp.acp_genpd->gpd.power_off = acp_poweroff; adev->acp.acp_genpd->gpd.power_on = acp_poweron; - - adev->acp.acp_genpd->adev = adev; pm_genpd_init(&adev->acp.acp_genpd->gpd, NULL, false); - adev->acp.acp_cell = kcalloc(ACP_DEVS, sizeof(struct mfd_cell), - GFP_KERNEL); + adev->acp.acp_cell = kcalloc(ACP_DEVS, sizeof(struct mfd_cell), GFP_KERNEL); - if (adev->acp.acp_cell == NULL) { + if (!adev->acp.acp_cell) { r = -ENOMEM; goto failure; } adev->acp.acp_res = kcalloc(5, sizeof(struct resource), GFP_KERNEL); - if (adev->acp.acp_res == NULL) { + if (!adev->acp.acp_res) { r = -ENOMEM; goto failure; } i2s_pdata = kcalloc(3, sizeof(struct i2s_platform_data), GFP_KERNEL); - if (i2s_pdata == NULL) { + if (!i2s_pdata) { r = -ENOMEM; goto failure; } @@ -346,8 +341,7 @@ static int acp_hw_init(void *handle) adev->acp.acp_cell[3].platform_data = &i2s_pdata[2]; adev->acp.acp_cell[3].pdata_size = sizeof(struct i2s_platform_data); - r = mfd_add_hotplug_devices(adev->acp.parent, adev->acp.acp_cell, - ACP_DEVS); + r = mfd_add_hotplug_devices(adev->acp.parent, adev->acp.acp_cell, ACP_DEVS); if (r) goto failure; @@ -546,8 +540,7 @@ static const struct amd_ip_funcs acp_ip_funcs = { .set_powergating_state = acp_set_powergating_state, }; -const struct amdgpu_ip_block_version acp_ip_block = -{ +const struct amdgpu_ip_block_version acp_ip_block = { .type = AMD_IP_BLOCK_TYPE_ACP, .major = 2, .minor = 2, -- 2.25.1

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Re: Sound not working after commit bbf7d3b1c4f40eb02dd1dffb500ba00b0bff0303 on Amlogic A311D device
by Pierre-Louis Bossart 27 Jun '22

27 Jun '22

[Adding alsa-devel] On 6/26/22 04:29, Alex Natalsson wrote: > Hello friends. > I have Khadas VIM3 SBC with Amlogic A311D SOC device. > After update kernel my media software (alsamixer, mplayer, ffplay and > etc.) not responding after launch and not terminating throught kill > -9. > > The log after alsamixer launch: > [ 103.786358] Internal error: Oops: 96000004 [#1] SMP > [ 103.786509] Modules linked in: snd_seq_dummy snd_hrtimer snd_seq > snd_seq_device bnep nls_utf8 mt7601u mac80211 hci_uart libarc4 joydev > brcmfmac btbcm panfrost bluetooth meson_vdec(C) brcmutil gpu_sched > cfg80211 dwmac_generic videobuf2_dma_contig v4l2_mem2mem > videobuf2_memops videobuf2_v4l2 videobuf2_common drm_shmem_helper > ecdh_generic dwmac_meson8b stmmac_platform dw_hdmi_i2s_audio stmmac > ecc ir_nec_decoder videodev pcs_xpcs rc_khadas phylink ptp > meson_saradc meson_ir rc_core snd_soc_meson_axg_frddr > snd_soc_meson_axg_toddr snd_soc_meson_axg_sound_card > snd_soc_meson_card_utils snd_soc_meson_axg_fifo ao_cec_g12a > snd_soc_meson_axg_tdmin snd_soc_meson_axg_tdm_interface > snd_soc_meson_axg_tdmout snd_soc_meson_axg_tdm_formatter rfkill > meson_gxbb_wdt fuse crypto_user uas usb_storage adc_keys > gpio_keys_polled industrialio > [ 103.857778] CPU: 3 PID: 530 Comm: pipewire-media- Tainted: G > C 5.16.0-ARCH+ #8 Any chance you could try with an updated kernel? It's not clear to me if this is a real issue or just a warning that's been fixed since by commit. bbf7d3b1c4f40 "ASoC: soc-pcm: Fix DPCM lockdep warning due to nested stream locks" You may also want to open a bugzilla issue to attach the full dmesg log and your configuration reported by 'alsa-info' > [ 103.857791] Hardware name: Khadas VIM3 (DT) > [ 103.857795] pstate: 60000005 (nZCv daif -PAN -UAO -TCO -DIT -SSBS BTYPE=--) > [ 103.857801] pc : dpcm_be_connect+0xd0/0x2d0 > [ 103.881758] lr : dpcm_add_paths.isra.0+0xcc/0x1e0 > [ 103.881763] sp : ffff800009f4b870 > [ 103.881765] x29: ffff800009f4b870 x28: ffff000003b944c0 x27: 0000000000000001 > [ 103.881772] x26: 0000000000000001 x25: 0000000000000340 x24: 00000000000002a0 > [ 103.881777] x23: 0000000000000001 x22: 0000000000000000 x21: ffff000005b18080 > [ 103.881782] x20: ffff000005b183c0 x19: ffff000005b1c080 x18: 0000000000000000 > [ 103.881787] x17: 0000000000000000 x16: 0000000000000000 x15: 0000000000000000 > [ 103.881792] x14: 0000000000000000 x13: 0000000000000000 x12: 0000000000000000 > [ 103.881797] x11: 0000000000000000 x10: 0000000000000000 x9 : 0000000000000000 > [ 103.881801] x8 : ffff000012020400 x7 : 0000000000000000 x6 : 000000000000003f > [ 103.881806] x5 : 0000000000000040 x4 : 0000000000000000 x3 : ffff000002822000 > [ 103.881811] x2 : ffff000002861b50 x1 : ffff00000282f800 x0 : ffff000002861800 > [ 103.881817] Call trace: > [ 103.881819] dpcm_be_connect+0xd0/0x2d0 > [ 103.881823] dpcm_add_paths.isra.0+0xcc/0x1e0 > [ 103.881826] dpcm_fe_dai_open+0x80/0x194 > [ 103.881831] snd_pcm_open_substream+0xa4/0x174 > [ 103.881838] snd_pcm_open.part.0+0xd8/0x1dc > [ 103.881843] snd_pcm_capture_open+0x64/0x94 > [ 103.992730] snd_open+0xac/0x1d0 > [ 103.992739] chrdev_open+0xe0/0x2d0 > [ 103.999389] do_dentry_open+0x12c/0x3a0 > [ 103.999397] vfs_open+0x30/0x3c > [ 103.999401] do_open+0x204/0x414 > [ 103.999405] path_openat+0x10c/0x27c > [ 103.999408] do_filp_open+0x80/0x130 > [ 103.999412] do_sys_openat2+0xb4/0x170 > [ 103.999414] __arm64_sys_openat+0x64/0xb0 > [ 103.999416] invoke_syscall+0x48/0x114 > [ 103.999426] el0_svc_common.constprop.0+0xd4/0xfc > [ 103.999429] do_el0_svc+0x28/0x90 > [ 103.999432] el0_svc+0x28/0x80 > > After I removed pipewire software the error message was > disappear. But the sound and media still doesn't working and alsamixer > cannot change sound card to my usb audio device.

2 1

[PATCH v2 0/2] Read _SUB from ACPI to be able to identify firmware
by Stefan Binding 27 Jun '22

27 Jun '22

CS35L41 has a DSP which is able to run firmware, as well as a tuning file. Different systems may want to use different firmwares and tuning files, and some firmwares/tunings may not be compatible with other systems. To allow a system to select the correct fimware/tuning, we can read an _SUB from the ACPI. This _SUB can then be used to uniquely identify the system in the firmware/tuning file name. Add a helper function which reads the _SUB, so this can be used by other parts in the future. Add support inside the CS35L41 ASoC driver to read this _SUB, and save it appropriately. Changes since v1: - Add length validation for SSID String - Rename API - Allocate memory inside API - Use ACPI_HANDLE macro instead of ACPI_COMPANION - Improve error handling Stefan Binding (2): ACPI: utils: Add api to read _SUB from ACPI ASoC: cs35l41: Read System Name from ACPI _SUB to identify firmware drivers/acpi/utils.c | 38 ++++++++++++++++++++++++++++++++++++++ include/linux/acpi.h | 6 ++++++ sound/soc/codecs/cs35l41.c | 24 ++++++++++++++++++++++++ 3 files changed, 68 insertions(+) -- 2.25.1

2 3

Pandora audio using machine_is_omap3_pandora() check
by Tony Lindgren 27 Jun '22

27 Jun '22

Hi Grazvydas, Arnd noticed that pandora audio is using machine_is_omap3_pandora() check that never succeeds for devicetree booting machines. Looks like this has been broken at some point many years ago with the devicetree conversion. Does anybody have an interest in fixing this driver? Regards, Tony

2 2

[PATCH 0/5] seq: code change for cell helper functionalities
by GitHub pull_request - opened 27 Jun '22

27 Jun '22

alsa-project/alsa-gobject pull request #80 was opened from takaswie: In design of ALSA Sequencer, cell is a unit to deliver events. This patchset adds code change for helper functionalities. ``` Takashi Sakamoto (5): event: add public API to compute pool consumption when scheduling event client-pool: change signedness for size properties user-client/client-pool: update function comments about event scheduling all: replace 'represent' with 'express' in comments ci: separate libudev-dev installation from the other .github/workflows/build.yml | 3 +- doc/ctl/overview.md | 2 +- doc/rawmidi/overview.md | 2 +- doc/seq/overview.md | 6 +-- doc/timer/overview.md | 2 +- src/ctl/card-info.c | 2 +- src/ctl/card.c | 6 +-- src/ctl/elem-id.c | 4 +- src/hwdep/device-info.c | 4 +- src/rawmidi/stream-pair.c | 6 +-- src/rawmidi/substream-info.c | 4 +- src/rawmidi/substream-params.c | 4 +- src/rawmidi/substream-status.c | 4 +- src/seq/addr.c | 4 +- src/seq/alsaseq.map | 1 + src/seq/client-info.c | 4 +- src/seq/client-pool.c | 91 ++++++++++++++++++---------------- src/seq/event-data-connect.c | 4 +- src/seq/event-data-ctl.c | 4 +- src/seq/event-data-note.c | 4 +- src/seq/event-data-queue.c | 4 +- src/seq/event-data-result.c | 4 +- src/seq/event.c | 16 ++++++ src/seq/event.h | 2 + src/seq/port-info.c | 4 +- src/seq/queue-info.c | 4 +- src/seq/queue-status.c | 4 +- src/seq/queue-tempo.c | 4 +- src/seq/remove-filter.c | 4 +- src/seq/subscribe-data.c | 4 +- src/seq/system-info.c | 4 +- src/seq/user-client.c | 16 ++++-- src/timer/device-id.c | 4 +- src/timer/device-info.c | 4 +- src/timer/device-params.c | 4 +- src/timer/device-status.c | 4 +- src/timer/instance-info.c | 4 +- src/timer/instance-params.c | 4 +- src/timer/instance-status.c | 2 +- src/timer/user-instance.c | 6 +-- tests/alsaseq-event | 1 + 41 files changed, 148 insertions(+), 116 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/80 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/80.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH v5 0/2] Add pinctrl support adsp bypass platforms
by Srinivasa Rao Mandadapu 27 Jun '22

27 Jun '22

This patch set is to make clock voting optinal for adsp bypass sc7280 platforms. Changes Since V4: -- Fix the improper device node variable usage. -- Optimize the code by removing redundant private variant data structure variable "is_clk_optional". Changes Since V3: -- Revert compatible type change in dt bindings. -- Update boolean flag description in dt bindings. -- Drop redundant clock optional flag initialization. -- Remove redundant if check. Changes Since V2: -- Remove redundant lpi pincontrol variant data structure and compatible entry. -- Add adsp bypass mode boolean param check. -- Remove compatible name in dt bindings. -- Update dt binding bypass mode boolean param. Changes Since V1: -- Update commit message. Srinivasa Rao Mandadapu (2): dt-bindings: pinctrl: qcom: sc7280: Add boolean param for ADSP bypass platforms pinctrl: qcom: sc7280: Add clock optional check for ADSP bypass targets .../devicetree/bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 5 +++++ drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 2 +- drivers/pinctrl/qcom/pinctrl-lpass-lpi.h | 1 - drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 1 - 4 files changed, 6 insertions(+), 3 deletions(-) -- 2.7.4

1 2

[PATCH 0/2] Update machine driver for audioreach solution
by Srinivasa Rao Mandadapu 27 Jun '22

27 Jun '22

This patch set is to update sc7280 machine driver for audioreach support. Srinivasa Rao Mandadapu (2): ASoC: google: dt-bindings: Update sc7280-audioreach-herobrine compatible name ASoC: qcom: Add driver support for audioreach solution .../bindings/sound/google,sc7280-herobrine.yaml | 1 + sound/soc/qcom/sc7280.c | 34 ++++++++++++++++++++++ 2 files changed, 35 insertions(+) -- 2.7.4

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[PATCH v8 0/8] ASoC: mediatek: Add support for MT8186 SoC
by Jiaxin Yu 27 Jun '22

27 Jun '22

This series of patches adds support for Mediatek AFE of MT8186 Soc. Patches are based on broonie tree "for-next" branch. Changes since v7: - Optimize the code logic, such as the return value and position of a function. - Use devm_pm_runtime_enable() instead of pm_runtime_enable(). - Use pm_runtime_resume_and_get() instead of pm_runtime_get_sync(). Changes since v6: - The Makefile and Kconfig updates are moved into the driver patches so that can keep independence of each patch. Changes since v5: - fix build error about function snd_soc_card_jack_new that modified by: Link: https://lore.kernel.org/r/20220408041114.6024-1-akihiko.odaki@gmail.com - some have been accepted, details are in the link below: Link: https://lore.kernel.org/all/165459931885.399031.2621579592368573898.b4-ty@k… Changes since v4: - [v5 07/20] - remove unsusd controls - [v5 09/20] - correct indent error - [v5 10/20] - [v5 13/20] - [v5 14/20] - fix the return value if the value is different from the previous value in mixer controls - [v5 17/20] - [v5 19/20] - correct the compatible name with '_' instead of '-' - [v5 18/20] - [v5 20/20] - correct the yaml after 'pip3 install dtschema --upgrade' Changes since v3: - [v4 09/18] - remove DEBUG_COEFF debugging code - [v4 10/18] - simplify the logic of the code - [v4 13/18] - split out the MT6366 bits into mt8186-mt6366-commom.c - fix build error of "error: 'struct dev_pm_info' has no member named 'runtime_error'" - fix bug of adda dai driver - add route for pcm interface channel 2. - [v4 15/18] - [v4 17/18] - commonize the configuration of the codec - move MT6366 common bits into mt8186-mt6366-common.c Changes since v2: - add a new compatible string "mediatek,mt6366-sound" - modify the log level for simplicity - use dev_err_probe(...) instead of dev_err(...) in dev probe() - optimized the logic of some code - use BIT() and GENMASK() macros to descript the registers Thanks for AngeloGioacchino's careful reviews. Changes since v1: [v2 01/17] - add a new ID to the existing mt6358 codec driver [v2 03/17] - modify log level in DAPM events - use standard numeric control with name ending in Switch - return 1 when the value changed in mixer control's .get callback [v2 05/17] - ending in Switch to the standard on/off controls - change to "HW Gain 1 Volume" and "HW Gain 2 Volume" [v2 09/17] - return an error in the default case rather than just picking one of the behaviours when do .set_fmt - use the new defines that are _PROVIDER_MASK, _DAIFMT_CBP_CFP and _DAIFMT_CBC_CFC [v2 10/17] [v2 11/17] - the clock and gpio are aplit out into separate patches The source file's GPL comment use c++ style, and the header fils's GPL comment use c style. We have added "Switch" after the names of all the controls that just are simple on/off. Jiaxin Yu (8): dt-bindings: mediatek: mt6358: add new compatible for using mt6366 ASoC: mediatek: mt8186: add platform driver ASoC: mediatek: mt8186: add mt8186-mt6366 common driver dt-bindings: mediatek: mt8186: add audio afe document ASoC: mediatek: mt8186: add machine driver with mt6366, da7219 and max98357 dt-bindings: mediatek: mt8186: add mt8186-mt6366-da7219-max98357 document ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682s dt-bindings: mediatek: mt8186: add mt8186-mt6366-rt1019-rt5682s document .../devicetree/bindings/sound/mt6358.txt | 4 +- .../bindings/sound/mt8186-afe-pcm.yaml | 175 + .../sound/mt8186-mt6366-da7219-max98357.yaml | 75 + .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 75 + sound/soc/mediatek/Kconfig | 44 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8186/Makefile | 22 + sound/soc/mediatek/mt8186/mt8186-afe-common.h | 235 ++ .../soc/mediatek/mt8186/mt8186-afe-control.c | 255 ++ sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3011 +++++++++++++++++ .../mediatek/mt8186/mt8186-mt6366-common.c | 57 + .../mediatek/mt8186/mt8186-mt6366-common.h | 17 + .../mt8186/mt8186-mt6366-da7219-max98357.c | 1002 ++++++ .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 978 ++++++ 14 files changed, 5950 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml create mode 100644 sound/soc/mediatek/mt8186/Makefile create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c -- 2.18.0

2 10

attach/detach by mixer class implementation of alsa-lib mixer API
by Takashi Sakamoto 27 Jun '22

27 Jun '22

Hi Jaroslav, Recent years I'm bothered about unexpected abort of pulseaudio and pipewire processes when testing user-defined control element set. They aborts at element removal event. ``` pulseaudio: mixer.c:149: hctl_elem_event_handler: Assertion `bag_empty(bag)' failed. wireplumber: mixer.c:149: hctl_elem_event_handler: Assertion `bag_empty(bag)' failed. ``` Would I ask your opinion about the design of alsa-lib mixer API? As long as I investigate, these programs seem to have careless coding as alsa-lib mixer API application. Both of them attaches an instance of snd_mixer_elem_t to an instance of snd_hctl_elem_t by calling snd_mixer_elem_attach() when detecting element addition, but never detach it even if detecting element removal. In the case, the link list (=bag) of mixer API internal never becomes empty. It has link entries as much as the number of registered mixer classes which attaches snd_mixer_elem_t. Then it hits the assertion. I think the design of alsa-lib mixer API demands mixer class implementation to detach at element removal which attached at element addition. But I have less conviction about it since enough unfamiliar. I'm glad if receiving your opinion about it. I wrote test program for the issued behaviour: - https://gist.github.com/takaswie/8378fe3bc04652d83428694cd7573bc0 For test, please use sample script in alsa-gobject project: - https://github.com/alsa-project/alsa-gobject/blob/master/samples/ctl The patches for pulseaudio/pipewire are prepared: - https://gitlab.freedesktop.org/takaswie/pulseaudio/-/commit/topic/fix-wrong… - https://gitlab.freedesktop.org/takaswie/pipewire/-/commits/topic/fix-wrong-… Regards Takashi Sakamoto

2 2

mixer: add documentation about postcondition of removal event processing
by GitHub pull_request - opened 27 Jun '22

27 Jun '22

alsa-project/alsa-lib pull request #244 was opened from takaswie: PulseAudio and PipeWire seems to appear including careless code to process events of mixer element in implementation of mixer class. * https://lore.kernel.org/alsa-devel/YrbxZ2b+3rIdi7Ut@workstation/ They register own implementation of mixer class. At addition event, they attach own mixer element into hcontrol element. However at removal event, they never detach the mixer element from hcontrol element. They hit assertion in mixer API internal due to unsatisfied postcondition. This commit adds documentation about postcondition of removal event processing so that developer for implementation of mixer class easily realize the way to satisfy the postcondition. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-lib/pull/244 Patch URL : https://github.com/alsa-project/alsa-lib/pull/244.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v3 0/2] [PATCH v4 0/2] Add pinctrl support adsp bypass platforms
by Srinivasa Rao Mandadapu 26 Jun '22

26 Jun '22

This patch set is to make clock voting optinal for adsp bypass sc7280 platforms. Changes Since V3: -- Revert compatible type change in dt bindings. -- Update boolean flag description in dt bindings. -- Drop redundant clock optional flag initialization. -- Remove redundant if check. Changes Since V2: -- Remove redundant lpi pincontrol variant data structure and compatible entry. -- Add adsp bypass mode boolean param check. -- Remove compatible name in dt bindings. -- Update dt binding bypass mode boolean param. Changes Since V1: -- Update commit message. Srinivasa Rao Mandadapu (2): dt-bindings: pinctrl: qcom: sc7280: Add boolean param for ADSP bypass platforms pinctrl: qcom: sc7280: Add clock optional check for ADSP bypass targets .../devicetree/bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 5 +++++ drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 2 ++ drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 1 - 3 files changed, 7 insertions(+), 1 deletion(-) -- 2.7.4

4 6

Questions about driver implementation (1 I2S controller to 2 codecs)
by Paul Cercueil 25 Jun '22

25 Jun '22

Hi, I'm trying to add support for a new board that uses the Ingenic JZ4770 SoC. This SoC has a I2S/AC97 controller, and has an internal codec. The controller has a register switch to select either the internal codec, or an external codec; both cannot be enabled at the same time. On this board, the external speakers / audio line are wired to the internal codec, while a HDMI chip (ITE66121) is wired as the external codec. I'm having a hard time trying to figure out how the codec selection switch should be exported. Should it be a regular widget, and the user is responsible for selecting the right codec? Should it be a DAPM, and plugging the HDMI cable auto-enables the switch? Is this configuration (one controller to two codecs) already somehow supported by ALSA? Right now we're using a "simple-audio-card" in the device tree, and trying to add HDMI sound support. I am not even sure if we should be using one sound card with the two codecs, or one sound card per codec, sharing the same I2S controller? Any thoughts? Thanks, -Paul

2 1

Quirks for MacroSilicon MS2100/MS2106
by John Veness 25 Jun '22

25 Jun '22

Hello, This is my first ever kernel work, so I hope I am doing everything correctly. I have a USB analogue AV capture dongle with a MacroSilicon MS2100E chip inside. PID 0x534d, VID 0x0021 (534d:0021). Apparently the MS2106 uses the same PID/VID, so hopefully acts the same, but I don't have one to check. For audio, the MS2100E has the same problems as the MS2109 USB HDMI capture dongle, namely claiming 96kHz mono sound when it's actually 48kHz stereo, and with left/right channel problems. Luckily, the fix is the exact same as that for the MS2109, as first implemented by Marcan (Hector Martin) in July 2020. Below is a patch which is basically a copy and paste of his, with a different VID. I have tested this patch in 5.19.0 RC2. I haven't recruited any other testers. Even with this patch, there is a remaining problem, which is not present in the MS2109. The sound sample values range from 0x0000 to 0x7fff, with silence around 0x4000, i.e. 15-bit-ish audio. This actually sounds OK to the ear (although half as loud as it should be), but looks odd when looking at the waveform, and makes volume meters always think the sound is very loud. To convert to s16le, I can bitshift one bit left, and subtract 32768. I'm told that this isn't something that can or should be done in the kernel, but should be in userspace. Any more advice on how to fix this remaining quirk would be very welcome. Nonetheless, as I say, with the following kernel patch the captured audio certainly sounds right, so is a big improvement and makes these dongles usable: diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 4f56e1784932..853da162fd18 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3802,6 +3802,54 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* + * MacroSilicon MS2100/MS2106 based AV capture cards + * + * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch. + * They also need QUIRK_FLAG_ALIGN_TRANSFER, which makes one wonder if + * they pretend to be 96kHz mono as a workaround for stereo being broken + * by that... + * + * They also have an issue with initial stream alignment that causes the + * channels to be swapped and out of phase, which is dealt with in quirks.c. + */ +{ + USB_AUDIO_DEVICE(0x534d, 0x0021), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MacroSilicon", + .product_name = "MS210x", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 48000, + .rate_max = 48000, + } + }, + { + .ifnum = -1 + } + } + } +}, + /* * MacroSilicon MS2109 based HDMI capture cards * diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e8468f9b007d..a72874bc0936 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1478,6 +1478,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; + case USB_ID(0x534d, 0x0021): /* MacroSilicon MS2100/MS2106 */ case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; @@ -1904,6 +1905,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x413c, 0xa506, /* Dell AE515 sound bar */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x534d, 0x0021, /* MacroSilicon MS2100/MS2106 */ + QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x534d, 0x2109, /* MacroSilicon MS2109 */ QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x1224, 0x2a25, /* Jieli Technology USB PHY 2.0 */ Thanks for reading and considering this patch. John Veness

4 7

[PATCH 00/11] ctl: elem-value: rewrite getter methods
by GitHub pull_request - opened 25 Jun '22

25 Jun '22

alsa-project/alsa-gobject pull request #79 was opened from takaswie: This patchset rewrites getter methods of ALSACtl.ElemValue so that they just return pointer of internal storage instead of copying to user buffer. ``` Takashi Sakamoto (11): timer: fix entry in exported symbol map ctl: elem-value: clear value field at setter method ctl: elem-value: rewrite getter for boolean values ctl: elem-value: rewrite getter for 32 bit signed integer values ctl: elem-value: rewrite getter for enumeration index values ctl: elem-value: rewrite getter for 8 bit unsigned integer values ctl: elem-value: rewrite getter for IEC 60958 channel status ctl: elem-value: rewrite getter for IEC 60958 user data ctl: elem-value: rewrite getter for 64 bit signed integer values ctl: elem-value: update class description update README with compatibility note README.rst | 124 ++++++++++++++++++++++++++++++-- samples/ctl | 11 ++- src/ctl/alsactl.map | 15 ++-- src/ctl/elem-value.c | 152 +++++++++++++++++++++------------------- src/ctl/elem-value.h | 14 ++-- src/timer/alsatimer.map | 3 +- 6 files changed, 218 insertions(+), 101 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/79 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/79.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

sof-soundwire: broken settings for RT711 and RT711-SDCA
by GitHub issues - opened 24 Jun '22

24 Jun '22

alsa-project/alsa-ucm-conf issue #179 was opened from plbossart: [plb@fedora ~]$ alsaucm -c sof-soundwire set _verb HiFi ALSA lib main.c:826:(execute_sequence) unable to execute cset 'name='PGA2.0 2 Master Capture Switch' 0' ALSA lib main.c:2573:(set_verb_user) error: failed to initialize new use case: HiFi There's not such thing as a switch here: amixer -Dhw:0 sget 'PGA2.0 2 Master',0 Simple mixer control 'PGA2.0 2 Master',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 80 Front Left: Capture 80 [100%] [30.00dB] Front Right: Capture 80 [100%] [30.00dB] In addition, it makes no sense to use an SOF control to set the volume/switch on the headset codec side. The volume control should be done in the codec, not in the SOF firmware. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/179 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

UCM for Behringer UMC204HD broken
by GitHub issues - reopened 24 Jun '22

24 Jun '22

alsa-project/alsa-ucm-conf issue #177 was reopened from gregory112: This is related: [https://github.com/alsa-project/alsa-ucm-conf/pull/128](https://github.com/…. I came from this issue I filed in Pipewire: [https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2471](https://git…. So previously a while back, Behringer UMC204HD was detected as a single stereo input channel. This sound card has two inputs. However, just recently, the channel was separated into two mono channels, which was perfect. However, with the most recent update (I don't know if it's pipewire, pipewire-alsa, alsa-card-profiles, or alsa-ucm), it reverted back into the stereo single input (left/right) mode. What can I do to bring back the previous correct settings (apart from the bugs I filed in Pipewire)? I did not modify any ALSA UCM files. Previously, a while back: it is detected as UMC204HD 192k (stereo input). Then: UMC204HD 192k Line 1 (mono) and UMC204HD 192k Line 2 (mono) (this is how it's supposed to be, I think it's from the pull request above). Now: Back to UMC204HD 192k (stereo input). Might also be helpful but I am using pipewire. This is the output of `alsaucm listcards`: ``` ALSA lib ucm_subs.c:799:(uc_mgr_get_substituted_value) variable '${var:DirectCardName}' is not defined in this context! ALSA lib main.c:1541:(snd_use_case_mgr_open) error: failed to import hw:2 use case configuration -22 ALSA lib parser.c:2957:(uc_mgr_scan_master_configs) Unable to open '-hw:2': Invalid argument alsaucm: error failed to get card list: Invalid argument ``` There is an error for hw:2, which I assume is the Behringer. Here is the result of `alsaucm -chw:2 dump text`. ``` ALSA lib ucm_subs.c:799:(uc_mgr_get_substituted_value) variable '${var:DirectCardName}' is not defined in this context! ALSA lib main.c:1541:(snd_use_case_mgr_open) error: failed to import hw:2 use case configuration -22 alsaucm: error failed to open sound card hw:2: Invalid argument ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/177 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

UCM for Behringer UMC204HD not working
by GitHub issues - opened 24 Jun '22

24 Jun '22

alsa-project/alsa-ucm-conf issue #178 was opened from maru-sama: Similar to #177 I have problem with the ucm setting for the UMC204HD. The problem is that apparently no mixer is being created for the device, when trying to open the card with alsamixer it just fails with cannot load mixer controls. the mixer has been created though if I do not missread the output from alsa-info.sh (attached) [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8977069/… I am using alsa-libs 2.0.7 and the latest ucm from master on ubuntu 22.04 to make sure that I have all the minimum requirements. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/178 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: rockchip: i2s: Fix missing error code in rockchip_i2s_probe()
by Jiapeng Chong 24 Jun '22

24 Jun '22

The error code is missing in this code scenario, add the error code '-EINVAL' to the return value 'ret'. This was found by coccicheck: sound/soc/rockchip/rockchip_i2s.c:810 rockchip_i2s_probe() warn: missing error code 'ret'. Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/rockchip/rockchip_i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index d300eee9ddaa..0ed01624a2db 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -807,6 +807,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) i2s->bclk_off = pinctrl_lookup_state(i2s->pinctrl, "bclk_off"); if (IS_ERR_OR_NULL(i2s->bclk_off)) { dev_err(&pdev->dev, "failed to find i2s bclk_off\n"); + ret = -EINVAL; goto err_clk; } } -- 2.20.1.7.g153144c

2 1

[PATCH] ASoC: audio_graph_card2: Fix port numbers in example
by Sascha Hauer 24 Jun '22

24 Jun '22

The example in audio-graph-card2.c has multiple nodes with the same name in it. Change the port numbers to get different names. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/generic/audio-graph-card2.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 77ac4051b8276..d34b29a49268e 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -90,12 +90,12 @@ links indicates connection part of CPU side (= A). ports@0 { (X) (A) mcpu: port@0 { mcpu0_ep: endpoint { remote-endpoint = <&mcodec0_ep>; }; }; (y) port@1 { mcpu1_ep: endpoint { remote-endpoint = <&cpu1_ep>; }; }; -(y) port@1 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; }; +(y) port@2 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; }; }; ports@1 { (X) port@0 { mcodec0_ep: endpoint { remote-endpoint = <&mcpu0_ep>; }; }; -(y) port@0 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; }; -(y) port@1 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; }; +(y) port@1 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; }; +(y) port@2 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; }; }; }; }; -- 2.30.2

2 1

[PATCH v1] ASoC: sgtl5000: Fix noise on shutdown/remove
by Francesco Dolcini 24 Jun '22

24 Jun '22

Put the SGTL5000 in a silent/safe state on shutdown/remove, this is required since the SGTL5000 produces a constant noise on its output after it is configured and its clock is removed. Without this change this is happening every time the module is unbound/removed or from reboot till the clock is enabled again. The issue was experienced on both a Toradex Colibri/Apalis iMX6, but can be easily reproduced everywhere just playing something on the codec and after that removing/unbinding the driver. Fixes: 9b34e6cc3bc2 ("ASoC: Add Freescale SGTL5000 codec support") Signed-off-by: Francesco Dolcini <francesco.dolcini(a)toradex.com> --- sound/soc/codecs/sgtl5000.c | 9 +++++++++ sound/soc/codecs/sgtl5000.h | 1 + 2 files changed, 10 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 2aa48aef6a97..3363d1696ad7 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1795,6 +1795,9 @@ static int sgtl5000_i2c_remove(struct i2c_client *client) { struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + regmap_write(sgtl5000->regmap, SGTL5000_CHIP_DIG_POWER, SGTL5000_DIG_POWER_DEFAULT); + regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, SGTL5000_ANA_POWER_DEFAULT); + clk_disable_unprepare(sgtl5000->mclk); regulator_bulk_disable(sgtl5000->num_supplies, sgtl5000->supplies); regulator_bulk_free(sgtl5000->num_supplies, sgtl5000->supplies); @@ -1802,6 +1805,11 @@ static int sgtl5000_i2c_remove(struct i2c_client *client) return 0; } +static void sgtl5000_i2c_shutdown(struct i2c_client *client) +{ + sgtl5000_i2c_remove(client); +} + static const struct i2c_device_id sgtl5000_id[] = { {"sgtl5000", 0}, {}, @@ -1822,6 +1830,7 @@ static struct i2c_driver sgtl5000_i2c_driver = { }, .probe_new = sgtl5000_i2c_probe, .remove = sgtl5000_i2c_remove, + .shutdown = sgtl5000_i2c_shutdown, .id_table = sgtl5000_id, }; diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 56ec5863f250..3a808c762299 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -80,6 +80,7 @@ /* * SGTL5000_CHIP_DIG_POWER */ +#define SGTL5000_DIG_POWER_DEFAULT 0x0000 #define SGTL5000_ADC_EN 0x0040 #define SGTL5000_DAC_EN 0x0020 #define SGTL5000_DAP_POWERUP 0x0010 -- 2.25.1

3 2

[PATCH] ASoC: cs35l41: Add support for CLSA3541 ACPI device ID
by Cristian Ciocaltea 24 Jun '22

24 Jun '22

Add support for the CLSA3541 ACPI device ID used on Valve's Steam Deck. The driver is fully compatible with the indicated hardware, hence no additional changes are required. Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea(a)collabora.com> --- sound/soc/codecs/cs35l41-spi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs35l41-spi.c b/sound/soc/codecs/cs35l41-spi.c index 9e19c946a66b..5c8bb24909eb 100644 --- a/sound/soc/codecs/cs35l41-spi.c +++ b/sound/soc/codecs/cs35l41-spi.c @@ -74,6 +74,7 @@ MODULE_DEVICE_TABLE(of, cs35l41_of_match); #ifdef CONFIG_ACPI static const struct acpi_device_id cs35l41_acpi_match[] = { { "CSC3541", 0 }, /* Cirrus Logic PnP ID + part ID */ + { "CLSA3541", 0 }, /* Cirrus Logic PnP ID + part ID */ {}, }; MODULE_DEVICE_TABLE(acpi, cs35l41_acpi_match); -- 2.36.1

3 2

[PATCH v4 00/17] ASoC: Intel: haswell and broadwell boards update
by Cezary Rojewski 24 Jun '22

24 Jun '22

A number of patches improving overall quality and readability of haswell.c and broadwell.c source files found in sound/soc/intel/boards. Both files are first renamed and only then actual changes are being incrementally added. The respective names are: hsw_rt5640 and bdw_rt286 to match the pattern found in more recent boards. Most patches bring no functional change - the more impactful patches at are placed the end: Refactor of suspend/resume flow for the bdw_rt286 board by dropping dev->remove() in favour of card->remove() and adjust jack handling to reduce code size slightly by implementing card_set_jack(). The last patch is removing of FE DAI ops. Given the existence of platform FE DAI capabilities (either static declaration or through topology file), this code is redundant. Changes in v4: - just a rebase to fix missed conflicts with for-next Changes in v3: - Patch 16/17 refactoring suspend/resume has been renamed to "Refactor jack handling". Dropped the usage of card->remove() in favor of link->exit() in that very patch Changes in v2: - fixed wording error in patch 02/17 so it correctly mentions 'haswell_rt5640', not 'broadwell_rt286' - decided not to add kernel module names changes to this patchset so the review is not complicated unnecessarily. Will send them separately instead Cezary Rojewski (17): ASoC: Intel: Rename haswell source file to hsw_rt5640 ASoC: Intel: hsw_rt5640: Reword prefixes of all driver members ASoC: Intel: hsw_rt5640: Reword driver name ASoC: Intel: hsw_rt5640: Update code indentation ASoC: Intel: hsw_rt5640: Update file comments ASoC: Intel: hsw_rt5640: Improve probe() function quality ASoC: Intel: hsw_rt5640: Improve hw_params() debug-ability ASoC: Intel: Rename broadwell source file to bdw_rt286 ASoC: Intel: bdw_rt286: Reword prefixes of all driver members ASoC: Intel: bdw_rt286: Reword driver name ASoC: Intel: bdw_rt286: Update code indentation ASoC: Intel: bdw_rt286: Update file comments ASoC: Intel: bdw_rt286: Improve probe() function quality ASoC: Intel: bdw_rt286: Improve hw_params() debug-ability ASoC: Intel: bdw_rt286: Improve codec_init() quality ASoC: Intel: bdw_rt286: Refactor jack handling ASoC: Intel: bdw_rt286: Remove FE DAI ops sound/soc/intel/boards/Kconfig | 4 +- sound/soc/intel/boards/Makefile | 4 +- sound/soc/intel/boards/bdw_rt286.c | 256 +++++++++++++ sound/soc/intel/boards/broadwell.c | 338 ------------------ sound/soc/intel/boards/haswell.c | 202 ----------- sound/soc/intel/boards/hsw_rt5640.c | 176 +++++++++ .../common/soc-acpi-intel-hsw-bdw-match.c | 6 +- 7 files changed, 439 insertions(+), 547 deletions(-) create mode 100644 sound/soc/intel/boards/bdw_rt286.c delete mode 100644 sound/soc/intel/boards/broadwell.c delete mode 100644 sound/soc/intel/boards/haswell.c create mode 100644 sound/soc/intel/boards/hsw_rt5640.c -- 2.25.1

3 24

[PATCH v2 00/17] ASoC: Intel: haswell and broadwell boards update
by Cezary Rojewski 24 Jun '22

24 Jun '22

A number of patches improving overall quality and readability of haswell.c and broadwell.c source files found in sound/soc/intel/boards. Both files are first renamed and only then actual changes are being incrementally added. The respective names are: hsw_rt5640 and bdw_rt286 to match the pattern found in more recent boards. Most patches bring no functional change - the more impactful patches at are placed the end: Refactor of suspend/resume flow for the bdw_rt286 board by dropping dev->remove() in favour of card->remove() and adjust jack handling to reduce code size slightly by implementing card_set_jack(). The last patch is removing of FE DAI ops. Given the existence of platform FE DAI capabilities (either static declaration or through topology file), this code is redundant. Changes in v2: - fixed wording error in patch 02/17 so it correctly mentions 'haswell_rt5640', not 'broadwell_rt286' - decided not to add kernel module names changes to this patchset so the review is not complicated unnecessarily. Will send them separately instead Cezary Rojewski (17): ASoC: Intel: Rename haswell source file to hsw_rt5640 ASoC: Intel: hsw_rt5640: Reword prefixes of all driver members ASoC: Intel: hsw_rt5640: Reword driver name ASoC: Intel: hsw_rt5640: Update code indentation ASoC: Intel: hsw_rt5640: Update file comments ASoC: Intel: hsw_rt5640: Improve probe() function quality ASoC: Intel: hsw_rt5640: Improve hw_params() debug-ability ASoC: Intel: Rename broadwell source file to bdw_rt286 ASoC: Intel: bdw_rt286: Reword prefixes of all driver members ASoC: Intel: bdw_rt286: Reword driver name ASoC: Intel: bdw_rt286: Update code indentation ASoC: Intel: bdw_rt286: Update file comments ASoC: Intel: bdw_rt286: Improve probe() function quality ASoC: Intel: bdw_rt286: Improve hw_params() debug-ability ASoC: Intel: bdw_rt286: Improve codec_link_init() quality ASoC: Intel: bdw_rt286: Refactor suspend/resume ASoC: Intel: bdw_rt286: Remove FE DAI ops sound/soc/intel/boards/Kconfig | 4 +- sound/soc/intel/boards/Makefile | 4 +- sound/soc/intel/boards/bdw_rt286.c | 257 +++++++++++++ sound/soc/intel/boards/broadwell.c | 338 ------------------ sound/soc/intel/boards/haswell.c | 202 ----------- sound/soc/intel/boards/hsw_rt5640.c | 176 +++++++++ .../common/soc-acpi-intel-hsw-bdw-match.c | 6 +- 7 files changed, 440 insertions(+), 547 deletions(-) create mode 100644 sound/soc/intel/boards/bdw_rt286.c delete mode 100644 sound/soc/intel/boards/broadwell.c delete mode 100644 sound/soc/intel/boards/haswell.c create mode 100644 sound/soc/intel/boards/hsw_rt5640.c -- 2.25.1

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[PATCH] ALSA: usb-audio: Workarounds for Behringer UMC 204/404 HD
by Takashi Iwai 24 Jun '22

24 Jun '22

Both Behringer UMC 202 HD and 404 HD need explicit quirks to enable the implicit feedback mode and start the playback stream primarily. The former seems fixing the stuttering and the latter is required for a playback-only case. Note that the "clock source 41 is not valid" error message still appears even after this fix, but it should be only once at probe. The reason of the error is still unknown, but this seems to be mostly harmless as it's a one-off error and the driver retires the clock setup and it succeeds afterwards. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215934 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e8468f9b007d..12ce69b04f63 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1842,6 +1842,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x1395, 0x740a, /* Sennheiser DECT */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x1397, 0x0508, /* Behringer UMC204HD */ + QUIRK_FLAG_PLAYBACK_FIRST | QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x1397, 0x0509, /* Behringer UMC404HD */ + QUIRK_FLAG_PLAYBACK_FIRST | QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x13e5, 0x0001, /* Serato Phono */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x154e, 0x1002, /* Denon DCD-1500RE */ -- 2.35.3

1 0

[PATCH] ALSA: hda/realtek: Add quirk for Clevo L140PU
by Tim Crawford 24 Jun '22

24 Jun '22

Fixes headset detection on Clevo L140PU. Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b937f63d0d09..55d74c546250 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9210,6 +9210,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x70f3, "Clevo NH77DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f4, "Clevo NH77EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f6, "Clevo NH77DPQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x7718, "Clevo L140PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8228, "Clevo NR40BU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8520, "Clevo NH50D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8521, "Clevo NH77D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.35.3

2 1

[PATCH v3 0/2] ASoC/SoundWire: Intel: add sdw BE dai trigger
by Bard Liao 24 Jun '22

24 Jun '22

For SOF IPC4, we need to set pipeline state in BE DAI trigger. @Vinod, Could you ack if the soundwire patch looks good to you? And we can go through the ASoC tree since the change is mainly on ASoC. v2: - Change "#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)" to "if (IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE))" v3: - Change the multi-line comment style in "soundwire: Intel: add trigger callback" patch and take Vinod's Ack. Bard Liao (2): soundwire: Intel: add trigger callback ASoC: SOF: Intel: add trigger callback into sdw_callback drivers/soundwire/intel.c | 9 +++++++++ include/linux/soundwire/sdw_intel.h | 1 + sound/soc/sof/intel/hda-dai.c | 14 +++++++++++--- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/intel/hda.h | 1 + 5 files changed, 23 insertions(+), 4 deletions(-) -- 2.17.1

1 2

UCM for Behringer UMC204HD broken
by GitHub issues - edited 24 Jun '22

24 Jun '22

alsa-project/alsa-ucm-conf issue #177 was edited from gregory112: This is related: [https://github.com/alsa-project/alsa-ucm-conf/pull/128](https://github.com/…. I came from this issue I filed in Pipewire: [https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2471](https://git…. So previously a while back, Behringer UMC204HD was detected as a single stereo input channel. This sound card has two inputs. However, just recently, the channel was separated into two mono channels, which was perfect. However, with the most recent update (I don't know if it's pipewire, pipewire-alsa, alsa-card-profiles, or alsa-ucm), it reverted back into the stereo single input (left/right) mode. What can I do to bring back the previous correct settings (apart from the bugs I filed in Pipewire)? I did not modify any ALSA UCM files. Previously, a while back: it is detected as UMC204HD 192k (stereo input). Then: UMC204HD 192k Line 1 (mono) and UMC204HD 192k Line 2 (mono) (this is how it's supposed to be, I think it's from the pull request above). Now: Back to UMC204HD 192k (stereo input). Might also be helpful but I am using pipewire. This is the output of `alsaucm listcards`: ``` ALSA lib ucm_subs.c:799:(uc_mgr_get_substituted_value) variable '${var:DirectCardName}' is not defined in this context! ALSA lib main.c:1541:(snd_use_case_mgr_open) error: failed to import hw:2 use case configuration -22 ALSA lib parser.c:2957:(uc_mgr_scan_master_configs) Unable to open '-hw:2': Invalid argument alsaucm: error failed to get card list: Invalid argument ``` There is an error for hw:2, which I assume is the Behringer. Here is the result of `alsaucm -chw:2 dump text`. ``` ALSA lib ucm_subs.c:799:(uc_mgr_get_substituted_value) variable '${var:DirectCardName}' is not defined in this context! ALSA lib main.c:1541:(snd_use_case_mgr_open) error: failed to import hw:2 use case configuration -22 alsaucm: error failed to open sound card hw:2: Invalid argument ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/177 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

UCM for Behringer UMC204HD broken
by GitHub issues - edited 24 Jun '22

24 Jun '22

alsa-project/alsa-ucm-conf issue #177 was edited from gregory112: This is related: [https://github.com/alsa-project/alsa-ucm-conf/pull/128](https://github.com/…. I came from this issue I filed in Pipewire: [https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2471](https://git…. So previously a while back, Behringer UMC204HD was detected as a single stereo input channel. This sound card has two inputs. However, just recently, the channel was separated into two mono channels, which was perfect. However, with the most recent update (I don't know if it's pipewire, pipewire-alsa, alsa-card-profiles, or alsa-ucm), it reverted back into the stereo single input (left/right) mode. What can I do to bring back the previous correct settings (apart from the bugs I filed in Pipewire)? I did not modify any ALSA UCM files. Previously, a while back: it is detected as UMC204HD 192k (stereo input). Then: UMC204HD 192k Line 1 (mono) and UMC204HD 192k Line 2 (mono) (this is how it's supposed to be, I think it's from the pull request above). Now: Back to UMC204HD 192k (stereo input). Might also be helpful but I am using pipewire. This is the output of `alsaucm listcards`: ``` ALSA lib ucm_subs.c:799:(uc_mgr_get_substituted_value) variable '${var:DirectCardName}' is not defined in this context! ALSA lib main.c:1541:(snd_use_case_mgr_open) error: failed to import hw:2 use case configuration -22 ALSA lib parser.c:2957:(uc_mgr_scan_master_configs) Unable to open '-hw:2': Invalid argument alsaucm: error failed to get card list: Invalid argument ``` There is an error for hw:2, which I assume is the Behringer. Here is the result of `alsaucm -chw:2 dump text`. ``` ALSA lib ucm_subs.c:799:(uc_mgr_get_substituted_value) variable '${var:DirectCardName}' is not defined in this context! ALSA lib main.c:1541:(snd_use_case_mgr_open) error: failed to import hw:2 use case configuration -22 ALSA lib parser.c:2957:(uc_mgr_scan_master_configs) Unable to open '-hw:2': Invalid argument alsaucm: error failed to get card list: Invalid argument ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/177 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 0/2] ASoC/SoundWire: Intel: add sdw BE dai trigger
by Bard Liao 24 Jun '22

24 Jun '22

For SOF IPC4, we need to set pipeline state in BE DAI trigger. @Vinod, Could you ack if the soundwire patch looks good to you? And we can go through the ASoC tree since the change is mainly on ASoC. v2: - Change "#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)" to "if (IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE))" Bard Liao (2): soundwire: Intel: add trigger callback ASoC: SOF: Intel: add trigger callback into sdw_callback drivers/soundwire/intel.c | 8 ++++++++ include/linux/soundwire/sdw_intel.h | 1 + sound/soc/sof/intel/hda-dai.c | 14 +++++++++++--- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/intel/hda.h | 1 + 5 files changed, 22 insertions(+), 4 deletions(-) -- 2.17.1

2 3

UCM for Behringer UMC204HD broken
by GitHub issues - edited 24 Jun '22

24 Jun '22

alsa-project/alsa-ucm-conf issue #177 was edited from gregory112: This is related: [https://github.com/alsa-project/alsa-ucm-conf/pull/128](https://github.com/…. I came from this issue I filed in Pipewire: [https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2471](https://git…. So previously a while back, Behringer UMC204HD was detected as a single stereo input channel. This sound card has two inputs. However, just recently, the channel was separated into two mono channels, which was perfect. However, with the most recent update (I don't know if it's pipewire, pipewire-alsa, alsa-card-profiles, or alsa-ucm), it reverted back into the stereo single input (left/right) mode. What can I do to bring back the previous correct settings (apart from the bugs I filed in Pipewire)? I did not modify any ALSA UCM files. Previously, a while back: it is detected as UMC204HD 192k (stereo input). Then: UMC204HD 192k Line 1 (mono) and UMC204HD 192k Line 2 (mono) (this is how it's supposed to be, I think it's from the pull request above). Now: Back to UMC204HD 192k (stereo input) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/177 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

UCM for Behringer UMC204HD broken
by GitHub issues - opened 24 Jun '22

24 Jun '22

alsa-project/alsa-ucm-conf issue #177 was opened from gregory112: This is related: [https://github.com/alsa-project/alsa-ucm-conf/pull/128](https://github.com/…. I came from this issue I filed in Pipewire: [https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2471](https://git…. So previously a while back, Behringer UMC204HD was detected as a single stereo input channel. This sound card has two inputs. However, just recently, the channel was separated into two mono channels, which was perfect. However, with the most recent update (I don't know if it's pipewire, pipewire-alsa, alsa-card-profiles, or alsa-ucm), it reverted back into the stereo single input (left/right) mode. What can I do to bring back the previous correct settings (apart from the bugs I filed in Pipewire)? I did not modify any ALSA UCM files. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/177 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 00/11] ASoC: codecs: Series of fixes for realtek codecs used on RVPs
by Amadeusz Sławiński 23 Jun '22

23 Jun '22

Our tests platforms do use realtek codecs, while implementing avs driver and machine boards for it, we identified quite a lot of problems with those codec drivers. This series aims to fix those issues. Amadeusz Sławiński (8): ASoC: codecs: rt274: Move irq registration and cleanup ASoC: codecs: rt286: Move irq registration and cleanup ASoC: codecs: rt298: Move irq registration and cleanup ASoC: codecs: rt274: Enable irq only when needed ASoC: codecs: rt286: Enable irq only when needed ASoC: codecs: rt298: Enable irq only when needed ASoC: codecs: rt298: Fix NULL jack in interrupt ASoC: codecs: rt298: Fix jack detection Cezary Rojewski (3): ASoC: codecs: rt274: Always init jack_detect_work ASoC: codecs: rt286: Reorganize jack detect handling ASoC: codecs: rt298: Reorganize jack detect handling sound/soc/codecs/rt274.c | 67 +++++++++++---------- sound/soc/codecs/rt286.c | 56 ++++++++---------- sound/soc/codecs/rt286.h | 2 - sound/soc/codecs/rt298.c | 93 +++++++++++++----------------- sound/soc/codecs/rt298.h | 2 - sound/soc/intel/boards/broadwell.c | 6 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- 8 files changed, 108 insertions(+), 122 deletions(-) -- 2.25.1

2 22

[PATCH 1/6] ASoC: dapm: Initialise kcontrol data for mux/demux controls
by Charles Keepax 23 Jun '22

23 Jun '22

DAPM keeps a copy of the current value of mux/demux controls, however this value is only initialised in the case of autodisable controls. This leads to false notification events when first modifying a DAPM kcontrol that has a non-zero default. Autodisable controls are left as they are, since they already initialise the value, and there would be more work required to support autodisable muxes where the first option isn't disabled and/or that isn't the default. Technically this issue could affect mixer/switch elements as well, although not on any of the devices I am currently running. There is also a little more work to do to address the issue there due to that side supporting stereo controls, so that has not been tackled in this patch. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-dapm.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 869c76506b669..a8e842e02cdc2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -62,6 +62,8 @@ struct snd_soc_dapm_widget * snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget); +static unsigned int soc_dapm_read(struct snd_soc_dapm_context *dapm, int reg); + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 1, @@ -442,6 +444,9 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, snd_soc_dapm_add_path(widget->dapm, data->widget, widget, NULL, NULL); + } else if (e->reg != SND_SOC_NOPM) { + data->value = soc_dapm_read(widget->dapm, e->reg) & + (e->mask << e->shift_l); } break; default: -- 2.30.2

2 6

[PATCH] ASoC: topology: KUnit: Followup prototype change of snd_soc_unregister_card()
by Uwe Kleine-König 23 Jun '22

23 Jun '22

snd_soc_unregister_card() was recently converted to return void. Only the first instance was adapted, so convert the remaining ones now to fix building the topology test. Reported-by: kernel test robot <lkp(a)intel.com> Fixes: 1892a991886a ("ASoC: core: Make snd_soc_unregister_card() return void") Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- Hello Mark, this fixes the issue the kernel test robot has reported in reply to the patch that became 1892a991886a. Up to you to apply on top of 1892a991886a or squash them together. I'm a bit surprised that sound/soc/soc-topology-test.c isn't built for ARCH=arm allmodconfig (that was my build test and that's the reason I didn't notice the issue before submission of my change). I didn't look into that though. Best regards Uwe sound/soc/soc-topology-test.c | 30 ++++++++++-------------------- 1 file changed, 10 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-topology-test.c b/sound/soc/soc-topology-test.c index 225d74355974..51d650bb05b7 100644 --- a/sound/soc/soc-topology-test.c +++ b/sound/soc/soc-topology-test.c @@ -313,8 +313,7 @@ static void snd_soc_tplg_test_load_with_null_ops(struct kunit *test) KUNIT_EXPECT_EQ(test, 0, ret); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); snd_soc_unregister_component(test_dev); } @@ -377,8 +376,7 @@ static void snd_soc_tplg_test_load_with_null_fw(struct kunit *test) KUNIT_EXPECT_EQ(test, 0, ret); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); snd_soc_unregister_component(test_dev); } @@ -426,8 +424,7 @@ static void snd_soc_tplg_test_load_empty_tplg(struct kunit *test) KUNIT_EXPECT_EQ(test, 0, ret); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); snd_soc_unregister_component(test_dev); } @@ -482,8 +479,7 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_magic(struct kunit *test) KUNIT_EXPECT_EQ(test, 0, ret); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); snd_soc_unregister_component(test_dev); } @@ -538,8 +534,7 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_abi(struct kunit *test) KUNIT_EXPECT_EQ(test, 0, ret); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); snd_soc_unregister_component(test_dev); } @@ -594,8 +589,7 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_size(struct kunit *test) KUNIT_EXPECT_EQ(test, 0, ret); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); snd_soc_unregister_component(test_dev); } @@ -653,8 +647,7 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_payload_size(struct kunit *tes /* cleanup */ snd_soc_unregister_component(test_dev); - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); } // TEST CASE @@ -702,8 +695,7 @@ static void snd_soc_tplg_test_load_pcm_tplg(struct kunit *test) snd_soc_unregister_component(test_dev); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); } // TEST CASE @@ -755,8 +747,7 @@ static void snd_soc_tplg_test_load_pcm_tplg_reload_comp(struct kunit *test) } /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); } // TEST CASE @@ -804,8 +795,7 @@ static void snd_soc_tplg_test_load_pcm_tplg_reload_card(struct kunit *test) if (ret != 0 && ret != -EPROBE_DEFER) KUNIT_FAIL(test, "Failed to register card"); - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); } /* cleanup */ base-commit: 1892a991886ace2c3450bec801df2cf4028a803a -- 2.36.1

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[PATCH v2 14/16] soundwire: Use acpi_dev_for_each_child()
by Rafael J. Wysocki 23 Jun '22

23 Jun '22

From: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> Instead of walking the list of children of an ACPI device directly, use acpi_dev_for_each_child() to carry out an action for all of the given ACPI device's children. This will help to eliminate the children list head from struct acpi_device as it is redundant and it is used in questionable ways in some places (in particular, locking is needed for walking the list pointed to it safely, but it is often missing). Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v1 -> v2: * Make sure errors are not lost (Pierre-Louis). * Add R-by and T-by from Pierre-Louis. --- drivers/soundwire/slave.c | 117 ++++++++++++++++++++++++++-------------------- 1 file changed, 67 insertions(+), 50 deletions(-) Index: linux-pm/drivers/soundwire/slave.c =================================================================== --- linux-pm.orig/drivers/soundwire/slave.c +++ linux-pm/drivers/soundwire/slave.c @@ -127,6 +127,71 @@ static bool find_slave(struct sdw_bus *b return true; } +struct sdw_acpi_child_walk_data { + struct sdw_bus *bus; + struct acpi_device *adev; + struct sdw_slave_id id; + bool ignore_unique_id; +}; + +static int sdw_acpi_check_duplicate(struct acpi_device *adev, void *data) +{ + struct sdw_acpi_child_walk_data *cwd = data; + struct sdw_bus *bus = cwd->bus; + struct sdw_slave_id id; + + if (adev == cwd->adev) + return 0; + + if (!find_slave(bus, adev, &id)) + return 0; + + if (cwd->id.sdw_version != id.sdw_version || cwd->id.mfg_id != id.mfg_id || + cwd->id.part_id != id.part_id || cwd->id.class_id != id.class_id) + return 0; + + if (cwd->id.unique_id != id.unique_id) { + dev_dbg(bus->dev, + "Valid unique IDs 0x%x 0x%x for Slave mfg_id 0x%04x, part_id 0x%04x\n", + cwd->id.unique_id, id.unique_id, cwd->id.mfg_id, + cwd->id.part_id); + cwd->ignore_unique_id = false; + return 0; + } + + dev_err(bus->dev, + "Invalid unique IDs 0x%x 0x%x for Slave mfg_id 0x%04x, part_id 0x%04x\n", + cwd->id.unique_id, id.unique_id, cwd->id.mfg_id, cwd->id.part_id); + return -ENODEV; +} + +static int sdw_acpi_find_one(struct acpi_device *adev, void *data) +{ + struct sdw_bus *bus = data; + struct sdw_acpi_child_walk_data cwd = { + .bus = bus, + .adev = adev, + .ignore_unique_id = true, + }; + int ret; + + if (!find_slave(bus, adev, &cwd.id)) + return 0; + + /* Brute-force O(N^2) search for duplicates. */ + ret = acpi_dev_for_each_child(ACPI_COMPANION(bus->dev), + sdw_acpi_check_duplicate, &cwd); + if (ret) + return ret; + + if (cwd.ignore_unique_id) + cwd.id.unique_id = SDW_IGNORED_UNIQUE_ID; + + /* Ignore errors and continue. */ + sdw_slave_add(bus, &cwd.id, acpi_fwnode_handle(adev)); + return 0; +} + /* * sdw_acpi_find_slaves() - Find Slave devices in Master ACPI node * @bus: SDW bus instance @@ -135,8 +200,7 @@ static bool find_slave(struct sdw_bus *b */ int sdw_acpi_find_slaves(struct sdw_bus *bus) { - struct acpi_device *adev, *parent; - struct acpi_device *adev2, *parent2; + struct acpi_device *parent; parent = ACPI_COMPANION(bus->dev); if (!parent) { @@ -144,54 +208,7 @@ int sdw_acpi_find_slaves(struct sdw_bus return -ENODEV; } - list_for_each_entry(adev, &parent->children, node) { - struct sdw_slave_id id; - struct sdw_slave_id id2; - bool ignore_unique_id = true; - - if (!find_slave(bus, adev, &id)) - continue; - - /* brute-force O(N^2) search for duplicates */ - parent2 = parent; - list_for_each_entry(adev2, &parent2->children, node) { - - if (adev == adev2) - continue; - - if (!find_slave(bus, adev2, &id2)) - continue; - - if (id.sdw_version != id2.sdw_version || - id.mfg_id != id2.mfg_id || - id.part_id != id2.part_id || - id.class_id != id2.class_id) - continue; - - if (id.unique_id != id2.unique_id) { - dev_dbg(bus->dev, - "Valid unique IDs 0x%x 0x%x for Slave mfg_id 0x%04x, part_id 0x%04x\n", - id.unique_id, id2.unique_id, id.mfg_id, id.part_id); - ignore_unique_id = false; - } else { - dev_err(bus->dev, - "Invalid unique IDs 0x%x 0x%x for Slave mfg_id 0x%04x, part_id 0x%04x\n", - id.unique_id, id2.unique_id, id.mfg_id, id.part_id); - return -ENODEV; - } - } - - if (ignore_unique_id) - id.unique_id = SDW_IGNORED_UNIQUE_ID; - - /* - * don't error check for sdw_slave_add as we want to continue - * adding Slaves - */ - sdw_slave_add(bus, &id, acpi_fwnode_handle(adev)); - } - - return 0; + return acpi_dev_for_each_child(parent, sdw_acpi_find_one, bus); } #endif

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[PATCH v5 0/3] ASoC: rockchip: i2s: switch BCLK to GPIO
by Judy Hsiao 23 Jun '22

23 Jun '22

The patches series is to fix the unexpected large DC output voltage of Max98357a that burns the speakers on the rockchip platform when BCLK and SD_MODE are ON but LRCLK is OFF. Changes Since V5: -- Check i2s->pinctrl is valid before using pinctrl_lookup_state. Changes Since V4: -- Fix indentation in the driver. (Align parameters with the parenthesis placement.) -- Fix incorrect return type of rockchip_snd_rxctrl. Changes Since V3: -- Fix indentation in the documentation. -- Put pinctrl-1 right after pinctrl-0 in dtsi. -- Fix indentation in the driver. -- Remove unnecessary dev_dbg() in the driver. Changes Since V2: -- Add documents of i2s pinctrl-names. -- Fix dtsi syntax error. -- Include the dtsi change and the driver change in the same series. -- Ensure that driver gets both bclk_on and bclk_off states before using them. Judy Hsiao (3): ASoC: rockchip: i2s: switch BCLK to GPIO arm64: dts: rk3399: i2s: switch BCLK to GPIO ASoC: dt-bindings: rockchip: Document pinctrl-names for i2s .../bindings/sound/rockchip-i2s.yaml | 7 + .../boot/dts/rockchip/rk3399-gru-scarlet.dtsi | 10 + arch/arm64/boot/dts/rockchip/rk3399.dtsi | 25 ++- sound/soc/rockchip/rockchip_i2s.c | 171 +++++++++++++----- 4 files changed, 165 insertions(+), 48 deletions(-) -- 2.37.0.rc0.104.g0611611a94-goog

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[PATCH 00/96] Refactor non_legacy_dai_naming flag
by Charles Keepax 23 Jun '22

23 Jun '22

Historically, the legacy DAI naming scheme was applied to platform drivers and the newer scheme to CODEC drivers. During componentisation the core lost the knowledge of if a driver was a CODEC or platform, they were all now components. To continue to support the legacy naming on older platform drivers a flag was added to the snd_soc_component_driver structure, non_legacy_dai_naming, to indicate to use the new scheme and this was applied to all CODECs as part of the migration. However, a slight issue appears to be developing with respect to this flag being opt in for the non-legacy scheme, which presumably we want to be the primary scheme used. Many codec drivers appear to forget to include this flag: grep -l -r "snd_soc_component_driver" sound/soc/codecs/*.c | xargs grep -L "non_legacy_dai_naming" | wc 48 48 556 Whilst in many cases the configuration of the DAIs themselves will cause the core to apply the new scheme anyway, it would seem more sensible to change the flag to legacy_dai_naming making the new scheme opt out. This patch series migrates across to such a scheme. Thanks, Charles Charles Keepax (96): ASoC: soc-component: Add legacy_dai_naming flag soundwire: intel: Migrate to new style legacy DAI naming flag drm/vc4: Migrate to new style legacy DAI naming flag ASoC: img: Migrate to new style legacy DAI naming flag ASoC: spear: Migrate to new style legacy DAI naming flag ASoC: jz4740-i2c: Migrate to new style legacy DAI naming flag ASoC: ep93xx: Migrate to new style legacy DAI naming flag ASoC: stm32: Migrate to new style legacy DAI naming flag ASoC: bcm: Migrate to new style legacy DAI naming flag ASoC: sh: Migrate to new style legacy DAI naming flag ASoC: tegra: Migrate to new style legacy DAI naming flag ASoC: hisilicon: Migrate to new style legacy DAI naming flag ASoC: xilinx: Migrate to new style legacy DAI naming flag ASoC: sunxi: Migrate to new style legacy DAI naming flag ASoC: Intel: Migrate to new style legacy DAI naming flag ASoC: meson: Migrate to new style legacy DAI naming flag ASoC: sti-uniperf: Migrate to new style legacy DAI naming flag ASoC: amd: Migrate to new style legacy DAI naming flag ASoC: atmel: Migrate to new style legacy DAI naming flag ASoC: fsl: Migrate to new style legacy DAI naming flag ASoC: xtensa: Migrate to new style legacy DAI naming flag ASoC: adi: Migrate to new style legacy DAI naming flag ASoC: dwc: Migrate to new style legacy DAI naming flag ASoC: qcom: Migrate to new style legacy DAI naming flag ASoC: test-component: Migrate to new style legacy DAI naming flag ASoC: rockchip: Migrate to new style legacy DAI naming flag ASoC: au1x: Migrate to new style legacy DAI naming flag ASoC: pxa: Migrate to new style legacy DAI naming flag ASoC: sof: Migrate to new style legacy DAI naming flag ASoC: ux500: Migrate to new style legacy DAI naming flag ASoC: ti: Migrate to new style legacy DAI naming flag ASoC: mxs-saif: Migrate to new style legacy DAI naming flag ASoC: samsung: Migrate to new style legacy DAI naming flag ASoC: core: Switch core to new DAI naming flag media: TDA1997x: Remove now redundant non_legacy_dai_naming flag ASoC: fsl: Remove now redundant non_legacy_dai_naming flag ASoC: meson: Remove now redundant non_legacy_dai_naming flag ASoC: pistachio: Remove now redundant non_legacy_dai_naming flag ASoC: samsung: Remove now redundant non_legacy_dai_naming flag ASoC: soc-utils: Remove now redundant non_legacy_dai_naming flag ASoC: sunxi: Remove now redundant non_legacy_dai_naming flag ASoC: tegra: Remove now redundant non_legacy_dai_naming flag ASoC: test-component: Remove now redundant non_legacy_dai_naming flag ASoC: topology: KUnit: Remove now redundant non_legacy_dai_naming flag ASoC: uniphier: Remove now redundant non_legacy_dai_naming flag ASoC: ad*: Remove now redundant non_legacy_dai_naming flag ASoC: ak*: Remove now redundant non_legacy_dai_naming flag ASoC: alc*: Remove now redundant non_legacy_dai_naming flag ASoC: cs*: Remove now redundant non_legacy_dai_naming flag ASoC: da*: Remove now redundant non_legacy_dai_naming flag ASoC: es*: Remove now redundant non_legacy_dai_naming flag ASoC: max*: Remove now redundant non_legacy_dai_naming flag ASoC: msm*: Remove now redundant non_legacy_dai_naming flag ASoC: nau*: Remove now redundant non_legacy_dai_naming flag ASoC: pcm*: Remove now redundant non_legacy_dai_naming flag ASoC: rt*: Remove now redundant non_legacy_dai_naming flag ASoC: spdif: Remove now redundant non_legacy_dai_naming flag ASoC: ssm*: Remove now redundant non_legacy_dai_naming flag ASoC: sta*: Remove now redundant non_legacy_dai_naming flag ASoC: tas*: Remove now redundant non_legacy_dai_naming flag ASoC: tfa*: Remove now redundant non_legacy_dai_naming flag ASoC: tlv320*: Remove now redundant non_legacy_dai_naming flag ASoC: twl*: Remove now redundant non_legacy_dai_naming flag ASoC: uda*: Remove now redundant non_legacy_dai_naming flag ASoC: wm*: Remove now redundant non_legacy_dai_naming flag ASoC: 88pm860x: Remove now redundant non_legacy_dai_naming flag ASoC: ab8500: Remove now redundant non_legacy_dai_naming flag ASoC: ac97: Remove now redundant non_legacy_dai_naming flag ASoC: ads117x: Remove now redundant non_legacy_dai_naming flag ASoC: bd28623: Remove now redundant non_legacy_dai_naming flag ASoC: bt-sco: Remove now redundant non_legacy_dai_naming flag ASoC: cpcap: Remove now redundant non_legacy_dai_naming flag ASoC: cq93vc: Remove now redundant non_legacy_dai_naming flag ASoC: cx20442: Remove now redundant non_legacy_dai_naming flag ASoC: dmic: Remove now redundant non_legacy_dai_naming flag ASoC: gtm601: Remove now redundant non_legacy_dai_naming flag ASoC: hdac_hdmi: Remove now redundant non_legacy_dai_naming flag ASoC: hdmi-codec: Remove now redundant non_legacy_dai_naming flag ASoC: ics43432: Remove now redundant non_legacy_dai_naming flag ASoC: inno_rk3036: Remove now redundant non_legacy_dai_naming flag ASoC: Intel: avs: Remove now redundant non_legacy_dai_naming flag ASoC: isabelle: Remove now redundant non_legacy_dai_naming flag ASoC: jz4740: Remove now redundant non_legacy_dai_naming flag ASoC: lm49453: Remove now redundant non_legacy_dai_naming flag ASoC: lochnagar: Remove now redundant non_legacy_dai_naming flag ASoC: mc13783: Remove now redundant non_legacy_dai_naming flag ASoC: ml26124: Remove now redundant non_legacy_dai_naming flag ASoC: rk817: Remove now redundant non_legacy_dai_naming flag ASoC: sgtl5000: Remove now redundant non_legacy_dai_naming flag ASoC: si476x: Remove now redundant non_legacy_dai_naming flag ASoC: stac9766: Remove now redundant non_legacy_dai_naming flag ASoC: sti-sas: Remove now redundant non_legacy_dai_naming flag ASoC: tscs42xx: Remove now redundant non_legacy_dai_naming flag ASoC: wl1273: Remove now redundant non_legacy_dai_naming flag ASoC: zl38060: Remove now redundant non_legacy_dai_naming flag ASoC: soc-component: Remove non_legacy_dai_naming flag drivers/gpu/drm/vc4/vc4_hdmi.c | 1 + drivers/media/i2c/tda1997x.c | 1 - drivers/soundwire/intel.c | 5 +++-- include/sound/soc-component.h | 2 +- sound/soc/adi/axi-i2s.c | 1 + sound/soc/adi/axi-spdif.c | 1 + sound/soc/amd/acp/acp-platform.c | 15 ++++++++------- sound/soc/amd/raven/acp3x-i2s.c | 3 ++- sound/soc/amd/renoir/acp3x-pdm-dma.c | 13 +++++++------ sound/soc/amd/vangogh/acp5x-i2s.c | 1 + sound/soc/amd/yc/acp6x-pdm-dma.c | 13 +++++++------ sound/soc/atmel/atmel-classd.c | 1 + sound/soc/atmel/atmel-i2s.c | 3 ++- sound/soc/atmel/atmel-pdmic.c | 1 + sound/soc/atmel/atmel_ssc_dai.c | 7 ++++--- sound/soc/atmel/mchp-i2s-mcc.c | 3 ++- sound/soc/atmel/mchp-pdmc.c | 1 + sound/soc/atmel/mchp-spdifrx.c | 3 ++- sound/soc/atmel/mchp-spdiftx.c | 3 ++- sound/soc/au1x/ac97c.c | 3 ++- sound/soc/au1x/i2sc.c | 3 ++- sound/soc/au1x/psc-ac97.c | 3 ++- sound/soc/au1x/psc-i2s.c | 3 ++- sound/soc/bcm/bcm2835-i2s.c | 3 ++- sound/soc/bcm/bcm63xx-i2s-whistler.c | 1 + sound/soc/bcm/cygnus-ssp.c | 7 ++++--- sound/soc/cirrus/ep93xx-ac97.c | 3 ++- sound/soc/cirrus/ep93xx-i2s.c | 7 ++++--- sound/soc/codecs/88pm860x-codec.c | 1 - sound/soc/codecs/ab8500-codec.c | 1 - sound/soc/codecs/ac97.c | 1 - sound/soc/codecs/ad1836.c | 1 - sound/soc/codecs/ad193x.c | 1 - sound/soc/codecs/ad1980.c | 1 - sound/soc/codecs/ad73311.c | 1 - sound/soc/codecs/adau1373.c | 1 - sound/soc/codecs/adau1701.c | 1 - sound/soc/codecs/adau1761.c | 1 - sound/soc/codecs/adau1781.c | 1 - sound/soc/codecs/adau1977.c | 1 - sound/soc/codecs/adau7002.c | 1 - sound/soc/codecs/adau7118.c | 1 - sound/soc/codecs/adav80x.c | 1 - sound/soc/codecs/ads117x.c | 1 - sound/soc/codecs/ak4104.c | 1 - sound/soc/codecs/ak4118.c | 1 - sound/soc/codecs/ak4375.c | 1 - sound/soc/codecs/ak4458.c | 2 -- sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ak4554.c | 1 - sound/soc/codecs/ak4613.c | 1 - sound/soc/codecs/ak4641.c | 1 - sound/soc/codecs/ak4642.c | 1 - sound/soc/codecs/ak4671.c | 1 - sound/soc/codecs/ak5386.c | 1 - sound/soc/codecs/ak5558.c | 2 -- sound/soc/codecs/alc5623.c | 1 - sound/soc/codecs/alc5632.c | 1 - sound/soc/codecs/bd28623.c | 1 - sound/soc/codecs/bt-sco.c | 1 - sound/soc/codecs/cpcap.c | 1 - sound/soc/codecs/cq93vc.c | 1 - sound/soc/codecs/cs35l32.c | 1 - sound/soc/codecs/cs35l33.c | 1 - sound/soc/codecs/cs35l34.c | 1 - sound/soc/codecs/cs35l35.c | 1 - sound/soc/codecs/cs35l36.c | 1 - sound/soc/codecs/cs4234.c | 1 - sound/soc/codecs/cs4265.c | 1 - sound/soc/codecs/cs4270.c | 1 - sound/soc/codecs/cs4271.c | 1 - sound/soc/codecs/cs42l42.c | 1 - sound/soc/codecs/cs42l51.c | 1 - sound/soc/codecs/cs42l52.c | 1 - sound/soc/codecs/cs42l56.c | 1 - sound/soc/codecs/cs42l73.c | 1 - sound/soc/codecs/cs42xx8.c | 1 - sound/soc/codecs/cs43130.c | 1 - sound/soc/codecs/cs4341.c | 1 - sound/soc/codecs/cs4349.c | 1 - sound/soc/codecs/cs47l15.c | 1 - sound/soc/codecs/cs47l24.c | 1 - sound/soc/codecs/cs47l35.c | 1 - sound/soc/codecs/cs47l85.c | 1 - sound/soc/codecs/cs47l90.c | 1 - sound/soc/codecs/cs47l92.c | 1 - sound/soc/codecs/cs53l30.c | 1 - sound/soc/codecs/cx20442.c | 1 - sound/soc/codecs/da7210.c | 1 - sound/soc/codecs/da7213.c | 1 - sound/soc/codecs/da7218.c | 1 - sound/soc/codecs/da7219.c | 1 - sound/soc/codecs/da732x.c | 1 - sound/soc/codecs/da9055.c | 1 - sound/soc/codecs/dmic.c | 1 - sound/soc/codecs/es7134.c | 1 - sound/soc/codecs/es7241.c | 1 - sound/soc/codecs/es8316.c | 1 - sound/soc/codecs/es8328.c | 1 - sound/soc/codecs/gtm601.c | 1 - sound/soc/codecs/hdac_hdmi.c | 1 - sound/soc/codecs/hdmi-codec.c | 1 - sound/soc/codecs/ics43432.c | 1 - sound/soc/codecs/inno_rk3036.c | 1 - sound/soc/codecs/isabelle.c | 1 - sound/soc/codecs/jz4740.c | 2 -- sound/soc/codecs/lm49453.c | 1 - sound/soc/codecs/lochnagar-sc.c | 1 - sound/soc/codecs/max98088.c | 1 - sound/soc/codecs/max98090.c | 1 - sound/soc/codecs/max98095.c | 1 - sound/soc/codecs/max98357a.c | 1 - sound/soc/codecs/max98371.c | 1 - sound/soc/codecs/max98373.c | 2 -- sound/soc/codecs/max98390.c | 1 - sound/soc/codecs/max98396.c | 2 -- sound/soc/codecs/max9850.c | 1 - sound/soc/codecs/max98520.c | 1 - sound/soc/codecs/max9860.c | 1 - sound/soc/codecs/max9867.c | 1 - sound/soc/codecs/max98925.c | 1 - sound/soc/codecs/max98926.c | 1 - sound/soc/codecs/max98927.c | 1 - sound/soc/codecs/mc13783.c | 1 - sound/soc/codecs/ml26124.c | 1 - sound/soc/codecs/msm8916-wcd-analog.c | 1 - sound/soc/codecs/msm8916-wcd-digital.c | 1 - sound/soc/codecs/nau8315.c | 1 - sound/soc/codecs/nau8540.c | 1 - sound/soc/codecs/nau8810.c | 1 - sound/soc/codecs/nau8821.c | 1 - sound/soc/codecs/nau8822.c | 1 - sound/soc/codecs/nau8824.c | 1 - sound/soc/codecs/nau8825.c | 1 - sound/soc/codecs/pcm1681.c | 1 - sound/soc/codecs/pcm1789.c | 1 - sound/soc/codecs/pcm179x.c | 1 - sound/soc/codecs/pcm186x.c | 2 -- sound/soc/codecs/pcm3008.c | 1 - sound/soc/codecs/pcm3168a.c | 1 - sound/soc/codecs/pcm5102a.c | 1 - sound/soc/codecs/pcm512x.c | 1 - sound/soc/codecs/rk817_codec.c | 1 - sound/soc/codecs/rt1011.c | 1 - sound/soc/codecs/rt1015.c | 1 - sound/soc/codecs/rt1015p.c | 1 - sound/soc/codecs/rt1016.c | 1 - sound/soc/codecs/rt1019.c | 1 - sound/soc/codecs/rt1305.c | 1 - sound/soc/codecs/rt1308.c | 1 - sound/soc/codecs/rt274.c | 1 - sound/soc/codecs/rt286.c | 1 - sound/soc/codecs/rt298.c | 1 - sound/soc/codecs/rt5514.c | 1 - sound/soc/codecs/rt5616.c | 1 - sound/soc/codecs/rt5631.c | 1 - sound/soc/codecs/rt5640.c | 2 -- sound/soc/codecs/rt5645.c | 1 - sound/soc/codecs/rt5651.c | 1 - sound/soc/codecs/rt5659.c | 1 - sound/soc/codecs/rt5660.c | 1 - sound/soc/codecs/rt5663.c | 1 - sound/soc/codecs/rt5665.c | 1 - sound/soc/codecs/rt5668.c | 1 - sound/soc/codecs/rt5670.c | 1 - sound/soc/codecs/rt5677.c | 1 - sound/soc/codecs/rt5682.c | 1 - sound/soc/codecs/rt5682s.c | 1 - sound/soc/codecs/sgtl5000.c | 1 - sound/soc/codecs/si476x.c | 1 - sound/soc/codecs/spdif_receiver.c | 1 - sound/soc/codecs/spdif_transmitter.c | 1 - sound/soc/codecs/ssm2518.c | 1 - sound/soc/codecs/ssm2602.c | 1 - sound/soc/codecs/ssm4567.c | 1 - sound/soc/codecs/sta32x.c | 1 - sound/soc/codecs/sta350.c | 1 - sound/soc/codecs/sta529.c | 1 - sound/soc/codecs/stac9766.c | 2 -- sound/soc/codecs/sti-sas.c | 1 - sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2562.c | 2 -- sound/soc/codecs/tas2764.c | 1 - sound/soc/codecs/tas2770.c | 1 - sound/soc/codecs/tas5086.c | 1 - sound/soc/codecs/tas571x.c | 1 - sound/soc/codecs/tas5720.c | 2 -- sound/soc/codecs/tas5805m.c | 1 - sound/soc/codecs/tas6424.c | 1 - sound/soc/codecs/tfa9879.c | 1 - sound/soc/codecs/tfa989x.c | 1 - sound/soc/codecs/tlv320adcx140.c | 1 - sound/soc/codecs/tlv320aic23.c | 1 - sound/soc/codecs/tlv320aic26.c | 1 - sound/soc/codecs/tlv320aic31xx.c | 1 - sound/soc/codecs/tlv320aic32x4.c | 2 -- sound/soc/codecs/tlv320aic3x.c | 1 - sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/tscs42xx.c | 1 - sound/soc/codecs/twl4030.c | 1 - sound/soc/codecs/twl6040.c | 1 - sound/soc/codecs/uda1334.c | 1 - sound/soc/codecs/uda134x.c | 1 - sound/soc/codecs/uda1380.c | 1 - sound/soc/codecs/wl1273.c | 1 - sound/soc/codecs/wm0010.c | 1 - sound/soc/codecs/wm1250-ev1.c | 1 - sound/soc/codecs/wm2000.c | 1 - sound/soc/codecs/wm2200.c | 1 - sound/soc/codecs/wm5100.c | 1 - sound/soc/codecs/wm5102.c | 1 - sound/soc/codecs/wm5110.c | 1 - sound/soc/codecs/wm8350.c | 1 - sound/soc/codecs/wm8400.c | 1 - sound/soc/codecs/wm8510.c | 1 - sound/soc/codecs/wm8523.c | 1 - sound/soc/codecs/wm8524.c | 1 - sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8711.c | 1 - sound/soc/codecs/wm8727.c | 1 - sound/soc/codecs/wm8728.c | 1 - sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8737.c | 1 - sound/soc/codecs/wm8741.c | 1 - sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 1 - sound/soc/codecs/wm8770.c | 1 - sound/soc/codecs/wm8776.c | 1 - sound/soc/codecs/wm8782.c | 1 - sound/soc/codecs/wm8804.c | 1 - sound/soc/codecs/wm8900.c | 1 - sound/soc/codecs/wm8903.c | 1 - sound/soc/codecs/wm8904.c | 1 - sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8955.c | 1 - sound/soc/codecs/wm8960.c | 1 - sound/soc/codecs/wm8961.c | 1 - sound/soc/codecs/wm8962.c | 1 - sound/soc/codecs/wm8971.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/codecs/wm8978.c | 1 - sound/soc/codecs/wm8983.c | 1 - sound/soc/codecs/wm8985.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8990.c | 1 - sound/soc/codecs/wm8991.c | 1 - sound/soc/codecs/wm8993.c | 1 - sound/soc/codecs/wm8994.c | 1 - sound/soc/codecs/wm8995.c | 1 - sound/soc/codecs/wm8996.c | 2 -- sound/soc/codecs/wm8997.c | 1 - sound/soc/codecs/wm8998.c | 1 - sound/soc/codecs/wm9081.c | 1 - sound/soc/codecs/wm9090.c | 1 - sound/soc/codecs/wm9705.c | 1 - sound/soc/codecs/wm9712.c | 1 - sound/soc/codecs/wm9713.c | 1 - sound/soc/codecs/zl38060.c | 1 - sound/soc/dwc/dwc-i2s.c | 7 ++++--- sound/soc/fsl/fsl_aud2htx.c | 3 ++- sound/soc/fsl/fsl_easrc.c | 7 ++++--- sound/soc/fsl/fsl_esai.c | 3 ++- sound/soc/fsl/fsl_mqs.c | 1 - sound/soc/fsl/fsl_rpmsg.c | 3 ++- sound/soc/fsl/fsl_sai.c | 3 ++- sound/soc/fsl/fsl_spdif.c | 3 ++- sound/soc/fsl/fsl_ssi.c | 1 + sound/soc/fsl/fsl_xcvr.c | 3 ++- sound/soc/fsl/mpc5200_psc_i2s.c | 3 ++- sound/soc/generic/test-component.c | 2 +- sound/soc/hisilicon/hi6210-i2s.c | 1 + sound/soc/img/img-i2s-in.c | 3 ++- sound/soc/img/img-i2s-out.c | 3 ++- sound/soc/img/img-parallel-out.c | 3 ++- sound/soc/img/img-spdif-in.c | 3 ++- sound/soc/img/img-spdif-out.c | 3 ++- sound/soc/img/pistachio-internal-dac.c | 1 - sound/soc/intel/avs/pcm.c | 2 -- sound/soc/intel/keembay/kmb_platform.c | 14 ++++++++------ sound/soc/jz4740/jz4740-i2s.c | 7 ++++--- sound/soc/meson/aiu-acodec-ctrl.c | 1 - sound/soc/meson/aiu-codec-ctrl.c | 1 - sound/soc/meson/axg-frddr.c | 3 +++ sound/soc/meson/axg-pdm.c | 4 +++- sound/soc/meson/axg-spdifin.c | 1 + sound/soc/meson/axg-spdifout.c | 1 + sound/soc/meson/axg-toddr.c | 3 +++ sound/soc/meson/g12a-toacodec.c | 2 -- sound/soc/meson/g12a-tohdmitx.c | 1 - sound/soc/meson/t9015.c | 1 - sound/soc/mxs/mxs-saif.c | 3 ++- sound/soc/pxa/mmp-sspa.c | 9 +++++---- sound/soc/pxa/pxa-ssp.c | 21 +++++++++++---------- sound/soc/pxa/pxa2xx-i2s.c | 21 +++++++++++---------- sound/soc/qcom/lpass-cpu.c | 1 + sound/soc/qcom/qdsp6/q6asm-dai.c | 23 ++++++++++++----------- sound/soc/rockchip/rockchip_i2s.c | 1 + sound/soc/rockchip/rockchip_i2s_tdm.c | 1 + sound/soc/rockchip/rockchip_pdm.c | 1 + sound/soc/rockchip/rockchip_spdif.c | 1 + sound/soc/samsung/aries_wm8994.c | 1 - sound/soc/samsung/i2s.c | 2 ++ sound/soc/samsung/pcm.c | 3 ++- sound/soc/samsung/s3c2412-i2s.c | 7 ++++--- sound/soc/samsung/s3c24xx-i2s.c | 7 ++++--- sound/soc/samsung/spdif.c | 7 ++++--- sound/soc/sh/hac.c | 3 ++- sound/soc/sh/rcar/core.c | 11 ++++++----- sound/soc/sh/rz-ssi.c | 9 +++++---- sound/soc/sh/siu_pcm.c | 17 +++++++++-------- sound/soc/sh/ssi.c | 3 ++- sound/soc/soc-core.c | 2 +- sound/soc/soc-topology-test.c | 3 --- sound/soc/soc-utils.c | 1 - sound/soc/sof/pcm.c | 2 ++ sound/soc/sof/sof-client-probes.c | 1 + sound/soc/spear/spdif_in.c | 3 ++- sound/soc/spear/spdif_out.c | 3 ++- sound/soc/sti/sti_uniperif.c | 3 ++- sound/soc/stm/stm32_adfsdm.c | 1 + sound/soc/stm/stm32_i2s.c | 1 + sound/soc/stm/stm32_sai_sub.c | 1 + sound/soc/stm/stm32_spdifrx.c | 1 + sound/soc/sunxi/sun4i-codec.c | 7 ++----- sound/soc/sunxi/sun4i-i2s.c | 3 ++- sound/soc/sunxi/sun4i-spdif.c | 3 ++- sound/soc/sunxi/sun8i-codec.c | 1 - sound/soc/tegra/tegra20_ac97.c | 3 ++- sound/soc/tegra/tegra20_i2s.c | 3 ++- sound/soc/tegra/tegra20_spdif.c | 1 + sound/soc/tegra/tegra210_i2s.c | 1 - sound/soc/tegra/tegra30_i2s.c | 3 ++- sound/soc/ti/davinci-i2s.c | 3 ++- sound/soc/ti/davinci-mcasp.c | 3 ++- sound/soc/ti/davinci-vcif.c | 3 ++- sound/soc/ti/omap-dmic.c | 3 ++- sound/soc/ti/omap-hdmi.c | 1 + sound/soc/ti/omap-mcbsp.c | 3 ++- sound/soc/ti/omap-mcpdm.c | 7 ++++--- sound/soc/uniphier/evea.c | 1 - sound/soc/ux500/ux500_msp_dai.c | 3 ++- sound/soc/xilinx/xlnx_formatter_pcm.c | 16 ++++++++-------- sound/soc/xilinx/xlnx_i2s.c | 1 + sound/soc/xilinx/xlnx_spdif.c | 1 + sound/soc/xtensa/xtfpga-i2s.c | 15 ++++++++------- 345 files changed, 274 insertions(+), 435 deletions(-) -- 2.30.2

4 100

[PATCH] ASoC: core: Make snd_soc_unregister_card() return void
by Uwe Kleine-König 23 Jun '22

23 Jun '22

The function snd_soc_unregister_card() returned 0 unconditionally and most callers don't care to check the return value. Make it return void and adapt the callers that didn't ignore the return value before. This is a preparation for making platform remove callbacks return void. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- include/sound/soc.h | 2 +- sound/soc/atmel/mikroe-proto.c | 4 +++- sound/soc/fsl/pcm030-audio-fabric.c | 5 ++--- sound/soc/soc-core.c | 4 +--- sound/soc/soc-topology-test.c | 4 +--- 5 files changed, 8 insertions(+), 11 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f20f5f890794..adcc756fe0c2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -416,7 +416,7 @@ enum snd_soc_pcm_subclass { }; int snd_soc_register_card(struct snd_soc_card *card); -int snd_soc_unregister_card(struct snd_soc_card *card); +void snd_soc_unregister_card(struct snd_soc_card *card); int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card); #ifdef CONFIG_PM_SLEEP int snd_soc_suspend(struct device *dev); diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c index ce46d8a0b7e4..954460719aa3 100644 --- a/sound/soc/atmel/mikroe-proto.c +++ b/sound/soc/atmel/mikroe-proto.c @@ -157,7 +157,9 @@ static int snd_proto_probe(struct platform_device *pdev) static int snd_proto_remove(struct platform_device *pdev) { - return snd_soc_unregister_card(&snd_proto); + snd_soc_unregister_card(&snd_proto); + + return 0; } static const struct of_device_id snd_proto_of_match[] = { diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 83b4a22bf15a..e4c805acc349 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -113,12 +113,11 @@ static int pcm030_fabric_probe(struct platform_device *op) static int pcm030_fabric_remove(struct platform_device *op) { struct pcm030_audio_data *pdata = platform_get_drvdata(op); - int ret; - ret = snd_soc_unregister_card(pdata->card); + snd_soc_unregister_card(pdata->card); platform_device_unregister(pdata->codec_device); - return ret; + return 0; } static const struct of_device_id pcm030_audio_match[] = { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9574f86dd4de..c5836066ead3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2326,14 +2326,12 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card); * @card: Card to unregister * */ -int snd_soc_unregister_card(struct snd_soc_card *card) +void snd_soc_unregister_card(struct snd_soc_card *card) { mutex_lock(&client_mutex); snd_soc_unbind_card(card, true); mutex_unlock(&client_mutex); dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); - - return 0; } EXPORT_SYMBOL_GPL(snd_soc_unregister_card); diff --git a/sound/soc/soc-topology-test.c b/sound/soc/soc-topology-test.c index ae3968161509..225d74355974 100644 --- a/sound/soc/soc-topology-test.c +++ b/sound/soc/soc-topology-test.c @@ -271,9 +271,7 @@ static void snd_soc_tplg_test_load_with_null_comp(struct kunit *test) KUNIT_EXPECT_EQ(test, 0, ret); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); - + snd_soc_unregister_card(&kunit_comp->card); snd_soc_unregister_component(test_dev); } base-commit: f2906aa863381afb0015a9eb7fefad885d4e5a56 -- 2.36.1

3 4

[PATCH 00/14] Canaan devicetree fixes
by Conor Dooley 23 Jun '22

23 Jun '22

From: Conor Dooley <conor.dooley(a)microchip.com> Hey all, This series should rid us of dtbs_check errors for the RISC-V Canaan k210 based boards (well, unless you enable W=1 but that's another days work). I *DO NOT* have any Canaan hardware so I have not tested any of this in anger. I based the series on next-20220617. For the bindings, I am never sure about which of {unevaluated,additional} Properties is correct to use, but the if statements in the binding didn't work with additional so I used unevaluated... @Mark, for your two bindings I was not sure about the properties that I made depend on the compatible, but I looked in tree and was not able to find other users to contradict what's in the Canaan devicetrees nor did I get that much help from their docs. @Rob, yesterday's removal of ilitek,ili9341.txt is moved to ths series since I was editing the dt-schema binding here anyway. Finally, @Palmer: This + Atul's stuff + the sifive dts watchdog patch will get us sorted in terms of dtbs_check errors. To make keeping it that way a little easier, I changed the Canaan devicetree Makefile so that it would build all of the devicetrees in the directory if SOC_CANAAN. Hopefully someone with a device can test it - but my build log *looked* fine but that's not exactly sufficient. Thanks, Conor. Conor Dooley (14): dt-bindings: display: convert ilitek,ili9341.txt to dt-schema dt-bindings: display: panel: allow ilitek,ili9341 in isolation ASoC: dt-bindings: convert designware-i2s to dt-schema dt-bindings: dma: add Canaan k210 to Synopsys DesignWare DMA dt-bindings: timer: add Canaan k210 to Synopsys DesignWare timer spi: dt-bindings: dw-apb-ssi: update spi-{r,t}x-bus-width for dwc-ssi riscv: dts: canaan: fix the k210's memory node riscv: dts: canaan: add a specific compatible for k210's dma riscv: dts: canaan: add a specific compatible for k210's timers riscv: dts: canaan: fix mmc node names riscv: dts: canaan: fix kd233 display spi frequency riscv: dts: canaan: use custom compatible for k210 i2s riscv: dts: canaan: remove spi-max-frequency from controllers riscv: dts: canaan: build all devicetress if SOC_CANAAN .../bindings/display/ilitek,ili9341.txt | 27 ------ .../display/panel/ilitek,ili9341.yaml | 60 ++++++++---- .../bindings/dma/snps,dw-axi-dmac.yaml | 35 +++++-- .../bindings/sound/designware-i2s.txt | 35 ------- .../bindings/sound/snps,designware-i2s.yaml | 93 +++++++++++++++++++ .../bindings/spi/snps,dw-apb-ssi.yaml | 48 +++++++--- .../bindings/timer/snps,dw-apb-timer.yaml | 28 ++++-- arch/riscv/boot/dts/canaan/Makefile | 10 +- arch/riscv/boot/dts/canaan/canaan_kd233.dts | 4 +- arch/riscv/boot/dts/canaan/k210.dtsi | 25 ++--- .../riscv/boot/dts/canaan/sipeed_maix_bit.dts | 2 +- .../boot/dts/canaan/sipeed_maix_dock.dts | 2 +- arch/riscv/boot/dts/canaan/sipeed_maix_go.dts | 2 +- .../boot/dts/canaan/sipeed_maixduino.dts | 2 +- 14 files changed, 239 insertions(+), 134 deletions(-) delete mode 100644 Documentation/devicetree/bindings/display/ilitek,ili9341.txt delete mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml base-commit: 07dc787be2316e243a16a33d0a9b734cd9365bd3 -- 2.36.1

5 30

[PATCH v4 0/4] Add MT8186 ADSP dt-binding
by Tinghan Shen 23 Jun '22

23 Jun '22

v3 -> v4: 1. Update commit message of patch 1/4 2. Add review tag to path 3/4 v2 -> v3: 1. Change mbox-names to rx/tx for both mt8186 and mt8195. 2. Update description of mbox-names 3. Use static string array instead of kasprintf 4. Align clock names in dsp driver with dt-bindings v1 -> v2: 1. Change mbox-names from mbox0/mbox1 to rep/req for both mt8186 and mt8195. 2. rename clock-names and remove unused headers Tinghan Shen (4): dt-bindings: dsp: mediatek: Use meaningful names for mbox firmware: mediatek: Use meaningful names for mbox dt-bindings: dsp: mediatek: Add mt8186 dsp document ASoC: SOF: mediatek: Align mt8186 clock names with dt-bindings .../bindings/dsp/mediatek,mt8186-dsp.yaml | 91 +++++++++++++++++++ .../bindings/dsp/mediatek,mt8195-dsp.yaml | 10 +- drivers/firmware/mtk-adsp-ipc.c | 36 +++----- sound/soc/sof/mediatek/mt8186/mt8186-clk.c | 4 +- 4 files changed, 110 insertions(+), 31 deletions(-) create mode 100644 Documentation/devicetree/bindings/dsp/mediatek,mt8186-dsp.yaml -- 2.18.0

3 6

[PATCH 00/49] regmap-irq cleanups and refactoring
by Aidan MacDonald 23 Jun '22

23 Jun '22

Hi Mark, Here's a bunch of cleanups for regmap-irq focused on simplifying the API and generalizing it a bit. It's broken up into three refactors, focusing on one area at a time. * Patches 01 and 02 are straightforward bugfixes, independent of the rest of the series. Neither of the bugs are triggered by in-tree drivers but they might be worth picking up early anyhow. * Patches 03-13 clean up everything related to configuring IRQ types. * Patches 14-45 deal with mask/unmask registers. First, make unmask registers behave more intuitively and usefully, and get rid of the mask_invert flag in favor of describing inverted mask registers as unmask registers. Second, make the mask_writeonly flag more useful and enable it for two chips where it makes sense. * Patches 46-49 refactor sub_irq_reg() as a get_irq_reg() callback, and use that to eliminate the not_fixed_stride flag. The approach I used when refactoring is pretty simple: (1) introduce new functionality in regmap-irq, (2) convert the drivers, and (3) remove any old code. Nothing should break in the middle. The patches can be re-ordered to some extent if that's preferable, but it's best to add get_irq_reg() last to avoid having to think about how it interacts with features that'll be removed anyway. I can't test most of the devices affected by this series so a lot of the code is only build tested. I've tested on real hardware with my AXP192 patchset[1], although it only provides limited code coverage. qcom-pm8008 in particular deserves careful testing - it used all of the features touched by the refactors and required the most changes. Other drivers only required trivial changes but there are three of them worth mentioning: wcd943x, wcd9335, and wcd938x. They have suspicious looking IRQ type definitions and I'm pretty sure aren't working properly, but I can't fix them myself. The refactor shouldn't affect their behavior so how / when / if they get fixed shouldn't be much of an issue. Oh, and I added the 'mask_writeonly' flag and volatile ranges to the stpmic1 driver based on its datasheet[2] as a small optimization. It's probably fine but testing would be a good idea. [1]: https://lore.kernel.org/linux-iio/20220618214009.2178567-1-aidanmacdonald.0… [2]: https://www.st.com/resource/en/datasheet/stpmic1.pdf Aidan MacDonald (49): regmap-irq: Fix a bug in regmap_irq_enable() for type_in_mask chips regmap-irq: Fix offset/index mismatch in read_sub_irq_data() regmap-irq: Remove an unnecessary restriction on type_in_mask regmap-irq: Introduce config registers for irq types mfd: qcom-pm8008: Convert irq chip to config regs mfd: wcd934x: Convert irq chip to config regs sound: soc: codecs: wcd9335: Convert irq chip to config regs sound: soc: codecs: wcd938x: Remove spurious type_base from irq chip mfd: max77650: Remove useless type_invert flag regmap-irq: Remove virtual registers support regmap-irq: Remove old type register support, refactor regmap-irq: Remove unused type_reg_stride field regmap-irq: Remove unused type_invert flag regmap-irq: Do not use regmap_irq_update_bits() for wake regs regmap-irq: Change the behavior of mask_writeonly regmap-irq: Rename regmap_irq_update_bits() regmap-irq: Add broken_mask_unmask flag mfd: qcom-pm8008: Add broken_mask_unmask irq chip flag mfd: stpmic1: Add broken_mask_unmask irq chip flag regmap-irq: Fix inverted handling of unmask registers mfd: tps65090: replace irqchip mask_invert with unmask_base mfd: sun4i-gpadc: replace irqchip mask_invert with unmask_base mfd: sprd-sc27xx-spi: replace irqchip mask_invert with unmask_base mfd: rt5033: replace irqchip mask_invert with unmask_base mfd: rohm-bd71828: replace irqchip mask_invert with unmask_base mfd: rn5t618: replace irqchip mask_invert with unmask_base mfd: gateworks-gsc: replace irqchip mask_invert with unmask_base mfd: axp20x: replace irqchip mask_invert with unmask_base mfd: atc260x: replace irqchip mask_invert with unmask_base mfd: 88pm800: replace irqchip mask_invert with unmask_base mfd: max14577: replace irqchip mask_invert with unmask_base mfd: max77693: replace irqchip mask_invert with unmask_base mfd: rohm-bd718x7: drop useless mask_invert flag on irqchip mfd: max77843: drop useless mask_invert flag on irqchip extcon: max77843: replace irqchip mask_invert with unmask_base extcon: sm5502: drop useless mask_invert flag on irqchip extcon: rt8973a: drop useless mask_invert flag on irqchip irqchip: sl28cpld: replace irqchip mask_invert with unmask_base gpio: sl28cpld: replace irqchip mask_invert with unmask_base mfd: stpmic1: Fix broken mask/unmask in irq chip mfd: stpmic1: Enable mask_writeonly flag for irq chip mfd: qcom-pm8008: Fix broken mask/unmask in irq chip mfd: qcom-pm8008: Enable mask_writeonly flag for irq chip regmap-irq: Remove broken_mask_unmask flag regmap-irq: Remove mask_invert flag regmap-irq: Refactor checks for status bulk read support regmap-irq: Add get_irq_reg() callback mfd: qcom-pm8008: Use get_irq_reg() for irq chip regmap-irq: Remove not_fixed_stride flag drivers/base/regmap/regmap-irq.c | 457 ++++++++++++++----------------- drivers/extcon/extcon-max77843.c | 3 +- drivers/extcon/extcon-rt8973a.c | 1 - drivers/extcon/extcon-sm5502.c | 2 - drivers/gpio/gpio-sl28cpld.c | 3 +- drivers/irqchip/irq-sl28cpld.c | 3 +- drivers/mfd/88pm800.c | 3 +- drivers/mfd/atc260x-core.c | 6 +- drivers/mfd/axp20x.c | 21 +- drivers/mfd/gateworks-gsc.c | 3 +- drivers/mfd/max14577.c | 7 +- drivers/mfd/max77650.c | 1 - drivers/mfd/max77693.c | 6 +- drivers/mfd/max77843.c | 1 - drivers/mfd/qcom-pm8008.c | 131 ++++----- drivers/mfd/rn5t618.c | 3 +- drivers/mfd/rohm-bd71828.c | 6 +- drivers/mfd/rohm-bd718x7.c | 1 - drivers/mfd/rt5033.c | 3 +- drivers/mfd/sprd-sc27xx-spi.c | 3 +- drivers/mfd/stpmic1.c | 7 +- drivers/mfd/sun4i-gpadc.c | 3 +- drivers/mfd/tps65090.c | 3 +- drivers/mfd/wcd934x.c | 11 +- include/linux/regmap.h | 59 ++-- sound/soc/codecs/wcd9335.c | 10 +- sound/soc/codecs/wcd938x.c | 1 - 27 files changed, 332 insertions(+), 426 deletions(-) -- 2.35.1

6 68

[PATCH v1 0/2] Read _SUB from ACPI to be able to identify firmware
by Stefan Binding 23 Jun '22

23 Jun '22

CS35L41 has a DSP which is able to run firmware, as well as a tuning file. Different systems may want to use different firmwares and tuning files, and some firmwares/tunings may not be compatible with other systems. To allow a system to select the correct fimware/tuning, we can read an _SUB from the ACPI. This _SUB can then be used to uniquely identify the system in the firmware/tuning file name. Add a helper function which reads the _SUB, so this can be used by other parts in the future. Add support inside the CS35L41 ASoC driver to read this _SUB, and save it appropriately. Stefan Binding (2): ACPI: utils: Add api to read _SUB from ACPI ASoC: cs35l41: Read System Name from ACPI _SUB to identify firmware drivers/acpi/utils.c | 26 ++++++++++++++++++++++++++ include/linux/acpi.h | 8 ++++++++ sound/soc/codecs/cs35l41.c | 27 +++++++++++++++++++++++++++ 3 files changed, 61 insertions(+) -- 2.25.1

3 6

ASoC: SOF: Race condition in ipc.c
by noman pouigt 22 Jun '22

22 Jun '22

Folks, I have borrowed part of SOF architecture for my own DSP framework development as the memory on the DSP is extremely small and wouldn't be able to support SOF. Currently I am running into a race condition as below: CPU DSP PCM_TRIGGER_START sof_ipc_send_msg ----> <------Immediate ACK ipc3_wait_tx_done (wait_event_timeout) <------ POSITION update snd_pcm_period_elapsed As you can see TRIGGER_START didn't even finish and waiting for it to complete in ipc3_wait_tx_done function. However, before it could complete the position interrupt was issued which results in calling period_elapsed function. In order to fix this I assume below is called in SOF framework: schedule_work(&spcm->stream[substream->stream].period_elapsed_work); How is this design working? If the interrupt is coming too fast from the DSP than the associated function with this schedule_work will not get called as the scheduler will not get time to schedule the workqueue and elapsed function will not be called thereby not increasing the hw_ptr. How is the flow control for data transfer achieved? I am facing the above problem in my design. I am wondering if I can simply add one more IPC command(don't call wait_event_interruptible for this) after TRIGGER_START to start the streaming.This way schedule_work for updating period_elapsed function can be avoided and it can be called in an interrupt context. Thanks, Variksla

2 2

[PATCH] ASoC: amd: acp: Fix error handling in .remove()
by Uwe Kleine-König 22 Jun '22

22 Jun '22

Even in the presence of problems (here: rn_acp_deinit() might fail), it's important to unregister all resources acquired during .probe() because even if .remove() returns an error code, the device is removed. As .remove() is only called after .probe() returned success, platdata must be valid, so the first check in .remove() can just be dropped. This is a preparation for making platform remove callbacks return void. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/amd/acp/acp-renoir.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/amd/acp/acp-renoir.c b/sound/soc/amd/acp/acp-renoir.c index 75c9229ece97..8375c00ff4c3 100644 --- a/sound/soc/amd/acp/acp-renoir.c +++ b/sound/soc/amd/acp/acp-renoir.c @@ -307,16 +307,10 @@ static int renoir_audio_remove(struct platform_device *pdev) int ret; chip = dev_get_platdata(&pdev->dev); - if (!chip || !chip->base) { - dev_err(&pdev->dev, "ACP chip data is NULL\n"); - return -ENODEV; - } ret = rn_acp_deinit(chip->base); - if (ret) { - dev_err(&pdev->dev, "ACP de-init Failed\n"); - return -EINVAL; - } + if (ret) + dev_err(&pdev->dev, "ACP de-init Failed (%pe)\n", ERR_PTR(ret)); acp_platform_unregister(dev); return 0; base-commit: f2906aa863381afb0015a9eb7fefad885d4e5a56 -- 2.36.1

2 1

[PATCH] ASoC: rockchip: i2s: Fix crash on missing pinctrl
by Chen-Yu Tsai 22 Jun '22

22 Jun '22

From: Chen-Yu Tsai <wens(a)csie.org> Commit 44f362c2cc6d ("ASoC: rockchip: i2s: switch BCLK to GPIO") added pinctrl lookups, but did not skip the lookup if there was no pinctrl device tied to the I2S controller. As a result, the lookup was done on an invalid pointer in such cases, causing a kernel panic. Only do the subsequent pinctrl state lookups and switch if a pinctrl device was found. i2s_pinctrl_select_bclk_{on,off} already guard against missing pinctrl device or pinctrl state, so those two functions aren't touched. Fixes: 44f362c2cc6d ("ASoC: rockchip: i2s: switch BCLK to GPIO") Signed-off-by: Chen-Yu Tsai <wens(a)csie.org> --- This is based on next-20220621, which just saw the inclusion of the commit this patch fixes. This issue was found and tested on ROC-RK3328-CC. Tried to test on RK3399, but that has some other DRM driver related crash. sound/soc/rockchip/rockchip_i2s.c | 29 ++++++++++++++--------------- 1 file changed, 14 insertions(+), 15 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index be051e48b97b..9fa8ffd712ea 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -808,24 +808,23 @@ static int rockchip_i2s_probe(struct platform_device *pdev) i2s->bclk_ratio = 64; i2s->pinctrl = devm_pinctrl_get(&pdev->dev); - if (IS_ERR(i2s->pinctrl)) + if (IS_ERR(i2s->pinctrl)) { dev_err(&pdev->dev, "failed to find i2s pinctrl\n"); + } else { + i2s->bclk_on = pinctrl_lookup_state(i2s->pinctrl, "bclk_on"); + if (IS_ERR_OR_NULL(i2s->bclk_on)) + dev_err(&pdev->dev, "failed to find i2s default state\n"); + else + dev_dbg(&pdev->dev, "find i2s bclk state\n"); - i2s->bclk_on = pinctrl_lookup_state(i2s->pinctrl, - "bclk_on"); - if (IS_ERR_OR_NULL(i2s->bclk_on)) - dev_err(&pdev->dev, "failed to find i2s default state\n"); - else - dev_dbg(&pdev->dev, "find i2s bclk state\n"); - - i2s->bclk_off = pinctrl_lookup_state(i2s->pinctrl, - "bclk_off"); - if (IS_ERR_OR_NULL(i2s->bclk_off)) - dev_err(&pdev->dev, "failed to find i2s gpio state\n"); - else - dev_dbg(&pdev->dev, "find i2s bclk_off state\n"); + i2s->bclk_off = pinctrl_lookup_state(i2s->pinctrl, "bclk_off"); + if (IS_ERR_OR_NULL(i2s->bclk_off)) + dev_err(&pdev->dev, "failed to find i2s gpio state\n"); + else + dev_dbg(&pdev->dev, "find i2s bclk_off state\n"); - i2s_pinctrl_select_bclk_off(i2s); + i2s_pinctrl_select_bclk_off(i2s); + } i2s->playback_dma_data.addr = res->start + I2S_TXDR; i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; -- 2.34.1

2 1

Add support for the Traktor Kontrol Z1
by GitHub pull_request - opened 22 Jun '22

22 Jun '22

alsa-project/alsa-ucm-conf pull request #176 was opened from g2p: This is a small DJ mixer with a main output and a monitor output. The outputs appear as a single subdevice, we use the SplitPCM macros to present separate outputs. https://www.native-instruments.com/en/products/traktor/dj-controllers/trakt… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/176 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/176.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda/realtek: Add quirk for Clevo NS50PU
by Tim Crawford 22 Jun '22

22 Jun '22

Fixes headset detection on Clevo NS50PU. Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b937f63d0d09..d4f7ce6d84c3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9210,6 +9210,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x70f3, "Clevo NH77DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f4, "Clevo NH77EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f6, "Clevo NH77DPQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x7716, "Clevo NS50PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8228, "Clevo NR40BU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8520, "Clevo NH50D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8521, "Clevo NH77D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.35.3

2 1

[PATCH 0/3] ASoC: dt-bindings: Add bindings for WCD9335 DAIs
by Yassine Oudjana 22 Jun '22

22 Jun '22

Add DT bindings for WCD9335 DAIs and use them in the driver as well as all device trees currently using WCD9335. Yassine Oudjana (3): ASoC: dt-bindings: Add bindings for WCD9335 DAIs ASoC: wcd9335: Use DT bindings instead of local DAI definitions arm64: dts: qcom: Use WCD9335 DT bindings MAINTAINERS | 1 + arch/arm64/boot/dts/qcom/apq8096-db820c.dts | 5 +++-- .../arm64/boot/dts/qcom/msm8996-xiaomi-gemini.dts | 5 +++-- .../boot/dts/qcom/msm8996-xiaomi-scorpio.dts | 5 +++-- include/dt-bindings/sound/qcom,wcd9335.h | 15 +++++++++++++++ sound/soc/codecs/wcd9335.c | 13 ++----------- 6 files changed, 27 insertions(+), 17 deletions(-) create mode 100644 include/dt-bindings/sound/qcom,wcd9335.h -- 2.36.1

2 6

[PATCH v7 00/14] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 22 Jun '22

22 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. changes since v6: - Fix warning by kernel test robot <lkp(a)intel.com> changes since v5: - Fix warning by kernel test robot <lkp(a)intel.com> changes since v4: - Fully remove tlv remnants from control add apis - Remove unnecessary debug - Rename variable to be more generic - Remove redundent length check from read/write control apis - Use SNDRV_CTL_ELEM_IFACE_CARD for firmware load controls - Make kcontrol add/remove synchronous - Load firmware asynchronous when loading via control - Used cached controls when reloading firmware; only delete controls when removing the driver itself - Improve kcontrol remove - Fix control write + notify - Cleanup of unnecessary code - Fix race condition when loading firmware before playback - Ensure errors are properly propogated - Fix include for Module parameters Stefan Binding (13): ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add module parameter to control firmware load Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 4 + sound/pci/hda/Kconfig | 8 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 902 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 39 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 1 + sound/pci/hda/hda_component.h | 4 + sound/pci/hda/hda_cs_dsp_ctl.c | 254 +++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 39 ++ sound/pci/hda/patch_realtek.c | 27 +- 12 files changed, 1277 insertions(+), 5 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.34.1

3 19

[PATCH v4 0/3] ASoC: rockchip: i2s: switch BCLK to GPIO
by Judy Hsiao 22 Jun '22

22 Jun '22

The patches series is to fix the unexpected large DC output voltage of Max98357a that burns the speakers on the rockchip platform when BCLK and SD_MODE are ON but LRCLK is OFF. Changes Since V4: -- Fix indentation in the driver. (Align parameters with the parenthesis placement.) -- Fix incorrect return type of rockchip_snd_rxctrl. Changes Since V3: -- Fix indentation in the documentation. -- Put pinctrl-1 right after pinctrl-0 in dtsi. -- Fix indentation in the driver. -- Remove unnecessary dev_dbg() in the driver. Changes Since V2: -- Add documents of i2s pinctrl-names. -- Fix dtsi syntax error. -- Include the dtsi change and the driver change in the same series. -- Ensure that driver gets both bclk_on and bclk_off states before using them. Judy Hsiao (3): ASoC: rockchip: i2s: switch BCLK to GPIO arm64: dts: rk3399: i2s: switch BCLK to GPIO ASoC: dt-bindings: rockchip: Document pinctrl-names for i2s .../bindings/sound/rockchip-i2s.yaml | 7 + .../boot/dts/rockchip/rk3399-gru-scarlet.dtsi | 10 + arch/arm64/boot/dts/rockchip/rk3399.dtsi | 25 ++- sound/soc/rockchip/rockchip_i2s.c | 171 +++++++++++++----- 4 files changed, 165 insertions(+), 48 deletions(-) -- 2.36.1.476.g0c4daa206d-goog

3 8

[PATCH v6 00/14] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 22 Jun '22

22 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. changes since v5: - Fix warning by kernel test robot <lkp(a)intel.com> changes since v4: - Fully remove tlv remnants from control add apis - Remove unnecessary debug - Rename variable to be more generic - Remove redundent length check from read/write control apis - Use SNDRV_CTL_ELEM_IFACE_CARD for firmware load controls - Make kcontrol add/remove synchronous - Load firmware asynchronous when loading via control - Used cached controls when reloading firmware; only delete controls when removing the driver itself - Improve kcontrol remove - Fix control write + notify - Cleanup of unnecessary code - Fix race condition when loading firmware before playback - Ensure errors are properly propogated - Fix include for Module parameters Stefan Binding (13): ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add module parameter to control firmware load Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 4 + sound/pci/hda/Kconfig | 8 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 902 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 39 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 1 + sound/pci/hda/hda_component.h | 4 + sound/pci/hda/hda_cs_dsp_ctl.c | 254 +++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 39 ++ sound/pci/hda/patch_realtek.c | 27 +- 12 files changed, 1277 insertions(+), 5 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.34.1

2 15

No sound on AMD audio device
by GitHub issues - edited 21 Jun '22

21 Jun '22

alsa-project/alsa-lib issue #241 was edited from chvmq: I have a laptop huawei 14 amd 2021 I don't know where write about it. maybe there I tried ubuntu | arch-linux and no sound at all ![image](https://user-images.githubusercontent.com/60788249/174885007-8eb96062-3753-47b6-9880-5a45b57071ac.png) ![image](https://user-images.githubusercontent.com/60788249/174885044-42b8c5c2-6c64-46df-ade6-6f5c8388a972.png) ![image](https://user-images.githubusercontent.com/60788249/174885077-b13fc2a4-18cb-4f20-84a7-3c611ebbecc3.png) Issue URL : https://github.com/alsa-project/alsa-lib/issues/241 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

No sound on AMD audio device
by GitHub issues - edited 21 Jun '22

21 Jun '22

alsa-project/alsa-lib issue #241 was edited from chvmq: I have a laptop huawei 14 amd 2021 I don't know where write about it. maybe there I tried ubuntu | arch-linux and no sound at all ![3dee5e90-3eaf-4c30-b11b-538282950d39](https://user-images.github ![e0a07559-3156-4cb9-8ca7-6c8087cb375e](https://user-images.githubusercontent.com/60788249/174884837-e59af122-bccf-4a82-b7a7-8f3666139c3d.png) usercontent.com/60788249/174884832-d3b8599b-8457-43df-849d-3a6fb7acda68.png) ![09f629fe-6c69-41ff-a708-b1ee7e482205](https://user-images.githubusercontent.com/60788249/174884845-12525b62-6a45-4efd-9988-3f91120495d5.png) Issue URL : https://github.com/alsa-project/alsa-lib/issues/241 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/4] ASoC: wm_adsp: Fix event for preloader
by Charles Keepax 21 Jun '22

21 Jun '22

The preloader controls on ADSP should return a value of 1 if the preloader value was changed, update to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6d7fd88243aa8..a7784ac15dde6 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -997,7 +997,7 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, snd_soc_dapm_sync(dapm); } - return 0; + return 1; } EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put); -- 2.30.2

3 5

[PATCH] ASoC: rt711-sdca: fix kernel NULL pointer dereference when IO error
by shumingf＠realtek.com 21 Jun '22

21 Jun '22

From: Shuming Fan <shumingf(a)realtek.com> The initial settings will be written before the codec probe function. But, the rt711->component doesn't be assigned yet. If IO error happened during initial settings operations, it will cause the kernel panic. This patch changed component->dev to slave->dev to fix this issue. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 9d226b1cb7e9..925268121901 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -34,7 +34,7 @@ static int rt711_sdca_index_write(struct rt711_sdca_priv *rt711, ret = regmap_write(regmap, addr, value); if (ret < 0) - dev_err(rt711->component->dev, + dev_err(&rt711->slave->dev, "Failed to set private value: %06x <= %04x ret=%d\n", addr, value, ret); @@ -50,7 +50,7 @@ static int rt711_sdca_index_read(struct rt711_sdca_priv *rt711, ret = regmap_read(regmap, addr, value); if (ret < 0) - dev_err(rt711->component->dev, + dev_err(&rt711->slave->dev, "Failed to get private value: %06x => %04x ret=%d\n", addr, *value, ret); -- 2.36.1

2 1

[PATCH 1/4] ASoC: fsl-asoc-card: add WM8904 support
by Alifer Moraes 21 Jun '22

21 Jun '22

From: Pierluigi Passaro <pierluigi.p(a)variscite.com> The codec WM8904 can use internal FLL as PLL source. Whenever the PLL source is not an external MCLK, this source must be setup during hw_params callback otherwise the BCLK could be wrongly evaluated. The SND_SOC_BIAS_PREPARE event is raised after the hw_params callback, so there is no need to set again PLL and SYSCLK and actually there's no need at all the set_bias_level function. Also, when esai is used, a dedicated snd_soc_dai_set_tdm_slot call is required. Signed-off-by: Pierluigi Passaro <pierluigi.p(a)variscite.com> Signed-off by: Alifer Moraes <alifer.m(a)variscite.com> --- sound/soc/fsl/fsl-asoc-card.c | 47 +++++++++++++++++++++++++++++++++++ 1 file changed, 47 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5ee945505281..817dbc1ec635 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -23,6 +23,7 @@ #include "imx-audmux.h" #include "../codecs/sgtl5000.h" +#include "../codecs/wm8904.h" #include "../codecs/wm8962.h" #include "../codecs/wm8960.h" #include "../codecs/wm8994.h" @@ -257,6 +258,38 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) } } + if (of_device_is_compatible(dev->of_node, "fsl,imx-audio-wm8904")) { + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + unsigned int pll_out; + + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0, 0, 2, + params_physical_width(params)); + if (ret) { + dev_err(dev, "failed to set TDM slot for cpu dai\n"); + return ret; + } + + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, + codec_priv->pll_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + } + return 0; } @@ -651,6 +684,19 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8904")) { + codec_dai_name = "wm8904-hifi"; + priv->card.set_bias_level = NULL; + priv->codec_priv.mclk_id = WM8904_CLK_FLL; + priv->codec_priv.fll_id = WM8904_CLK_FLL; + priv->codec_priv.pll_id = WM8904_FLL_MCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + if (strstr(cpu_np->name, "esai")) { + priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; + priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; + } } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { codec_dai_name = "ac97-hifi"; priv->dai_fmt = SND_SOC_DAIFMT_AC97; @@ -900,6 +946,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-tlv320aic32x4", }, { .compatible = "fsl,imx-audio-tlv320aic31xx", }, { .compatible = "fsl,imx-audio-sgtl5000", }, + { .compatible = "fsl,imx-audio-wm8904", }, { .compatible = "fsl,imx-audio-wm8962", }, { .compatible = "fsl,imx-audio-wm8960", }, { .compatible = "fsl,imx-audio-mqs", }, -- 2.25.1

4 10

Intel HDMI probe regression on IVB (and older?)
by Takashi Iwai 21 Jun '22

21 Jun '22

Hi, we've got a regression report about Intel HDMI. It seems that the recent change to skip the component binding (commit c9db8a30d9f0) throws away the devices incorrectly on IvyBridge. I guess the similar issue could happen on older chips. The bug report is found at https://bugzilla.opensuse.org/show_bug.cgi?id=1200611 Judging from the logs there, on IvyBridge, the GPU is 00:02.0 0380: 8086:0162 (rev 09) DeviceName: Onboard IGD Subsystem: 1043:84ca Flags: bus master, fast devsel, latency 0, IRQ 27 Memory at f7800000 (64-bit, non-prefetchable) [size=4M] Memory at d0000000 (64-bit, prefetchable) [size=256M] I/O ports at f000 [size=64] Capabilities: <access denied> Kernel driver in use: i915 Kernel modules: i915 while HD-audio is 00:1b.0 0403: 8086:1e20 (rev 04) Subsystem: 1043:84a8 Flags: bus master, fast devsel, latency 0, IRQ 31 Memory at f7d10000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel Unlike Haswell, IVB has no dedicated HDMI audio controller, and it's connected on HD-audio bus of 00:1b.0. Kai, could you check this issue? thanks, Takashi

2 7

[PATCH] ALSA: hda: Fix discovery of i915 graphics PCI device
by Takashi Iwai 21 Jun '22

21 Jun '22

It's been reported that the recent fix for skipping the component-binding with D-GPU caused a regression on some systems; it resulted in the completely missing component binding with i915 GPU. The problem was the use of pci_get_class() function. It matches with the full PCI class bits, while we want to match only partially the PCI base class bits. So, when a system has an i915 graphics device with the PCI class 0380, it won't hit because we're looking for only the PCI class 0300. This patch fixes i915_gfx_present() to look up each PCI device and match with PCI base class explicitly instead of pci_get_class(). Fixes: c9db8a30d9f0 ("ALSA: hda/i915 - skip acomp init if no matching display") Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Tested-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: <stable(a)vger.kernel.org> Link: https://bugzilla.opensuse.org/show_bug.cgi?id=1200611 Link: https://lore.kernel.org/r/87bkunztec.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_i915.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 3f35972e1cf7..161a9711cd63 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -119,21 +119,18 @@ static int i915_component_master_match(struct device *dev, int subcomponent, /* check whether Intel graphics is present and reachable */ static int i915_gfx_present(struct pci_dev *hdac_pci) { - unsigned int class = PCI_BASE_CLASS_DISPLAY << 16; struct pci_dev *display_dev = NULL; - bool match = false; - do { - display_dev = pci_get_class(class, display_dev); - - if (display_dev && display_dev->vendor == PCI_VENDOR_ID_INTEL && + for_each_pci_dev(display_dev) { + if (display_dev->vendor == PCI_VENDOR_ID_INTEL && + (display_dev->class >> 16) == PCI_BASE_CLASS_DISPLAY && connectivity_check(display_dev, hdac_pci)) { pci_dev_put(display_dev); - match = true; + return true; } - } while (!match && display_dev); + } - return match; + return false; } /** -- 2.35.3

1 0

[PATCH 0/3] ASoC: sun50i-codec-analog: Internal bias support
by Samuel Holland 21 Jun '22

21 Jun '22

This series adds support for enabling the codec's internal microphone bias, which is needed on at least some versions of the PinePhone. Arnaud Ferraris (2): ASoC: dt-bindings: sun50i-codec: Add binding for internal bias ASoC: sun50i-codec-analog: Add support for internal bias Samuel Holland (1): arm64: dts: allwinner: pinephone: Enable internal HMIC bias .../allwinner,sun50i-a64-codec-analog.yaml | 5 ++++ .../allwinner/sun50i-a64-pinephone-1.0.dts | 4 +++ .../allwinner/sun50i-a64-pinephone-1.1.dts | 4 +++ sound/soc/sunxi/sun50i-codec-analog.c | 29 +++++++++++++++++++ 4 files changed, 42 insertions(+) -- 2.35.1

2 5

[PATCH 00/10] timer: rename tstamp methods with the word real_time
by GitHub pull_request - opened 21 Jun '22

21 Jun '22

alsa-project/alsa-gobject pull request #78 was opened from takaswie: In ALSATimer library some methods are named with the word `tstamp`. On the other hand, in ALSASeq library the similar methods are named with the word `real_time`. For consistency, this patchset renames methods in ALSATimer library. As a result, the word `tstamp` is not used at all. ``` Takashi Sakamoto (10): utils: successfully parse the string teminated by '\n' timer: move ALSATimer.TickEvent files for future rename timer: move ALSATimer.TstampEvent files for future rename timer: tick-time-event: rename boxed structure timer: real-time-event: rename boxed structure timer: enums: rename to ALSATimer.EventType.TICK_TIME and REAL_TIME timer: enums: rename to ALSATimer.RealTimeEventType timer: instance-status: rename method with real time timer: query: rename to ALSATimer.get_real_time_clock_id() seq: event: fix description about methods relevant to time stamp samples/timer | 26 +++---- src/seq/event.c | 16 ++-- src/timer/alsatimer-enum-types.h | 74 +++++++++---------- src/timer/alsatimer.h | 4 +- src/timer/alsatimer.map | 20 ++--- src/timer/instance-params.c | 20 ++--- src/timer/instance-params.h | 4 +- src/timer/instance-status.c | 22 +++--- src/timer/instance-status.h | 3 +- src/timer/meson.build | 8 +- src/timer/query.c | 16 ++-- src/timer/query.h | 2 +- src/timer/real-time-event.c | 65 ++++++++++++++++ src/timer/real-time-event.h | 25 +++++++ src/timer/tick-event.c | 51 ------------- src/timer/tick-event.h | 23 ------ src/timer/tick-time-event.c | 50 +++++++++++++ src/timer/tick-time-event.h | 22 ++++++ src/timer/tstamp-event.c | 64 ---------------- src/timer/tstamp-event.h | 24 ------ src/timer/user-instance.c | 56 +++++++------- src/timer/user-instance.h | 17 +++-- src/utils/string.c | 2 +- tests/alsatimer-enums | 8 +- tests/alsatimer-functions | 2 +- tests/alsatimer-instance-status | 2 +- ...tstamp-event => alsatimer-real-time-event} | 4 +- ...r-tick-event => alsatimer-tick-time-event} | 4 +- tests/alsatimer-user-instance | 8 +- tests/meson.build | 4 +- 30 files changed, 324 insertions(+), 322 deletions(-) create mode 100644 src/timer/real-time-event.c create mode 100644 src/timer/real-time-event.h delete mode 100644 src/timer/tick-event.c delete mode 100644 src/timer/tick-event.h create mode 100644 src/timer/tick-time-event.c create mode 100644 src/timer/tick-time-event.h delete mode 100644 src/timer/tstamp-event.c delete mode 100644 src/timer/tstamp-event.h rename tests/{alsatimer-tstamp-event => alsatimer-real-time-event} (84%) rename tests/{alsatimer-tick-event => alsatimer-tick-time-event} (83%) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/78 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/78.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

GoXLR Broken in 1.2.7.1
by GitHub issues - opened 21 Jun '22

21 Jun '22

alsa-project/alsa-ucm-conf issue #175 was opened from FrostyCoolSlug: The alsa profiles no longer set correctly for the GoXLR or GoXLR Mini since 5dcd69. >From what I can tell with some quick tests, if the DirectCardName isn't set, the profile doesn't seem to apply anymore. Removing the direct config, or adding a DirectCardName seems to solve the issue (although obviously, neither are ideal). Thanks. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/175 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v5 00/14] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 21 Jun '22

21 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. changes since v4: - Fully remove tlv remnants from control add apis - Remove unnecessary debug - Rename variable to be more generic - Remove redundent length check from read/write control apis - Use SNDRV_CTL_ELEM_IFACE_CARD for firmware load controls - Make kcontrol add/remove synchronous - Load firmware asynchronous when loading via control - Used cached controls when reloading firmware; only delete controls when removing the driver itself - Improve kcontrol remove - Fix control write + notify - Cleanup of unnecessary code - Fix race condition when loading firmware before playback - Ensure errors are properly propogated - Fix include for Module parameters Stefan Binding (13): ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add module parameter to control firmware load Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 4 + sound/pci/hda/Kconfig | 8 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 902 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 39 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 1 + sound/pci/hda/hda_component.h | 4 + sound/pci/hda/hda_cs_dsp_ctl.c | 254 +++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 39 ++ sound/pci/hda/patch_realtek.c | 27 +- 12 files changed, 1277 insertions(+), 5 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.34.1

2 15

Enabling fsl_sai MCLK before codec probe
by Michael Welling 20 Jun '22

20 Jun '22

So I have a tricky race condition that comes up with hardware that combines imx8 with a pcm1681 codec. The pcm1681 requires that MCLK be enabled for 65536 cycles before accessing the I2C management port. The imx8 sai driver on the other hand does not start the built in MCLK until a confluence of events that is not triggered until just before streaming starts which is too late. Is there a clean way to start the imx8 sai MCLK before the codec probe occurs? Or have it started and delayed during the codec probe before it tries to access i2c? We have a working solution but it is a bit overreaching and adds a callback function to the snd_soc_dai_ops struct to start the clock, and adds the code to the fsl_sai.c to toggle the appropriate bits in the SoC sai driver to make a clock appear and registers to the new function callback. Then the simple-card was amended to add the appropriate enabling through the new callback such that the clock is on long enough before the codec is probed. There was some suggestion of caching the i2c registers until the clock is enabled but this would be difficult given the clock is not started until streaming starts and we don't want to stream until the codec is initialized. I am looking for a solution that is minimal and robust. Any pointers would be greatly appreciated.

1 1

Re: [PATCH 4/4] ASoC: wm8904: add DMIC support
by Pierluigi Passaro 20 Jun '22

20 Jun '22

> > Just for my understanding, are you suggesting to set a device tree > > property to force a fixed behavior in the driver ? > Yes. Why should we use a fixed behavior ? > > WM8904 allows using both a DMIC and LINEIN, switching between one or > > the other and this is how we currently use it. > > Why the user should not be allowed to switch between DMIC and LINEIN ? > The device shares pins between the line inputs and the DMIC inputs so at > least some of the configuration is going to be determinted at system > design time, that will fix the usable values of at least one of the > controls which ought to be reflected in the runtime behaviour. In our design we use: - pin 1: DMIC_CLK - pin 24: LINEIN2R - pin 26: LINEIN2L - pin 27: DMIC_DATA we have no pins shared among DMIC and LINEIN. We have several customer switching between DMIC and LINEIN at runtime. > Please fix your mail client to word wrap within paragraphs at something > substantially less than 80 columns. Doing this makes your messages much > easier to read and reply to. I beg your pardon: it should be fixed now.

2 1

Re: [PATCH 4/4] ASoC: wm8904: add DMIC support
by Pierluigi Passaro 20 Jun '22

20 Jun '22

> > > Via firmware description. > > Can you please provide any reference approach in the kernel code ? > git grep of_ > git grep fwnode_ > and I don't immediately remember what the prefix is for ACPI functions. Just for my understanding, are you suggesting to set a device tree property to force a fixed behavior in the driver ? WM8904 allows using both a DMIC and LINEIN, switching between one or the other and this is how we currently use it. Why the user should not be allowed to switch between DMIC and LINEIN ?

2 1

Re: [PATCH 3/4] ASoC: wm8904: extend device tree support
by Pierluigi Passaro 20 Jun '22

20 Jun '22

Hi All, > > + - num-drc-cfgs: Number of available DRC modes from drc-cfg-regs property > > + > > + - drc-cfg-regs: Default registers value for R40/41/42/43 (DRC) > > + The list must be (4 x num-drc-cfgs) entries long. > > + If absent or incomplete, DRC is disabled. > > What is the purpose of having num-drc-cfgs? We can tell how large > drc-cfg-regs is so it just seems redundant. Can you please point me to any reference implementation doing this ? > > + - num-retune-mobile-cfgs: Number of retune modes available from > > + retune-mobile-cfg-regs property > > Same here. Thanks Best Regards Pier

2 1

Re: [PATCH 4/4] ASoC: wm8904: add DMIC support
by Pierluigi Passaro 20 Jun '22

20 Jun '22

> > > > +static const char *cin_text[] = { > > > > + "ADC", "DMIC" > > > > +}; > > > > +static SOC_ENUM_SINGLE_DECL(cin_enum, > > > > + WM8904_DIGITAL_MICROPHONE_0, 12, cin_text); > > > Why would this be runtime selectable? I'd expect the decision to use > > > an analogue or digital microphone to be made in the hardware design. > > I agree that dedicated HW is required, but currently SW side there's no support at all. > > This patch is aiming to provide a way to enable DMIC on boards using it. > > Is this supposed to be managed in a different way ? > Via firmware description. Can you please provide any reference approach in the kernel code ?

2 1

Re: [PATCH 4/4] ASoC: wm8904: add DMIC support
by Pierluigi Passaro 20 Jun '22

20 Jun '22

Hi All, > > +static const char *cin_text[] = { > > + "ADC", "DMIC" > > +}; > > + > > +static SOC_ENUM_SINGLE_DECL(cin_enum, > > + WM8904_DIGITAL_MICROPHONE_0, 12, cin_text); > > Why would this be runtime selectable? I'd expect the decision to use > an analogue or digital microphone to be made in the hardware design. I agree that dedicated HW is required, but currently SW side there's no support at all. This patch is aiming to provide a way to enable DMIC on boards using it. Is this supposed to be managed in a different way ? Thanks Best Regards Pier

2 1

[PATCH] MAINTAINERS: update ASoC Qualcomm maintainer email-id
by Srinivas Kandagatla 20 Jun '22

20 Jun '22

Update Banajit's email address from codeaurora.org to quicinc.com, as codeaurora.org is not in use anymore. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 440f3d7c93b9..171fa3160696 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -16247,7 +16247,7 @@ F: drivers/crypto/qat/ QCOM AUDIO (ASoC) DRIVERS M: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> -M: Banajit Goswami <bgoswami(a)codeaurora.org> +M: Banajit Goswami <bgoswami(a)quicinc.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/codecs/lpass-va-macro.c -- 2.25.1

2 1

No sound on main speaker : Zenbook UX3402/ALC294/Fedora 36
by GitHub issues - opened 20 Jun '22

20 Jun '22

alsa-project/alsa-ucm-conf issue #174 was opened from hereblur: My laptop has no sound on main speakers. 3.5mm jack and bluetooth is working fine. Asus Zenbook UX3402 ALC294 Fedora 36 Here is my `alsaucm -c hw:0 dump text` and `alsa-info.sh` output [https://paste.debian.net/1244631/](https://paste.debian.net/1244631/) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/174 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: fsl_asrc_dma: Use dmaengine_terminate_async()
by Sascha Hauer 20 Jun '22

20 Jun '22

dmaengine_terminate_all() is deprecated and should no longer be used. Use dmaengine_terminate_async() instead. This involves no functional change since both functions do the same. After dmaengine_terminate_async() dmaengine_synchronize() must be called to make sure the channel has really stopped before the underlying memory is freed. This is done implicitly by dma_release_channel() called from the .hw_free hook. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/fsl/fsl_asrc_dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index fec56dbb64e60..6c4d161ddad85 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -114,8 +114,8 @@ static int fsl_asrc_dma_trigger(struct snd_soc_component *component, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dmaengine_terminate_all(pair->dma_chan[OUT]); - dmaengine_terminate_all(pair->dma_chan[IN]); + dmaengine_terminate_async(pair->dma_chan[OUT]); + dmaengine_terminate_async(pair->dma_chan[IN]); break; default: return -EINVAL; -- 2.30.2

2 1

[PATCH 1/2] ALSA: hda/conexant: Fix missing beep setup
by Takashi Iwai 20 Jun '22

20 Jun '22

Currently the Conexant codec driver sets up the beep NID after calling snd_hda_gen_parse_auto_config(). It turned out that this results in the insufficient setup for the beep control, as the generic parser handles the fake path in snd_hda_gen_parse_auto_config() only if the beep_nid is set up beforehand. For dealing with the beep widget properly, call cx_auto_parse_beep() before snd_hda_gen_parse_auto_config() call. Fixes: 51e19ca5f755 ("ALSA: hda/conexant - Clean up beep code") Cc: <stable(a)vger.kernel.org> Link: https://bugzilla.kernel.org/show_bug.cgi?id=216152 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1248d1a51cf0..3e541a4c0423 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1079,11 +1079,11 @@ static int patch_conexant_auto(struct hda_codec *codec) if (err < 0) goto error; - err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); + err = cx_auto_parse_beep(codec); if (err < 0) goto error; - err = cx_auto_parse_beep(codec); + err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); if (err < 0) goto error; -- 2.35.3

1 1

[PATCH 00/23] add annotation to object properties
by GitHub pull_request - opened 20 Jun '22

20 Jun '22

alsa-project/alsa-gobject pull request #77 was opened from takaswie: The gi-docgen tool requires explicit annotation for object properties to generate pages. This patchset adds missing properties for the purpose. ``` Takashi Sakamoto (23): ctl: card: add annotation to object properties ctl: card-info: add annotation to object properties card: elem-value: add annotation to object properties rawmidi: stream-pair: add annotation to object properties rawmidi: substream-info: add annotation to object properties rawmidi: substream-params: add annotation to object properties rawmidi: substream-status: add annotation to object properties hwdep: device-info: add annotation to object properties timer: device-info: add annotation to object properties timer: device-params: add annotation to object properties timer: device-status: add annotation to object properties timer: instance-info: add annotation to object properties timer: instance-params: add annotation to object properties timer: instance-status: add annotation to object properties seq: fix description of object properties in which enumerations are available seq: client-info: add annotation to object properties seq: client-pool: add annotation to object properties seq: port-info: fix tag indentation seq: queue-info: add annotation to object properties seq: queue-status: add annotation to object properties seq: queue-tempo: add annotation to object properties seq: system-info: add annotation to object properties seq: user-client: add annotation to object properties src/ctl/card-info.c | 37 ++++++++++++++++++- src/ctl/card.c | 12 ++++++- src/ctl/elem-value.c | 5 +++ src/hwdep/device-info.c | 34 +++++++++++++++--- src/rawmidi/stream-pair.c | 7 +++- src/rawmidi/substream-info.c | 66 +++++++++++++++++++++++++++++----- src/rawmidi/substream-params.c | 22 +++++++++--- src/rawmidi/substream-status.c | 10 ++++++ src/seq/client-info.c | 59 ++++++++++++++++++++++++++---- src/seq/client-pool.c | 52 ++++++++++++++++++++------- src/seq/port-info.c | 2 +- src/seq/query.c | 4 +-- src/seq/queue-info.c | 28 ++++++++++++--- src/seq/queue-status.c | 19 ++++++++-- src/seq/queue-tempo.c | 19 ++++++++-- src/seq/queue-timer-common.c | 2 +- src/seq/remove-filter.c | 19 +++++----- src/seq/system-info.c | 30 ++++++++++++++++ src/seq/user-client.c | 19 ++++++---- src/timer/device-info.c | 46 ++++++++++++++++++++++-- src/timer/device-params.c | 10 ++++++ src/timer/device-status.c | 15 ++++++++ src/timer/instance-info.c | 27 +++++++++++++- src/timer/instance-params.c | 18 ++++++++-- src/timer/instance-status.c | 20 +++++++++++ 25 files changed, 506 insertions(+), 76 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/77 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/77.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH v3 0/4] Add MT8186 ADSP dt-binding
by Tinghan Shen 20 Jun '22

20 Jun '22

v2 -> v3: 1. Change mbox-names to rx/tx for both mt8186 and mt8195. 2. Update description of mbox-names 3. Use static string array instead of kasprintf 4. Align clock names in dsp driver with dt-bindings v1 -> v2: 1. Change mbox-names from mbox0/mbox1 to rep/req for both mt8186 and mt8195. 2. rename clock-names and remove unused headers Tinghan Shen (4): dt-bindings: dsp: mediatek: Use meaningful names for mbox firmware: mediatek: Use meaningful names for mbox dt-bindings: dsp: mediatek: Add mt8186 dsp document ASoC: SOF: mediatek: Align mt8186 clock names with dt-bindings .../bindings/dsp/mediatek,mt8186-dsp.yaml | 91 +++++++++++++++++++ .../bindings/dsp/mediatek,mt8195-dsp.yaml | 10 +- drivers/firmware/mtk-adsp-ipc.c | 36 +++----- sound/soc/sof/mediatek/mt8186/mt8186-clk.c | 4 +- 4 files changed, 110 insertions(+), 31 deletions(-) create mode 100644 Documentation/devicetree/bindings/dsp/mediatek,mt8186-dsp.yaml -- 2.18.0

2 9

[PATCH v3 00/17] ASoC: Intel: haswell and broadwell boards update
by Cezary Rojewski 20 Jun '22

20 Jun '22

A number of patches improving overall quality and readability of haswell.c and broadwell.c source files found in sound/soc/intel/boards. Both files are first renamed and only then actual changes are being incrementally added. The respective names are: hsw_rt5640 and bdw_rt286 to match the pattern found in more recent boards. Most patches bring no functional change - the more impactful patches at are placed the end: Refactor of suspend/resume flow for the bdw_rt286 board by dropping dev->remove() in favour of card->remove() and adjust jack handling to reduce code size slightly by implementing card_set_jack(). The last patch is removing of FE DAI ops. Given the existence of platform FE DAI capabilities (either static declaration or through topology file), this code is redundant. Changes in v3: - Patch 16/17 refactoring suspend/resume has been renamed to "Refactor jack handling". Dropped the usage of card->remove() in favor of link->exit() in that very patch Changes in v2: - fixed wording error in patch 02/17 so it correctly mentions 'haswell_rt5640', not 'broadwell_rt286' - decided not to add kernel module names changes to this patchset so the review is not complicated unnecessarily. Will send them separately instead Cezary Rojewski (17): ASoC: Intel: Rename haswell source file to hsw_rt5640 ASoC: Intel: hsw_rt5640: Reword prefixes of all driver members ASoC: Intel: hsw_rt5640: Reword driver name ASoC: Intel: hsw_rt5640: Update code indentation ASoC: Intel: hsw_rt5640: Update file comments ASoC: Intel: hsw_rt5640: Improve probe() function quality ASoC: Intel: hsw_rt5640: Improve hw_params() debug-ability ASoC: Intel: Rename broadwell source file to bdw_rt286 ASoC: Intel: bdw_rt286: Reword prefixes of all driver members ASoC: Intel: bdw_rt286: Reword driver name ASoC: Intel: bdw_rt286: Update code indentation ASoC: Intel: bdw_rt286: Update file comments ASoC: Intel: bdw_rt286: Improve probe() function quality ASoC: Intel: bdw_rt286: Improve hw_params() debug-ability ASoC: Intel: bdw_rt286: Improve codec_init() quality ASoC: Intel: bdw_rt286: Refactor jack handling ASoC: Intel: bdw_rt286: Remove FE DAI ops sound/soc/intel/boards/Kconfig | 4 +- sound/soc/intel/boards/Makefile | 4 +- sound/soc/intel/boards/bdw_rt286.c | 256 +++++++++++++ sound/soc/intel/boards/broadwell.c | 336 ------------------ sound/soc/intel/boards/haswell.c | 202 ----------- sound/soc/intel/boards/hsw_rt5640.c | 176 +++++++++ .../common/soc-acpi-intel-hsw-bdw-match.c | 6 +- 7 files changed, 439 insertions(+), 545 deletions(-) create mode 100644 sound/soc/intel/boards/bdw_rt286.c delete mode 100644 sound/soc/intel/boards/broadwell.c delete mode 100644 sound/soc/intel/boards/haswell.c create mode 100644 sound/soc/intel/boards/hsw_rt5640.c -- 2.25.1

1 17

[PATCH] ALSA: memalloc: Drop x86-specific hack for WC allocations
by Takashi Iwai 20 Jun '22

20 Jun '22

The recent report for a crash on Haswell machines implied that the x86-specific (rather hackish) implementation for write-cache memory buffer allocation in ALSA core is buggy with the recent kernel in some corner cases. This patch drops the x86-specific implementation and uses the standard dma_alloc_wc() & co generically for avoiding the bug and also for simplification. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216112 Cc: <stable(a)vger.kernel.org> # v5.18+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 23 +---------------------- 1 file changed, 1 insertion(+), 22 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 15dc7160ba34..8cfdaee77905 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -431,33 +431,17 @@ static const struct snd_malloc_ops snd_dma_iram_ops = { */ static void *snd_dma_dev_alloc(struct snd_dma_buffer *dmab, size_t size) { - void *p; - - p = dma_alloc_coherent(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); -#ifdef CONFIG_X86 - if (p && dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC) - set_memory_wc((unsigned long)p, PAGE_ALIGN(size) >> PAGE_SHIFT); -#endif - return p; + return dma_alloc_coherent(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); } static void snd_dma_dev_free(struct snd_dma_buffer *dmab) { -#ifdef CONFIG_X86 - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC) - set_memory_wb((unsigned long)dmab->area, - PAGE_ALIGN(dmab->bytes) >> PAGE_SHIFT); -#endif dma_free_coherent(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); } static int snd_dma_dev_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { -#ifdef CONFIG_X86 - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC) - area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); -#endif return dma_mmap_coherent(dmab->dev.dev, area, dmab->area, dmab->addr, dmab->bytes); } @@ -471,10 +455,6 @@ static const struct snd_malloc_ops snd_dma_dev_ops = { /* * Write-combined pages */ -#ifdef CONFIG_X86 -/* On x86, share the same ops as the standard dev ops */ -#define snd_dma_wc_ops snd_dma_dev_ops -#else /* CONFIG_X86 */ static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) { return dma_alloc_wc(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); @@ -497,7 +477,6 @@ static const struct snd_malloc_ops snd_dma_wc_ops = { .free = snd_dma_wc_free, .mmap = snd_dma_wc_mmap, }; -#endif /* CONFIG_X86 */ #ifdef CONFIG_SND_DMA_SGBUF static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size); -- 2.35.3

1 0

Enabling tstamp in proc status file externally
by Pavel Hofman 20 Jun '22

20 Jun '22

Hi, Please is there any way to enable the tstamp in stream status without modifying the client calling the alsa-lib API? I wanted to measure samplerate ratio between soundcards using data in their status proc files (comparing advancement of tstamp vs. hw_ptr). The method seems to work quite good, but some clients enable the stream status tstamp (e.g. pulseaudio) and some don't (e.g. sox, aplay), resulting in zeros in the status proc file. Thanks a lot for any help or hint. Best regards, Pavel.

2 3

alsa-utils 1.2.7 fails to build if rawmidi support is disabled in alsa-lib
by GitHub issues - opened 19 Jun '22

19 Jun '22

alsa-project/alsa-utils issue #154 was opened from Saur2000: If alsa-lib is configured using `--disable-rawmidi`, then alsa-utils fails to build: ``` alsactl/info.c:83:44: error: unknown type name 'snd_rawmidi_stream_t'; did you mean 'snd_pcm_stream_t'? 83 | static const char *snd_rawmidi_stream_name(snd_rawmidi_stream_t stream) | ^~~~~~~~~~~~~~~~~~~~ | snd_pcm_stream_t alsactl/info.c:92:48: error: unknown type name 'snd_rawmidi_stream_t'; did you mean 'snd_pcm_stream_t'? 92 | static int rawmidi_device_list(snd_ctl_t *ctl, snd_rawmidi_stream_t stream, bool *first) | ^~~~~~~~~~~~~~~~~~~~ | snd_pcm_stream_t ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/154 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Alsa-to-Jack bridge broken in alsa* 1.2.7.x
by GitHub issues - opened 19 Jun '22

19 Jun '22

alsa-project/alsa-lib issue #243 was opened from hlekin: I am using a setup according to https://alsa.opensrc.org/Jack_and_Loopback_device_as_Alsa-to-Jack_bridge Since the upgrade of alsa-lib, alsa-ucm-conf, and alsa-utils to version 1.2.7, my alsa-to-jack-bridge setup is broken: $ jack_connect system:capture_1 ploop:playback_1 ERROR ploop:playback_1 not a valid port A downgrade to versions 1.2.6 resolves the issue. Versions 1.2.7.1 result in the same error. The relevant output of qjackctl 0.9.7-2 > Messages is: Cannot write socket fd = 25 err = Broken pipe CheckRes error Could not write notification ClientNotify fails name = cloop notification = 0 val1 = 0 val2 = 0 NotifyAddClient old_client fails name = ploop Cannot write socket fd = 28 err = Broken pipe CheckRes error Could not write notification ClientNotify fails name = ploop notification = 1 val1 = 0 val2 = 0 ERROR ploop:playback_1 not a valid port ERROR ploop:playback_2 not a valid port ERROR cloop:capture_1 not a valid port ERROR cloop:capture_2 not a valid port Please find attached alsa-info.sh output for alsa-versions 1.2.7.1 and 1.2.6 [alsa-info.sh-output-220619.zip](https://github.com/alsa-project/alsa-lib/fi… . I am aware, that this is a very poor bug report. Please ask for any information you need! By the way, many thanks for all your programs and the effort you put in to them. Issue URL : https://github.com/alsa-project/alsa-lib/issues/243 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

pcm: Return unsigned int if error occurs by `read` or `write` system call
by GitHub pull_request - opened 19 Jun '22

19 Jun '22

alsa-project/alsa-lib pull request #242 was opened from Korilakkuma: `read` or `write` system call returns `-1` if error occurs. Request URL : https://github.com/alsa-project/alsa-lib/pull/242 Patch URL : https://github.com/alsa-project/alsa-lib/pull/242.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] aplay: Support setting timestamp
by Pavel Hofman 18 Jun '22

18 Jun '22

To allow enabling timestamp and specify its type, a new option --tstamp-type=TYPE is added. Recognized values are none (default), gettimeofday, monotonic, monotonic-raw. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- aplay/aplay.1 | 4 ++++ aplay/aplay.c | 32 ++++++++++++++++++++++++++++++++ 2 files changed, 36 insertions(+) diff --git a/aplay/aplay.1 b/aplay/aplay.1 index 3bba59d..d3b7dce 100644 --- a/aplay/aplay.1 +++ b/aplay/aplay.1 @@ -207,6 +207,10 @@ sampling rates, numbers of channels, period and buffer bytes/sizes/times. For raw device hw:X this option basically lists hardware capabilities of the soundcard. .TP +\fI\-\-tstamp\-type=TYPE\fP +Specifies timestamp type inside the software configuration container. +Types are: none (default), gettimeofday, monotonic, monotonic\-raw. +.TP \fI\-\-fatal\-errors\fP Disables recovery attempts when errors (e.g. xrun) are encountered; the aplay process instead aborts immediately. diff --git a/aplay/aplay.c b/aplay/aplay.c index 63a4e34..5d15a32 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -139,6 +139,8 @@ static int use_strftime = 0; volatile static int recycle_capture_file = 0; static long term_c_lflag = -1; static int dump_hw_params = 0; +static int enable_tstamp = 0; +static snd_pcm_tstamp_type_t tstamp_type = SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY; static int fd = -1; static off64_t pbrec_count = LLONG_MAX, fdcount; @@ -244,6 +246,8 @@ _("Usage: %s [OPTION]... [FILE]...\n" " --process-id-file write the process ID here\n" " --use-strftime apply the strftime facility to the output file name\n" " --dump-hw-params dump hw_params of the device\n" +" --tstamp-type=TYPE set timestamp (TYPE: none (default), gettimeofday,\n" +" monotonic, monotonic-raw)\n" " --fatal-errors treat all errors as fatal\n" ) , command); @@ -430,6 +434,7 @@ enum { OPT_PROCESS_ID_FILE, OPT_USE_STRFTIME, OPT_DUMP_HWPARAMS, + OPT_TSTAMP_TYPE, OPT_FATAL_ERRORS, }; @@ -517,6 +522,7 @@ int main(int argc, char *argv[]) {"use-strftime", 0, 0, OPT_USE_STRFTIME}, {"interactive", 0, 0, 'i'}, {"dump-hw-params", 0, 0, OPT_DUMP_HWPARAMS}, + {"tstamp-type", 1, 0, OPT_TSTAMP_TYPE}, {"fatal-errors", 0, 0, OPT_FATAL_ERRORS}, #ifdef CONFIG_SUPPORT_CHMAP {"chmap", 1, 0, 'm'}, @@ -799,6 +805,23 @@ int main(int argc, char *argv[]) case OPT_DUMP_HWPARAMS: dump_hw_params = 1; break; + case OPT_TSTAMP_TYPE: + if (strcasecmp(optarg, "gettimeofday") == 0) { + enable_tstamp = 1; + tstamp_type = SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY; + } else if (strcasecmp(optarg, "monotonic") == 0) { + enable_tstamp = 1; + tstamp_type = SND_PCM_TSTAMP_TYPE_MONOTONIC; + } else if (strcasecmp(optarg, "monotonic-raw") == 0) { + enable_tstamp = 1; + tstamp_type = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW; + } else if (strcasecmp(optarg, "none") == 0) + enable_tstamp = 0; + else { + error(_("unrecognized timestamp type %s"), optarg); + return 1; + } + break; case OPT_FATAL_ERRORS: fatal_errors = 1; break; @@ -1453,6 +1476,15 @@ static void set_params(void) stop_threshold = (double) rate * stop_delay / 1000000; err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold); assert(err >= 0); + if (enable_tstamp) { + err = snd_pcm_sw_params_set_tstamp_mode(handle, swparams, SND_PCM_TSTAMP_ENABLE); + assert(err >= 0); + err = snd_pcm_sw_params_set_tstamp_type(handle, swparams, tstamp_type); + if (err < 0) { + error(_("Unable to set the requested timestamp type.")); + prg_exit(EXIT_FAILURE); + } + } if (snd_pcm_sw_params(handle, swparams) < 0) { error(_("unable to install sw params:")); -- 2.25.1

3 6

[PATCH v2 0/3] firewire: fix minor issues
by Takashi Sakamoto 18 Jun '22

18 Jun '22

This second version of patchset is revised version of previous one[1] to fix mistake of macro usage pointed out by reviewer[2]. As I note, they are not so urgent changes, thus I don't mind postponing until next merge window. [1] https://lore.kernel.org/alsa-devel/20220512111756.103008-1-o-takashi@sakamo… [2] https://lore.kernel.org/alsa-devel/87o7yvpf4t.wl-tiwai@suse.de/ Jiapeng Chong (1): firewire: convert sysfs sprintf/snprintf family to sysfs_emit Lv Ruyi (1): firewire: Fix using uninitialized value Minghao Chi (CGEL ZTE) (1): firewire: use struct_size over open coded arithmetic drivers/firewire/core-device.c | 6 ++---- drivers/firewire/core-transaction.c | 3 ++- 2 files changed, 4 insertions(+), 5 deletions(-) -- 2.34.1

2 7

MPG X570S EDGE MAX WIFI compatibility
by GitHub issues - edited 18 Jun '22

18 Jun '22

alsa-project/alsa-ucm-conf issue #172 was edited from yhojann-cl: Please, add `0db0:151f` to `realtek-alc4080` chip compatibility in MPG X570S EDGE MAX WIFI motherboard: ``` $ lsusb | grep 0db0 Bus 001 Device 003: ID 0db0:151f Micro Star International USB Audio ``` In `ucm2/USB-Audio/USB-Audio.conf` add `151f` to regex of `0db0`: ``` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|151f)))" ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/172 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MPG X570S EDGE MAX WIFI compatibility
by GitHub issues - opened 18 Jun '22

18 Jun '22

alsa-project/alsa-ucm-conf issue #172 was opened from yhojann-cl: Please, add `0db0:151f` to `realtek-alc4080` chip compatibility in MPG X570S EDGE MAX WIFI motherboard: In `ucm2/USB-Audio/USB-Audio.conf` add `151f` to regex of `0db0`: ``` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|151f)))" ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/172 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/2] ASoC: SOF/soundwire: use resume_and_get on component probe
by Pierre-Louis Bossart 18 Jun '22

18 Jun '22

While testing driver bind/unbind sequences, I stumbled on a corner case where the SoundWire bus can be suspended before the ASoC card registration happens. During the registration, register accesses would then lead to timeouts. This does not happen in regular usages where the card registration happens within the 3-second time window before suspend. Adding a simple pm_runtime_resume_and_get() on component probe solves the issue, but experiments showed it was too invasive to add at the ASoC core level, with multiple regressions reported by our CI. This patchset limits the additional resume to the SOF and SoundWire codec drivers. An additional patch for the soundwire/intel component will be sent separately. Pierre-Louis Bossart (2): ASoC: SOF: pcm: use pm_resume_and_get() on component probe ASoC: codecs: soundwire: call pm_runtime_resume() in component probe sound/soc/codecs/max98373.c | 14 +++++++++++++- sound/soc/codecs/rt1308-sdw.c | 12 ++++++++++++ sound/soc/codecs/rt1316-sdw.c | 12 ++++++++++++ sound/soc/codecs/rt700.c | 5 +++++ sound/soc/codecs/rt711-sdca.c | 5 +++++ sound/soc/codecs/rt711.c | 5 +++++ sound/soc/codecs/rt715-sdca.c | 12 ++++++++++++ sound/soc/codecs/rt715.c | 12 ++++++++++++ sound/soc/sof/pcm.c | 11 +++++++++++ 9 files changed, 87 insertions(+), 1 deletion(-) -- 2.34.1

2 9

[PATCH v1] ASoC: ops: Fix integer detection for when max possible values > 1
by Stefan Binding 18 Jun '22

18 Jun '22

The standard snd_soc_info_volsw() allows a two value control to be defined as an integer control only if the control name ends in "Volume". It achieves this by creating a substring if it contains " Volume", and ensuring this exists at the end of the name. The volume substring is then used to decide whether the type is a SNDRV_CTL_ELEM_TYPE_INTEGER or SNDRV_CTL_ELEM_TYPE_BOOLEAN. However this volume substring is only computed for a two value control. This means for controls where there are more than two possible values, the substring is never created, so in this case the substring remains NULL, and the condition yields SNDRV_CTL_ELEM_TYPE_BOOLEAN, even though there are more than 2 possible values. If there are more than 2 possible values for the control, then it should always be an integer control. Fixes: aa2a4b897132 ("ASoC: ops: Fix boolean/integer detection for simple controls") Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/soc/soc-ops.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 0267e39de2a8..1970bda074d8 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -183,17 +183,16 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, if (mc->platform_max && mc->platform_max < max) max = mc->platform_max; - /* Even two value controls ending in Volume should always be integer */ if (max == 1) { + /* Even two value controls ending in Volume should always be integer */ vol_string = strstr(kcontrol->id.name, " Volume"); - if (vol_string && strcmp(vol_string, " Volume")) - vol_string = NULL; - } - - if (!vol_string) - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - else + if (vol_string && !strcmp(vol_string, " Volume")) + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + } else { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + } uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; -- 2.25.1

3 2

[PATCH v2] ASoC: mediatek: mt8186: Fix mutex double unlock in GPIO request
by Fei Shao 18 Jun '22

18 Jun '22

The lockdep mechanism revealed an unbalanced unlocking on MT8186: [ 2.993966] WARNING: bad unlock balance detected! [ 2.993978] ------------------------------------- [ 2.993983] kworker/u16:1/10 is trying to release lock (gpio_request_mutex) at: [ 2.993994] [<ffffffdcd9adebf8>] mt8186_afe_gpio_request+0xf8/0x210 [ 2.994012] but there are no more locks to release! The cause is that the mutex will be double unlocked if dai is unknown during GPIO selection, and this patch fixes it. Fixes: cfa9a966f12a ("ASoC: mediatek: mt8186: support gpio control in platform driver") Signed-off-by: Fei Shao <fshao(a)chromium.org> --- Changes in v2: - Trimmed the commit message sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c index 255ffba637d3..274c0c8ec2f2 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c @@ -230,7 +230,6 @@ int mt8186_afe_gpio_request(struct device *dev, bool enable, sel = enable ? MT8186_AFE_GPIO_PCM_ON : MT8186_AFE_GPIO_PCM_OFF; break; default: - mutex_unlock(&gpio_request_mutex); dev_err(dev, "%s(), invalid dai %d\n", __func__, dai); goto unlock; } -- 2.36.1.476.g0c4daa206d-goog

2 1

[PATCH] ASoC: mediatek: mt8186: Fix mutex double unlock in GPIO request
by Fei Shao 18 Jun '22

18 Jun '22

The lockdep mechanism reveals an unbalanced unlocking on MT8186: [ 2.993966] WARNING: bad unlock balance detected! [ 2.993971] 5.15.46-421fef3b44d7-lockdep #16 Not tainted [ 2.993978] ------------------------------------- [ 2.993983] kworker/u16:1/10 is trying to release lock (gpio_request_mutex) at: [ 2.993994] [<ffffffdcd9adebf8>] mt8186_afe_gpio_request+0xf8/0x210 [ 2.994012] but there are no more locks to release! [ 2.994017] [ 2.994017] other info that might help us debug this: [ 2.994022] 3 locks held by kworker/u16:1/10: [ 2.994028] #0: ffffff80c004a148 ((wq_completion)events_unbound){+.+.}-{0:0}, at: process_one_work+0x1a8/0x364 [ 2.994054] #1: ffffffc0080d3d58 (deferred_probe_work){+.+.}-{0:0}, at: process_one_work+0x1d0/0x364 [ 2.994074] #2: ffffff80c12ab9a0 (&dev->mutex){....}-{3:3}, at: __device_attach+0x44/0x150 [ 2.994096] [ 2.994096] stack backtrace: [ 2.994102] CPU: 6 PID: 10 Comm: kworker/u16:1 Not tainted 5.15.46-421fef3b44d7-lockdep #16 a2af0e2c6b3eeab6799b5aa8091c1d05d3a7bbb1 [ 2.994113] Hardware name: Google Kingler board (DT) [ 2.994120] Workqueue: events_unbound deferred_probe_work_func [ 2.994130] Call trace: [ 2.994135] dump_backtrace+0x0/0x1d4 [ 2.994144] show_stack+0x20/0x2c [ 2.994152] dump_stack_lvl+0x78/0x9c [ 2.994163] dump_stack+0x18/0x44 [ 2.994172] print_unlock_imbalance_bug+0xf0/0xf4 [ 2.994184] lock_release+0x150/0x388 [ 2.994193] __mutex_unlock_slowpath+0x4c/0x1bc [ 2.994202] mutex_unlock+0x44/0x50 [ 2.994210] mt8186_afe_gpio_request+0xf8/0x210 [ 2.994220] mt8186_afe_gpio_init+0xcc/0x134 [ 2.994230] mt8186_mt6366_rt1019_rt5682s_dev_probe+0x14c/0x220 [ 2.994238] platform_probe+0xb0/0xd0 [ 2.994247] really_probe+0xcc/0x2c8 [ 2.994253] __driver_probe_device+0xbc/0xe8 [ 2.994264] driver_probe_device+0x48/0xf0 [ 2.994271] __device_attach_driver+0xa0/0xc8 [ 2.994278] bus_for_each_drv+0x8c/0xd8 [ 2.994288] __device_attach+0xc4/0x150 [ 2.994298] device_initial_probe+0x1c/0x28 [ 2.994308] bus_probe_device+0x3c/0xa0 [ 2.994318] deferred_probe_work_func+0xa0/0xe0 [ 2.994325] process_one_work+0x208/0x364 [ 2.994334] worker_thread+0x288/0x3fc [ 2.994343] kthread+0x140/0x160 [ 2.994351] ret_from_fork+0x10/0x20 The cause is that the mutex will be double unlocked if dai is unknown during GPIO selection, and this patch fixes it. Fixes: cfa9a966f12a ("ASoC: mediatek: mt8186: support gpio control in platform driver") Signed-off-by: Fei Shao <fshao(a)chromium.org> --- sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c index 255ffba637d3..274c0c8ec2f2 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c @@ -230,7 +230,6 @@ int mt8186_afe_gpio_request(struct device *dev, bool enable, sel = enable ? MT8186_AFE_GPIO_PCM_ON : MT8186_AFE_GPIO_PCM_OFF; break; default: - mutex_unlock(&gpio_request_mutex); dev_err(dev, "%s(), invalid dai %d\n", __func__, dai); goto unlock; } -- 2.36.1.476.g0c4daa206d-goog

2 3

[PATCH 0/3] ASoC: SOF: Intel: fix resume from hibernate
by Pierre-Louis Bossart 18 Jun '22

18 Jun '22

The enablement of IMR-based DSP boot helped reduce resume latency, but unfortunately the context is not saved in S4 and S5 which leads to multiple reports of boot failures. This patchset forces a full firmware reload/reboot when resuming from S4/S5 and restores functionality. Pierre-Louis Bossart (3): ASoC: SOF: pm: add explicit behavior for ACPI S1 and S2 ASoC: SOF: pm: add definitions for S4 and S5 states ASoC: SOF: Intel: disable IMR boot when resuming from ACPI S4 and S5 states sound/soc/sof/intel/hda-loader.c | 3 ++- sound/soc/sof/pm.c | 21 ++++++++++++++++++++- sound/soc/sof/sof-priv.h | 2 ++ 3 files changed, 24 insertions(+), 2 deletions(-) -- 2.34.1

2 4

[PATCH] ASoC: ops: Fix multiple value control type
by Sameer Pujar 18 Jun '22

18 Jun '22

The commit aa2a4b897132("ASoC: ops: Fix boolean/integer detection for simple controls") fixes false positives with controls not ending in " Volume" string. But it now forces boolean type for the multi value controls. Fix this by adding a max check before assigning types. Fixes: aa2a4b897132("ASoC: ops: Fix boolean/integer detection for simple controls") Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 0267e39..21be8e8 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -190,7 +190,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, vol_string = NULL; } - if (!vol_string) + if (!vol_string && max == 1) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; -- 2.7.4

3 2

[PATCH v2 0/3] ASoC: rockchip: i2s: switch BCLK to GPIO
by Judy Hsiao 17 Jun '22

17 Jun '22

The patches series is to fix the unexpected large DC output voltage of Max98357a to burn the speakers on rockchip platform when BCLK and SD_MODE are ON but LRCLK is OFF. Changes Since V2: -- Add documents of i2s pinctrl-names. -- Fix dtsi syntax error. -- Include the dtsi change and the driver change in the same series. -- Ensure that driver gets both bclk_on and bclk_off states before using them. Judy Hsiao (3): ASoC: rockchip: i2s: switch BCLK to GPIO arm64: dts: rk3399: i2s: switch BCLK to GPIO ASoC: dt-bindings: rockchip: Document pinctrl-names for i2s .../bindings/sound/rockchip-i2s.yaml | 7 + .../boot/dts/rockchip/rk3399-gru-scarlet.dtsi | 10 ++ arch/arm64/boot/dts/rockchip/rk3399.dtsi | 25 +++- sound/soc/rockchip/rockchip_i2s.c | 134 ++++++++++++++---- 4 files changed, 149 insertions(+), 27 deletions(-) -- 2.36.1.476.g0c4daa206d-goog

4 14

[PATCH] ASoC: SOF: Intel: hda: Fix compressed stream position tracking
by Pierre-Louis Bossart 17 Jun '22

17 Jun '22

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Commit 288fad2f71fa ("ASoC: SOF: Intel: hda: add quirks for HDAudio DMA position information") modified the PCM path only, but left the compressed data patch using an obsolete option. Move the functionality in a helper that can be called for both PCM and compressed data. Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Fixes: 288fad2f71fa ("ASoC: SOF: Intel: hda: add quirks for HDAudio DMA position information") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-pcm.c | 74 +------------------------ sound/soc/sof/intel/hda-stream.c | 94 ++++++++++++++++++++++++++++++-- sound/soc/sof/intel/hda.h | 3 + 3 files changed, 94 insertions(+), 77 deletions(-) diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index dc1f743730c0d..6888e0a4665d2 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -192,79 +192,7 @@ snd_pcm_uframes_t hda_dsp_pcm_pointer(struct snd_sof_dev *sdev, goto found; } - switch (sof_hda_position_quirk) { - case SOF_HDA_POSITION_QUIRK_USE_SKYLAKE_LEGACY: - /* - * This legacy code, inherited from the Skylake driver, - * mixes DPIB registers and DPIB DDR updates and - * does not seem to follow any known hardware recommendations. - * It's not clear e.g. why there is a different flow - * for capture and playback, the only information that matters is - * what traffic class is used, and on all SOF-enabled platforms - * only VC0 is supported so the work-around was likely not necessary - * and quite possibly wrong. - */ - - /* DPIB/posbuf position mode: - * For Playback, Use DPIB register from HDA space which - * reflects the actual data transferred. - * For Capture, Use the position buffer for pointer, as DPIB - * is not accurate enough, its update may be completed - * earlier than the data written to DDR. - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - } else { - /* - * For capture stream, we need more workaround to fix the - * position incorrect issue: - * - * 1. Wait at least 20us before reading position buffer after - * the interrupt generated(IOC), to make sure position update - * happens on frame boundary i.e. 20.833uSec for 48KHz. - * 2. Perform a dummy Read to DPIB register to flush DMA - * position value. - * 3. Read the DMA Position from posbuf. Now the readback - * value should be >= period boundary. - */ - usleep_range(20, 21); - snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - pos = snd_hdac_stream_get_pos_posbuf(hstream); - } - break; - case SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS: - /* - * In case VC1 traffic is disabled this is the recommended option - */ - pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - break; - case SOF_HDA_POSITION_QUIRK_USE_DPIB_DDR_UPDATE: - /* - * This is the recommended option when VC1 is enabled. - * While this isn't needed for SOF platforms it's added for - * consistency and debug. - */ - pos = snd_hdac_stream_get_pos_posbuf(hstream); - break; - default: - dev_err_once(sdev->dev, "hda_position_quirk value %d not supported\n", - sof_hda_position_quirk); - pos = 0; - break; - } - - if (pos >= hstream->bufsize) - pos = 0; - + pos = hda_dsp_stream_get_position(hstream, substream->stream, true); found: pos = bytes_to_frames(substream->runtime, pos); diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index daeb64c495e40..d95ae17e81cc4 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -707,12 +707,13 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev) } static void -hda_dsp_set_bytes_transferred(struct hdac_stream *hstream, u64 buffer_size) +hda_dsp_compr_bytes_transferred(struct hdac_stream *hstream, int direction) { + u64 buffer_size = hstream->bufsize; u64 prev_pos, pos, num_bytes; div64_u64_rem(hstream->curr_pos, buffer_size, &prev_pos); - pos = snd_hdac_stream_get_pos_posbuf(hstream); + pos = hda_dsp_stream_get_position(hstream, direction, false); if (pos < prev_pos) num_bytes = (buffer_size - prev_pos) + pos; @@ -748,8 +749,7 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) if (s->substream && sof_hda->no_ipc_position) { snd_sof_pcm_period_elapsed(s->substream); } else if (s->cstream) { - hda_dsp_set_bytes_transferred(s, - s->cstream->runtime->buffer_size); + hda_dsp_compr_bytes_transferred(s, s->cstream->direction); snd_compr_fragment_elapsed(s->cstream); } } @@ -1009,3 +1009,89 @@ void hda_dsp_stream_free(struct snd_sof_dev *sdev) devm_kfree(sdev->dev, hda_stream); } } + +snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, + int direction, bool can_sleep) +{ + struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream); + struct sof_intel_hda_stream *hda_stream = hstream_to_sof_hda_stream(hext_stream); + struct snd_sof_dev *sdev = hda_stream->sdev; + snd_pcm_uframes_t pos; + + switch (sof_hda_position_quirk) { + case SOF_HDA_POSITION_QUIRK_USE_SKYLAKE_LEGACY: + /* + * This legacy code, inherited from the Skylake driver, + * mixes DPIB registers and DPIB DDR updates and + * does not seem to follow any known hardware recommendations. + * It's not clear e.g. why there is a different flow + * for capture and playback, the only information that matters is + * what traffic class is used, and on all SOF-enabled platforms + * only VC0 is supported so the work-around was likely not necessary + * and quite possibly wrong. + */ + + /* DPIB/posbuf position mode: + * For Playback, Use DPIB register from HDA space which + * reflects the actual data transferred. + * For Capture, Use the position buffer for pointer, as DPIB + * is not accurate enough, its update may be completed + * earlier than the data written to DDR. + */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + } else { + /* + * For capture stream, we need more workaround to fix the + * position incorrect issue: + * + * 1. Wait at least 20us before reading position buffer after + * the interrupt generated(IOC), to make sure position update + * happens on frame boundary i.e. 20.833uSec for 48KHz. + * 2. Perform a dummy Read to DPIB register to flush DMA + * position value. + * 3. Read the DMA Position from posbuf. Now the readback + * value should be >= period boundary. + */ + if (can_sleep) + usleep_range(20, 21); + + snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + pos = snd_hdac_stream_get_pos_posbuf(hstream); + } + break; + case SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS: + /* + * In case VC1 traffic is disabled this is the recommended option + */ + pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + break; + case SOF_HDA_POSITION_QUIRK_USE_DPIB_DDR_UPDATE: + /* + * This is the recommended option when VC1 is enabled. + * While this isn't needed for SOF platforms it's added for + * consistency and debug. + */ + pos = snd_hdac_stream_get_pos_posbuf(hstream); + break; + default: + dev_err_once(sdev->dev, "hda_position_quirk value %d not supported\n", + sof_hda_position_quirk); + pos = 0; + break; + } + + if (pos >= hstream->bufsize) + pos = 0; + + return pos; +} diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index a3118499e34fc..dc713c20ba1d9 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -565,6 +565,9 @@ int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev); bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev); +snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, + int direction, bool can_sleep); + struct hdac_ext_stream * hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags); int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag); -- 2.34.1

2 1

No sound on AMD audio device
by GitHub issues - edited 17 Jun '22

17 Jun '22

alsa-project/alsa-lib issue #241 was edited from chvmq: I have a laptop huawei 14 amd 2021 I don't know where write about it. maybe there I tried ubuntu 22.01 | arch-linux and no sound at all Issue URL : https://github.com/alsa-project/alsa-lib/issues/241 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsa-lib parser error: "sequence command 'devdisall' is ignored" when attempting alsactl restore, nrestore, init, etc.
by GitHub issues - opened 17 Jun '22

17 Jun '22

alsa-project/alsa-lib issue #240 was opened from salperinlea: This may play into a much larger issue I have been having with an Intel tiger lake soundcard, which cannot be activated after my machine has been suspended, but which always loads on boot. In an attempt to solve this error, I have tried to use alsactl restore, at which point I get the error: <code><pre> sudo alsactl restore alsa-lib parser.c:1194:(parse_sequence) error: sequence command 'devdisall' is ignored alsa-lib parser.c:1194:(parse_sequence) error: sequence command 'devdisall' is ignored </code></pre> I believe the reason that this repeats is that I have two soundcards; one is an Nvidia card, hard-wired to a dGPU and the HDMI port. Info: Arch Linux 251.2, Kernel 5.18.3, KDE 5.25 DE. <code><pre> lspci | grep 'Audio' 0000:00:1f.3 Multimedia audio controller: Intel Corporation Tiger Lake-H HD Audio Controller (rev 11) 0000:01:00.1 Audio device: NVIDIA Corporation Device 228e (rev a1) </code></pre> I am not sure what other info may be of use, but I will provide it upon request. Issue URL : https://github.com/alsa-project/alsa-lib/issues/240 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 00/13] ASoC: SOF/Intel: remove __func__ from dev_dbg()
by Pierre-Louis Bossart 17 Jun '22

17 Jun '22

Cleanups suggested by Greg KH during SoundWire reviews, since the __func__ information can be added with the dyndbg kernel parameter [1]. The first two patches change the error level in cases where the use of dev_dbg() was inconsistent with an aborted programming sequence. [1] https://lore.kernel.org/alsa-devel/20220610023537.27223-2-yung-chuan.liao@l… Pierre-Louis Bossart (13): ASoC: SOF: Intel: hda-dsp: report error on power-up/down ASoC: SOF: Intel: hda-stream: report error on stream not opened ASoC: SOF: Intel: hda-dai: remove use of __func__ in dev_dbg ASoC: SOF: Intel: hda-stream: remove use of __func__ in dev_dbg ASoC: SOF: Intel: mtl: remove use of __func__ in dev_dbg ASoC: SOF: ipc3-dtrace: remove use of __func__ in dev_dbg ASoC: SOF: ipc3-loader: remove use of __func__ in dev_dbg ASoC: SOF: ipc3-topology: remove use of __func__ in dev_dbg ASoC: SOF: ipc4-topology remove use of __func__ in dev_dbg ASoC: SOF: sof-client: remove use of __func__ in dev_dbg ASoC: SOF: ipc4: remove use of __func__ in dev_dbg ASoC: Intel: boards: hda: remove use of __func__ in dev_dbg ASoC: Intel: boards: sof_sdw: remove use of __func__ in dev_dbg sound/soc/intel/boards/hda_dsp_common.c | 4 ++-- sound/soc/intel/boards/skl_hda_dsp_generic.c | 4 ++-- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/sof/intel/hda-dai.c | 6 +++--- sound/soc/sof/intel/hda-dsp.c | 4 ++-- sound/soc/sof/intel/hda-stream.c | 6 +++--- sound/soc/sof/intel/mtl.c | 3 +-- sound/soc/sof/ipc3-dtrace.c | 5 ++--- sound/soc/sof/ipc3-loader.c | 11 +++++------ sound/soc/sof/ipc3-topology.c | 4 ++-- sound/soc/sof/ipc4-topology.c | 18 +++++++++--------- sound/soc/sof/ipc4.c | 2 +- sound/soc/sof/sof-client.c | 4 ++-- 13 files changed, 35 insertions(+), 38 deletions(-) -- 2.34.1

3 15

[PATCH 00/13] ASoC: Intel: Chromebooks: remap jack pins
by Pierre-Louis Bossart 17 Jun '22

17 Jun '22

This patchset adds the missing jack pin remapping needed by upstream UCM on Chromebooks, suggested by Jaroslav Kysela [1]. These patches were succesfully tested on GeminiLake and JasperLake platforms, and generalized to other Chromebooks. [1] https://mailman.alsa-project.org/pipermail/sound-open-firmware/2022-June/00… Pierre-Louis Bossart (13): ASoC: Intel: skl_nau88l25_max98357a: remap jack pins ASoC: Intel: skl_nau88l25_ssm4567: remap jack pins ASoC: Intel: kbl_rt5663_max98927: remap jack pins ASoC: Intel: kbl_da7219_max98357a: remap jack pins ASoC: Intel: kbl_da7219_max98927: remap jack pins ASoC: Intel: kbl_rt5663_rt5514_max98927: remap jack pins ASoC: Intel: bxt_da7219_max98357a: remap jack pins ASoC: Intel: glk_rt5682_max98357a: remap jack pins ASoC: Intel: cml_rt1011_rt5682: remap jack pins ASoC: Intel: sof_cs42l42: remap jack pins ASoC: Intel: sof_da7219_max98373: remap jack pins ASoC: Intel: sof_nau8825: remap jack pins ASoC: Intel: sof_rt5682: remap jack pins sound/soc/intel/boards/bxt_da7219_max98357a.c | 21 +++++++++++++---- sound/soc/intel/boards/cml_rt1011_rt5682.c | 23 +++++++++++++++---- sound/soc/intel/boards/glk_rt5682_max98357a.c | 21 +++++++++++++---- sound/soc/intel/boards/kbl_da7219_max98357a.c | 21 +++++++++++++---- sound/soc/intel/boards/kbl_da7219_max98927.c | 21 +++++++++++++---- sound/soc/intel/boards/kbl_rt5663_max98927.c | 21 +++++++++++++---- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 21 +++++++++++++---- .../soc/intel/boards/skl_nau88l25_max98357a.c | 19 ++++++++++++--- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 19 ++++++++++++--- sound/soc/intel/boards/sof_cs42l42.c | 23 +++++++++++++++---- sound/soc/intel/boards/sof_da7219_max98373.c | 23 +++++++++++++++---- sound/soc/intel/boards/sof_nau8825.c | 23 +++++++++++++++---- sound/soc/intel/boards/sof_rt5682.c | 23 +++++++++++++++---- 13 files changed, 224 insertions(+), 55 deletions(-) -- 2.34.1

2 14

[PATCH 0/2] ASoC: minor kernel-doc corrections
by Pierre-Louis Bossart 17 Jun '22

17 Jun '22

Remove warnings with make W=1 Pierre-Louis Bossart (2): ASoC: cs4270: update kernel-doc ASoC: sunxi: sun4i-i2s: update kernel-doc sound/soc/codecs/cs4270.c | 1 - sound/soc/sunxi/sun4i-i2s.c | 2 ++ 2 files changed, 2 insertions(+), 1 deletion(-) -- 2.34.1

2 3

[PATCH] ASoC: twl4030: Drop legacy, non DT boot support
by Peter Ujfalusi 17 Jun '22

17 Jun '22

Legacy or non DT boot is no longer possible on systems where the tw4030/5030 is used. Drop the support for handling legacy pdata and replace it with a local board_params struct to allow further cleanups on the mfd side. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> --- sound/soc/codecs/twl4030.c | 101 ++++++++++++++++++++----------------- 1 file changed, 54 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 0ba3546ef870..87b58017094b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -34,6 +34,14 @@ #define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1) +struct twl4030_board_params { + unsigned int digimic_delay; /* in ms */ + unsigned int ramp_delay_value; + unsigned int offset_cncl_path; + unsigned int hs_extmute:1; + int hs_extmute_gpio; +}; + /* codec private data */ struct twl4030_priv { unsigned int codec_powered; @@ -58,7 +66,7 @@ struct twl4030_priv { u8 carkitl_enabled, carkitr_enabled; u8 ctl_cache[TWL4030_REG_PRECKR_CTL - TWL4030_REG_EAR_CTL + 1]; - struct twl4030_codec_data *pdata; + struct twl4030_board_params *board_params; }; static void tw4030_init_ctl_cache(struct twl4030_priv *twl4030) @@ -193,73 +201,71 @@ static void twl4030_codec_enable(struct snd_soc_component *component, int enable udelay(10); } -static void twl4030_setup_pdata_of(struct twl4030_codec_data *pdata, - struct device_node *node) +static void +twl4030_get_board_param_values(struct twl4030_board_params *board_params, + struct device_node *node) { int value; - of_property_read_u32(node, "ti,digimic_delay", - &pdata->digimic_delay); - of_property_read_u32(node, "ti,ramp_delay_value", - &pdata->ramp_delay_value); - of_property_read_u32(node, "ti,offset_cncl_path", - &pdata->offset_cncl_path); + of_property_read_u32(node, "ti,digimic_delay", &board_params->digimic_delay); + of_property_read_u32(node, "ti,ramp_delay_value", &board_params->ramp_delay_value); + of_property_read_u32(node, "ti,offset_cncl_path", &board_params->offset_cncl_path); if (!of_property_read_u32(node, "ti,hs_extmute", &value)) - pdata->hs_extmute = value; + board_params->hs_extmute = value; - pdata->hs_extmute_gpio = of_get_named_gpio(node, - "ti,hs_extmute_gpio", 0); - if (gpio_is_valid(pdata->hs_extmute_gpio)) - pdata->hs_extmute = 1; + board_params->hs_extmute_gpio = of_get_named_gpio(node, "ti,hs_extmute_gpio", 0); + if (gpio_is_valid(board_params->hs_extmute_gpio)) + board_params->hs_extmute = 1; } -static struct twl4030_codec_data *twl4030_get_pdata(struct snd_soc_component *component) +static struct twl4030_board_params* +twl4030_get_board_params(struct snd_soc_component *component) { - struct twl4030_codec_data *pdata = dev_get_platdata(component->dev); + struct twl4030_board_params *board_params = NULL; struct device_node *twl4030_codec_node = NULL; twl4030_codec_node = of_get_child_by_name(component->dev->parent->of_node, "codec"); - if (!pdata && twl4030_codec_node) { - pdata = devm_kzalloc(component->dev, - sizeof(struct twl4030_codec_data), - GFP_KERNEL); - if (!pdata) { + if (twl4030_codec_node) { + board_params = devm_kzalloc(component->dev, + sizeof(struct twl4030_board_params), + GFP_KERNEL); + if (!board_params) { of_node_put(twl4030_codec_node); return NULL; } - twl4030_setup_pdata_of(pdata, twl4030_codec_node); + twl4030_get_board_param_values(board_params, twl4030_codec_node); of_node_put(twl4030_codec_node); } - return pdata; + return board_params; } static void twl4030_init_chip(struct snd_soc_component *component) { - struct twl4030_codec_data *pdata; + struct twl4030_board_params *board_params; struct twl4030_priv *twl4030 = snd_soc_component_get_drvdata(component); u8 reg, byte; int i = 0; - pdata = twl4030_get_pdata(component); + board_params = twl4030_get_board_params(component); - if (pdata && pdata->hs_extmute) { - if (gpio_is_valid(pdata->hs_extmute_gpio)) { + if (board_params && board_params->hs_extmute) { + if (gpio_is_valid(board_params->hs_extmute_gpio)) { int ret; - if (!pdata->hs_extmute_gpio) + if (!board_params->hs_extmute_gpio) dev_warn(component->dev, "Extmute GPIO is 0 is this correct?\n"); - ret = gpio_request_one(pdata->hs_extmute_gpio, + ret = gpio_request_one(board_params->hs_extmute_gpio, GPIOF_OUT_INIT_LOW, "hs_extmute"); if (ret) { dev_err(component->dev, "Failed to get hs_extmute GPIO\n"); - pdata->hs_extmute_gpio = -1; + board_params->hs_extmute_gpio = -1; } } else { u8 pin_mux; @@ -290,14 +296,14 @@ static void twl4030_init_chip(struct snd_soc_component *component) twl4030_write(component, TWL4030_REG_ARXR2_APGA_CTL, 0x32); /* Machine dependent setup */ - if (!pdata) + if (!board_params) return; - twl4030->pdata = pdata; + twl4030->board_params = board_params; reg = twl4030_read(component, TWL4030_REG_HS_POPN_SET); reg &= ~TWL4030_RAMP_DELAY; - reg |= (pdata->ramp_delay_value << 2); + reg |= (board_params->ramp_delay_value << 2); twl4030_write(component, TWL4030_REG_HS_POPN_SET, reg); /* initiate offset cancellation */ @@ -305,7 +311,7 @@ static void twl4030_init_chip(struct snd_soc_component *component) reg = twl4030_read(component, TWL4030_REG_ANAMICL); reg &= ~TWL4030_OFFSET_CNCL_SEL; - reg |= pdata->offset_cncl_path; + reg |= board_params->offset_cncl_path; twl4030_write(component, TWL4030_REG_ANAMICL, reg | TWL4030_CNCL_OFFSET_START); @@ -692,7 +698,7 @@ static void headset_ramp(struct snd_soc_component *component, int ramp) { unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = snd_soc_component_get_drvdata(component); - struct twl4030_codec_data *pdata = twl4030->pdata; + struct twl4030_board_params *board_params = twl4030->board_params; /* Base values for ramp delay calculation: 2^19 - 2^26 */ unsigned int ramp_base[] = {524288, 1048576, 2097152, 4194304, 8388608, 16777216, 33554432, 67108864}; @@ -705,9 +711,9 @@ static void headset_ramp(struct snd_soc_component *component, int ramp) /* Enable external mute control, this dramatically reduces * the pop-noise */ - if (pdata && pdata->hs_extmute) { - if (gpio_is_valid(pdata->hs_extmute_gpio)) { - gpio_set_value(pdata->hs_extmute_gpio, 1); + if (board_params && board_params->hs_extmute) { + if (gpio_is_valid(board_params->hs_extmute_gpio)) { + gpio_set_value(board_params->hs_extmute_gpio, 1); } else { hs_pop |= TWL4030_EXTMUTE; twl4030_write(component, TWL4030_REG_HS_POPN_SET, hs_pop); @@ -741,9 +747,9 @@ static void headset_ramp(struct snd_soc_component *component, int ramp) } /* Disable external mute */ - if (pdata && pdata->hs_extmute) { - if (gpio_is_valid(pdata->hs_extmute_gpio)) { - gpio_set_value(pdata->hs_extmute_gpio, 0); + if (board_params && board_params->hs_extmute) { + if (gpio_is_valid(board_params->hs_extmute_gpio)) { + gpio_set_value(board_params->hs_extmute_gpio, 0); } else { hs_pop &= ~TWL4030_EXTMUTE; twl4030_write(component, TWL4030_REG_HS_POPN_SET, hs_pop); @@ -806,10 +812,10 @@ static int digimic_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct twl4030_priv *twl4030 = snd_soc_component_get_drvdata(component); - struct twl4030_codec_data *pdata = twl4030->pdata; + struct twl4030_board_params *board_params = twl4030->board_params; - if (pdata && pdata->digimic_delay) - twl4030_wait_ms(pdata->digimic_delay); + if (board_params && board_params->digimic_delay) + twl4030_wait_ms(board_params->digimic_delay); return 0; } @@ -2168,10 +2174,11 @@ static int twl4030_soc_probe(struct snd_soc_component *component) static void twl4030_soc_remove(struct snd_soc_component *component) { struct twl4030_priv *twl4030 = snd_soc_component_get_drvdata(component); - struct twl4030_codec_data *pdata = twl4030->pdata; + struct twl4030_board_params *board_params = twl4030->board_params; - if (pdata && pdata->hs_extmute && gpio_is_valid(pdata->hs_extmute_gpio)) - gpio_free(pdata->hs_extmute_gpio); + if (board_params && board_params->hs_extmute && + gpio_is_valid(board_params->hs_extmute_gpio)) + gpio_free(board_params->hs_extmute_gpio); } static const struct snd_soc_component_driver soc_component_dev_twl4030 = { -- 2.36.1

2 1

ALSA 1.2.7.1 release
by Jaroslav Kysela 17 Jun '22

17 Jun '22

Hello all, new ALSA userspace packages were released. 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-lib alsa-ucm-conf alsa-plugins Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.7_v1.2.7.1 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.

1 0

[PATCH 0/5] ALSA: rawmidi: Make code robust for external calls
by Takashi Iwai 17 Jun '22

17 Jun '22

Hi, here is a small patch set for ALSA core rawmidi code to make the code a bit more robust, especially for the case where the exported functions get called from the external drivers. Currently most of those functions assume naively that they aren't called at a wrong timing. With the patch set, it tries to harden a bit not to hit serious breakage. Takashi === Takashi Iwai (5): ALSA: rawmidi: Make internal functions local static ALSA: rawmidi: Move lock to snd_rawmidi_substream ALSA: rawmidi: Take open_mutex around parameter changes ALSA: rawmidi: Check stream state at exported functions ALSA: rawmidi: Take buffer refcount while draining output include/sound/rawmidi.h | 6 +- sound/core/rawmidi.c | 274 +++++++++++++++++++++++++--------------- 2 files changed, 170 insertions(+), 110 deletions(-) -- 2.35.3

1 5

[PATCH] ALSA: hda/realtek: Add quirk for Clevo PD70PNT
by Tim Crawford 17 Jun '22

17 Jun '22

Fixes speaker output and headset detection on Clevo PD70PNT. Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b937f63d0d09..ff9a09a670ed 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2634,6 +2634,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67f1, "Clevo PC70H[PRS]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x67f5, "Clevo PD70PN[NRT]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x70d1, "Clevo PC70[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170SM", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x7715, "Clevo X170KM-G", ALC1220_FIXUP_CLEVO_PB51ED), -- 2.35.3

2 1

Silent speaker on ThinkPad T14s Gen 2i
by GitHub issues - opened 17 Jun '22

17 Jun '22

alsa-project/alsa-ucm-conf issue #171 was opened from joakim-tjernlund: Noted that speakers are silent and not sure when that happened. I note that pavucontrol shows sound playing on HDMI / Display port 2 (which is unplugged) when I test audio in MATE volume control. It seems that HDMI now is default even though it is unplugged ? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/171 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Can anyone test with AMD onboard or HDMI/DP?
by Takashi Iwai 17 Jun '22

17 Jun '22

Hi, can anyone have an AMD onboard audio device and/or AMD HDMI/DP output for testing a patch below with 5.18.x or 5.19-rc kernels? It's a pending fix (for 5.18+), but currently it can't be verified whether it causes a regression on the actual audio I/O (while it fixes the kernel crash). If the generic allocator still doesn't work as expected here, it should show some audio stuttering or such effect. Thanks! Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: memalloc: Drop x86-specific hack for WC allocations The recent report for a crash on Haswell machines implied that the x86-specific (rather hackish) implementation for write-cache memory buffer allocation in ALSA core is buggy with the recent kernel in some corner cases. This patch drops the x86-specific implementation and uses the standard dma_alloc_wc() & co generically for avoiding the bug and also for simplification. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216112 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 23 +---------------------- 1 file changed, 1 insertion(+), 22 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 15dc7160ba34..8cfdaee77905 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -431,33 +431,17 @@ static const struct snd_malloc_ops snd_dma_iram_ops = { */ static void *snd_dma_dev_alloc(struct snd_dma_buffer *dmab, size_t size) { - void *p; - - p = dma_alloc_coherent(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); -#ifdef CONFIG_X86 - if (p && dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC) - set_memory_wc((unsigned long)p, PAGE_ALIGN(size) >> PAGE_SHIFT); -#endif - return p; + return dma_alloc_coherent(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); } static void snd_dma_dev_free(struct snd_dma_buffer *dmab) { -#ifdef CONFIG_X86 - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC) - set_memory_wb((unsigned long)dmab->area, - PAGE_ALIGN(dmab->bytes) >> PAGE_SHIFT); -#endif dma_free_coherent(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); } static int snd_dma_dev_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { -#ifdef CONFIG_X86 - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC) - area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); -#endif return dma_mmap_coherent(dmab->dev.dev, area, dmab->area, dmab->addr, dmab->bytes); } @@ -471,10 +455,6 @@ static const struct snd_malloc_ops snd_dma_dev_ops = { /* * Write-combined pages */ -#ifdef CONFIG_X86 -/* On x86, share the same ops as the standard dev ops */ -#define snd_dma_wc_ops snd_dma_dev_ops -#else /* CONFIG_X86 */ static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) { return dma_alloc_wc(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); @@ -497,7 +477,6 @@ static const struct snd_malloc_ops snd_dma_wc_ops = { .free = snd_dma_wc_free, .mmap = snd_dma_wc_mmap, }; -#endif /* CONFIG_X86 */ #ifdef CONFIG_SND_DMA_SGBUF static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size); -- 2.35.3

2 3

[PATCH 00/23] tests: refine test implementation
by GitHub pull_request - opened 17 Jun '22

17 Jun '22

alsa-project/alsa-gobject pull request #76 was opened from takaswie: Current implementation of test just supports GObject-derived object, enumerations, and flags. It's not possible to test the other types of glib/gobject elements such as boxed structure. This patchset refines test implementation. Some helper functions for boxed structure and namespace/object functions are added. The existent helper function is renamed as 'test_object' and rewritten to walk through Python MRO hierarchy to check properties, virtual methods, and signals, defined by both GObject-derived objects and interfaces. I note that the tests are not done to execute actual symbols. They are just to check they are available or not via interface described in metadata of GObject Introspection. ``` Takashi Sakamoto (23): tests: rename helper function to test object interface tests: add helper function to test enumerations and flags tests: add test script for ALSACtl.ElemId boxed structure tests: add test script for ALSATimer.DeviceId boxed structure tests: add test script for ALSATimer.TickEvent boxed structure tests: add test script for ALSATimer.TstampEvent boxed structure tests: add test script for ALSASeq.Addr boxed structure tests: add test script for ALSASeq.EventCntr boxed structure tests: add test script for ALSASeq.Event boxed structure tests: add test script for ALSASeq.EventDataConnect boxed object tests: add test script for ALSASeq.EventDataCtl boxed structure tests: add test script for ALSASeq.EventDataNote boxed structure tests: add test script for ALSASeq.EventDataQueue boxed structure tests: add test script for ALSASeq.EventDataResult boxed structure tests: add test script for ALSASeq.RemoveFilter boxed structure tests: test object type instead of its instance tests: refine helper function to test object tests: test virtual functions in object tests: add test script for ALSACtl.ElemInfoCommon interface tests: add test script for ALSACtl.ElemInfoSingleArray interface tests: add test script for ALSAHwdep.DeviceCommon interface tests: add test script for ALSASeq.QueueTimerCommon interface tests: add test scripts for namespace functions tests/alsactl-card | 8 ++-- tests/alsactl-card-info | 7 +-- tests/alsactl-elem-id | 14 +++--- tests/alsactl-elem-info-boolean | 8 ++-- tests/alsactl-elem-info-bytes | 8 ++-- tests/alsactl-elem-info-common | 24 +++++++++++ tests/alsactl-elem-info-enumerated | 10 +++-- tests/alsactl-elem-info-iec60958 | 8 ++-- tests/alsactl-elem-info-integer | 14 +++--- tests/alsactl-elem-info-integer64 | 14 +++--- tests/alsactl-elem-info-single-array | 21 +++++++++ tests/alsactl-elem-value | 7 +-- tests/alsactl-enums | 11 ++--- tests/alsactl-functions | 26 +++++++++++ tests/alsahwdep-device-common | 32 ++++++++++++++ tests/alsahwdep-device-info | 7 +-- tests/alsahwdep-enums | 11 ++--- tests/alsahwdep-functions | 27 ++++++++++++ tests/alsarawmidi-enums | 11 ++--- tests/alsarawmidi-functions | 27 ++++++++++++ tests/alsarawmidi-stream-pair | 8 ++-- tests/alsarawmidi-substream-info | 7 +-- tests/alsarawmidi-substream-params | 7 +-- tests/alsarawmidi-substream-status | 7 +-- tests/alsaseq-addr | 21 +++++++++ tests/alsaseq-client-info | 7 +-- tests/alsaseq-client-pool | 7 +-- tests/alsaseq-enums | 13 +++--- tests/alsaseq-event | 61 ++++++++++++++++++++++++++ tests/alsaseq-event-cntr | 18 ++++++++ tests/alsaseq-event-data-connect | 21 +++++++++ tests/alsaseq-event-data-ctl | 23 ++++++++++ tests/alsaseq-event-data-note | 27 ++++++++++++ tests/alsaseq-event-data-queue | 33 ++++++++++++++ tests/alsaseq-event-data-result | 24 +++++++++++ tests/alsaseq-functions | 35 +++++++++++++++ tests/alsaseq-port-info | 7 +-- tests/alsaseq-queue-info | 7 +-- tests/alsaseq-queue-status | 7 +-- tests/alsaseq-queue-tempo | 7 +-- tests/alsaseq-queue-timer-alsa | 14 +++--- tests/alsaseq-queue-timer-common | 22 ++++++++++ tests/alsaseq-remove-filter | 21 +++++++++ tests/alsaseq-subscribe-data | 7 +-- tests/alsaseq-system-info | 7 +-- tests/alsaseq-user-client | 13 ++++-- tests/alsatimer-device-id | 18 +++----- tests/alsatimer-device-info | 7 +-- tests/alsatimer-device-params | 7 +-- tests/alsatimer-device-status | 7 +-- tests/alsatimer-enums | 10 ++--- tests/alsatimer-functions | 29 +++++++++++++ tests/alsatimer-instance-info | 7 +-- tests/alsatimer-instance-params | 7 +-- tests/alsatimer-instance-status | 7 +-- tests/alsatimer-tick-event | 19 +++++++++ tests/alsatimer-tstamp-event | 20 +++++++++ tests/alsatimer-user-instance | 14 +++--- tests/helper.py | 64 +++++++++++++++++++++++----- tests/meson.build | 22 ++++++++++ 60 files changed, 800 insertions(+), 164 deletions(-) create mode 100644 tests/alsactl-elem-info-common create mode 100644 tests/alsactl-elem-info-single-array create mode 100644 tests/alsactl-functions create mode 100644 tests/alsahwdep-device-common create mode 100644 tests/alsahwdep-functions create mode 100644 tests/alsarawmidi-functions create mode 100644 tests/alsaseq-addr create mode 100644 tests/alsaseq-event create mode 100644 tests/alsaseq-event-cntr create mode 100644 tests/alsaseq-event-data-connect create mode 100644 tests/alsaseq-event-data-ctl create mode 100644 tests/alsaseq-event-data-note create mode 100644 tests/alsaseq-event-data-queue create mode 100644 tests/alsaseq-event-data-result create mode 100644 tests/alsaseq-functions create mode 100644 tests/alsaseq-queue-timer-common create mode 100644 tests/alsaseq-remove-filter create mode 100644 tests/alsatimer-functions create mode 100644 tests/alsatimer-tick-event create mode 100644 tests/alsatimer-tstamp-event ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/76 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/76.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 0/2] ALSA: x86: intel_hdmi_audio: pm_runtime updates
by Pierre-Louis Bossart 17 Jun '22

17 Jun '22

While cleaning up the code to use pm_runtime_resume_and_get(), I realized we never called pm_runtime_enable() for this driver. This is potentially very invasive and error-prone, so feedback and test results are welcome - all my BYT/CHT devices are not longer functional unfortunately, and our CI MinnowBoard uses the 'other' HDMI solution based on HDaudio. In addition, the resume was assumed to be always successful, which isn't a very good idea. Pierre-Louis Bossart (2): ALSA: x86: intel_hdmi_audio: enable pm_runtime and set autosuspend delay ALSA: x86: intel_hdmi_audio: use pm_runtime_resume_and_get() sound/x86/intel_hdmi_audio.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) -- 2.34.1

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[PATCH] ALSA: hda: intel-nhlt: remove use of __func__ in dev_dbg
by Pierre-Louis Bossart 17 Jun '22

17 Jun '22

The module and function information can be added with 'modprobe foo dyndbg=+pmf' Suggested-by: Greg KH <gregkh(a)linuxfoundation.org> 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> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/hda/intel-nhlt.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 4063da3782833..9db5ccd9aa2db 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -55,8 +55,8 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) /* find max number of channels based on format_configuration */ if (fmt_configs->fmt_count) { - dev_dbg(dev, "%s: found %d format definitions\n", - __func__, fmt_configs->fmt_count); + dev_dbg(dev, "found %d format definitions\n", + fmt_configs->fmt_count); for (i = 0; i < fmt_configs->fmt_count; i++) { struct wav_fmt_ext *fmt_ext; @@ -66,9 +66,9 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) if (fmt_ext->fmt.channels > max_ch) max_ch = fmt_ext->fmt.channels; } - dev_dbg(dev, "%s: max channels found %d\n", __func__, max_ch); + dev_dbg(dev, "max channels found %d\n", max_ch); } else { - dev_dbg(dev, "%s: No format information found\n", __func__); + dev_dbg(dev, "No format information found\n"); } if (cfg->device_config.config_type != NHLT_CONFIG_TYPE_MIC_ARRAY) { @@ -95,17 +95,16 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) } if (dmic_geo > 0) { - dev_dbg(dev, "%s: Array with %d dmics\n", __func__, dmic_geo); + dev_dbg(dev, "Array with %d dmics\n", dmic_geo); } if (max_ch > dmic_geo) { - dev_dbg(dev, "%s: max channels %d exceed dmic number %d\n", - __func__, max_ch, dmic_geo); + dev_dbg(dev, "max channels %d exceed dmic number %d\n", + max_ch, dmic_geo); } } } - dev_dbg(dev, "%s: dmic number %d max_ch %d\n", - __func__, dmic_geo, max_ch); + dev_dbg(dev, "dmic number %d max_ch %d\n", dmic_geo, max_ch); return dmic_geo; } -- 2.34.1

2 1

[PATCH] ALSA: hda: intel-dspcfg: use SOF for UpExtreme and UpExtreme11 boards
by Pierre-Louis Bossart 17 Jun '22

17 Jun '22

The UpExtreme BIOS reports microphones that are not physically present, so this module ends-up selecting SOF, while the UpExtreme11 BIOS does not report microphones so the snd-hda-intel driver is selected. For consistency use SOF unconditionally in autodetection mode. The use of the snd-hda-intel driver can still be enabled with 'options snd-intel-dspcfg dsp_driver=1' Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index a8fe01764b254..ec9cbb219bc14 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -196,6 +196,12 @@ static const struct config_entry config_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Google"), } }, + { + .ident = "UP-WHL", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + } + }, {} } }, @@ -358,6 +364,12 @@ static const struct config_entry config_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Google"), } }, + { + .ident = "UPX-TGL", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + } + }, {} } }, -- 2.34.1

2 1

[PATCH] firewire: cdev: fix potential leak of kernel stack due to uninitialized value
by Takashi Sakamoto 17 Jun '22

17 Jun '22

Recent change brings potential leak of value on kernel stack to userspace due to uninitialized value. This commit fixes the bug. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Fixes: baa914cd81f ("firewire: add kernel API to access CYCLE_TIME register") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- drivers/firewire/core-cdev.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/firewire/core-cdev.c b/drivers/firewire/core-cdev.c index 8e9670036e5c..032ee56c34a3 100644 --- a/drivers/firewire/core-cdev.c +++ b/drivers/firewire/core-cdev.c @@ -1211,7 +1211,7 @@ static int ioctl_get_cycle_timer2(struct client *client, union ioctl_arg *arg) struct fw_cdev_get_cycle_timer2 *a = &arg->get_cycle_timer2; struct fw_card *card = client->device->card; struct timespec64 ts = {0, 0}; - u32 cycle_time; + u32 cycle_time = 0; int ret = 0; local_irq_disable(); -- 2.34.1

2 4

Regression: Upgrade to 1.2.7 Causes No HDMI Audio Device on AMD Trinity APU
by GitHub issues - opened 17 Jun '22

17 Jun '22

alsa-project/alsa-lib issue #239 was opened from lovenemesis: After upgrading to alsa-lib 1.2.7, HDMI Audio from AMD A10-5800K "Trinity" series APU stopped working. HDMI Audio is no longer presented as a Output Device in GNOME Setting - Audio. aplay -l shows error for HDMI device: `aplay: device_list:304: control digital audio info (0): Input/output error` **Downgrading to alsa-lib to 1.2.6** restore the HDMI audio device on A10-5800K system. Below are some system info I collected when running alsa-lib-1.2.6 and alsa-lib-1.2.7 [alsa-lib-1.2.6.txt](https://github.com/alsa-project/alsa-lib/files/8923675/… [alsa-lib-1.2.7.txt](https://github.com/alsa-project/alsa-lib/files/8923676/… It appears to me that kernel, pipewire, wireplumber versions have no impact on the occurrence of the issue. It is also reported in [RedHat Bugzilla](https://bugzilla.redhat.com/show_bug.cgi?id=2097574) Issue URL : https://github.com/alsa-project/alsa-lib/issues/239 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Re: [PATCH v2] ASoC: ti: omap-mcbsp: duplicate sysfs error
by Mark Brown 16 Jun '22

16 Jun '22

On Wed, Jun 15, 2022 at 10:09:55AM -0500, David Owens wrote: > Convert to managed versions of sysfs and clk allocation to simplify > unbinding and error handling in probe. Managed sysfs node > creation specifically addresses the following error seen the second time > probe is attempted after sdma_pcm_platform_register() previously requsted > probe deferral: This doesn't apply against current code (either for-5.19 or for-5.20), please check and resend.

1 0

Re: [PATCH v2] ASoC: ti: omap-mcbsp: duplicate sysfs error
by Péter Ujfalusi 16 Jun '22

16 Jun '22

Hi David, On 6/15/22 18:09, David Owens wrote: > Convert to managed versions of sysfs and clk allocation to simplify > unbinding and error handling in probe. Managed sysfs node > creation specifically addresses the following error seen the second time > probe is attempted after sdma_pcm_platform_register() previously requsted > probe deferral: > > sysfs: cannot create duplicate filename '/devices/platform/68000000.ocp/49022000.mcbsp/max_tx_thres' I have tried to looked up devm_sysfs_remove_group(), which did not existed, but you found the correct devm ;) Looks good, thank you! Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> > Signed-off-by: David Owens <dowens(a)precisionplanting.com> > --- > sound/soc/ti/omap-mcbsp-priv.h | 2 -- > sound/soc/ti/omap-mcbsp-st.c | 21 ++++----------------- > sound/soc/ti/omap-mcbsp.c | 26 ++++---------------------- > 3 files changed, 8 insertions(+), 41 deletions(-) > > diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h > index 7865cda4bf0a..da519ea1f303 100644 > --- a/sound/soc/ti/omap-mcbsp-priv.h > +++ b/sound/soc/ti/omap-mcbsp-priv.h > @@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, > > /* Sidetone specific API */ > int omap_mcbsp_st_init(struct platform_device *pdev); > -void omap_mcbsp_st_cleanup(struct platform_device *pdev); > - > int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp); > int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp); > > diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c > index 0bc7d26c660a..402a57a502e6 100644 > --- a/sound/soc/ti/omap-mcbsp-st.c > +++ b/sound/soc/ti/omap-mcbsp-st.c > @@ -292,14 +292,11 @@ static ssize_t st_taps_store(struct device *dev, > > static DEVICE_ATTR_RW(st_taps); > > -static const struct attribute *sidetone_attrs[] = { > +static struct attribute *sidetone_attrs[] = { > &dev_attr_st_taps.attr, > NULL, > }; > - > -static const struct attribute_group sidetone_attr_group = { > - .attrs = (struct attribute **)sidetone_attrs, > -}; > +ATTRIBUTE_GROUPS(sidetone); > > int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp) > { > @@ -347,7 +344,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) > if (!st_data) > return -ENOMEM; > > - st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); > + st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick"); > if (IS_ERR(st_data->mcbsp_iclk)) { > dev_warn(mcbsp->dev, > "Failed to get ick, sidetone might be broken\n"); > @@ -359,7 +356,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) > if (!st_data->io_base_st) > return -ENOMEM; > > - ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); > + ret = devm_device_add_group(mcbsp->dev, &sidetone_group); > if (ret) > return ret; > > @@ -368,16 +365,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev) > return 0; > } > > -void omap_mcbsp_st_cleanup(struct platform_device *pdev) > -{ > - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); > - > - if (mcbsp->st_data) { > - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); > - clk_put(mcbsp->st_data->mcbsp_iclk); > - } > -} > - > static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, > struct snd_ctl_elem_info *uinfo) > { > diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c > index 4479d74f0a45..395493a2d965 100644 > --- a/sound/soc/ti/omap-mcbsp.c > +++ b/sound/soc/ti/omap-mcbsp.c > @@ -595,16 +595,13 @@ static ssize_t dma_op_mode_store(struct device *dev, > > static DEVICE_ATTR_RW(dma_op_mode); > > -static const struct attribute *additional_attrs[] = { > +static struct attribute *additional_attrs[] = { > &dev_attr_max_tx_thres.attr, > &dev_attr_max_rx_thres.attr, > &dev_attr_dma_op_mode.attr, > NULL, > }; > - > -static const struct attribute_group additional_attr_group = { > - .attrs = (struct attribute **)additional_attrs, > -}; > +ATTRIBUTE_GROUPS(additional); > > /* > * McBSP1 and McBSP3 are directly mapped on 1610 and 1510. > @@ -702,8 +699,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) > mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; > mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; > > - ret = sysfs_create_group(&mcbsp->dev->kobj, > - &additional_attr_group); > + ret = devm_device_add_group(mcbsp->dev, &additional_group); > if (ret) { > dev_err(mcbsp->dev, > "Unable to create additional controls\n"); > @@ -711,16 +707,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) > } > } > > - ret = omap_mcbsp_st_init(pdev); > - if (ret) > - goto err_st; > - > - return 0; > - > -err_st: > - if (mcbsp->pdata->buffer_size) > - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); > - return ret; > + return omap_mcbsp_st_init(pdev); > } > > /* > @@ -1431,11 +1418,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) > if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req)) > cpu_latency_qos_remove_request(&mcbsp->pm_qos_req); > > - if (mcbsp->pdata->buffer_size) > - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); > - > - omap_mcbsp_st_cleanup(pdev); > - > return 0; > } > > -- > 2.34.1 > > This email is intended solely for the use of the individual to whom it is addressed and may contain confidential and/or privileged material. Any views or opinions presented are solely those of the author and do not necessarily represent those of Precision Planting. If you are not the intended recipient, be advised that you have received this email in error and that any use, dissemination, forwarding, printing, or copying of this email is strictly prohibited. Neither AGCO nor the sender accepts any responsibility for viruses, and it is your responsibility to scan, and virus check the e-mail and its attachment(s) (if any). -- Péter

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[PATCH] ASoC: SOF: Intel: IPC4: enable IMR boot
by Peter Ujfalusi 16 Jun '22

16 Jun '22

IPC4 based firmwares have unconditional support for IMR boot. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-loader.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 9e99f376f2b3..bca9dc5917f4 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -538,7 +538,8 @@ int hda_dsp_post_fw_run(struct snd_sof_dev *sdev) /* Check if IMR boot is usable */ if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT) && - sdev->fw_ready.flags & SOF_IPC_INFO_D3_PERSISTENT) + (sdev->fw_ready.flags & SOF_IPC_INFO_D3_PERSISTENT || + sdev->pdata->ipc_type == SOF_INTEL_IPC4)) hdev->imrboot_supported = true; } -- 2.36.1

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[PATCH] ASoC: SOF: ipc4-topology: Fix error code in sof_ipc4_volume_put()
by Dan Carpenter 16 Jun '22

16 Jun '22

The sof_ipc4_volume_put() function returns type bool so returning -ENOENT means returning true. Return false instead. Fixes: 955e84fc0b6d ("ASoC: SOF: ipc4-topology: Add control IO ops") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/sof/ipc4-control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-control.c b/sound/soc/sof/ipc4-control.c index 95ee121dd3cf..0d5a578c3496 100644 --- a/sound/soc/sof/ipc4-control.c +++ b/sound/soc/sof/ipc4-control.c @@ -142,7 +142,7 @@ static bool sof_ipc4_volume_put(struct snd_sof_control *scontrol, if (!widget_found) { dev_err(scomp->dev, "Failed to find widget for kcontrol %s\n", scontrol->name); - return -ENOENT; + return false; } ret = sof_ipc4_set_volume_data(sdev, swidget, scontrol); -- 2.35.1

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[PATCH] ASoC: Intel: Skylake: remove redundant re-assignments to pointer array
by Colin Ian King 16 Jun '22

16 Jun '22

There are two occurrences where the pointer array is being assigned a value that is never read, the pointer gets updated in the next iteration of a loop. These assignments are redundant and can be removed. Cleans up clang scan-build warnings: sound/soc/intel/skylake/skl-topology.c:2953:3: warning: Value stored to 'array' is never read [deadcode.DeadStores] sound/soc/intel/skylake/skl-topology.c:3602:3: warning: Value stored to 'array' is never read [deadcode.DeadStores] Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/intel/skylake/skl-topology.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 9bdf020a2b64..e06eac592da1 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2950,9 +2950,6 @@ static int skl_tplg_get_pvt_data(struct snd_soc_tplg_dapm_widget *tplg_w, block_size = ret; off += array->size; - array = (struct snd_soc_tplg_vendor_array *) - (tplg_w->priv.data + off); - data = (tplg_w->priv.data + off); if (block_type == SKL_TYPE_TUPLE) { @@ -3599,9 +3596,6 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, block_size = ret; off += array->size; - array = (struct snd_soc_tplg_vendor_array *) - (manifest->priv.data + off); - data = (manifest->priv.data + off); if (block_type == SKL_TYPE_TUPLE) { -- 2.35.3

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[PATCH] ASoC: tegra: Fix clock DAI format on Tegra210
by Sameer Pujar 16 Jun '22

16 Jun '22

I2S reset failures are seen on Tegra210 and later platforms. This indicates absence of I2S bit clock, which is required to perform the reset operation. Following failures are seen with I2S based tests on Tegra210 and later: tegra210-i2s 2901100.i2s: timeout: failed to reset I2S for playback tegra210-i2s 2901100.i2s: ASoC: PRE_PMU: I2S2 RX event failed: -110 tegra210-i2s 2901100.i2s: timeout: failed to reset I2S for capture tegra210-i2s 2901100.i2s: ASoC: PRE_PMU: I2S2 TX event failed: -110 The commit d92ad6633fa7 ("ASoC: tegra: Update to use set_fmt_new callback") regressed I2S functionality on Tegra platforms. Basically it flipped clock provider and consumer DAI formats. This configures Tegra I2S in consumer mode by default now and there is none to provide bit clock during loopback tests. The external codec based tests also fail because both Tegra I2S and codec I2S get configured in consumer mode. ASoC core flips the DAI format before calling set_fmt() for CPU DAIs. This is negated in above commit. Fix this by swapping SND_SOC_DAIFMT_BC_FC and SND_SOC_DAIFMT_BP_FP switch cases. Fixes: d92ad6633fa7 ("ASoC: tegra: Update to use set_fmt_new callback") Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Cc: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/tegra/tegra210_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index a289458..01c76ba 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -215,10 +215,10 @@ static int tegra210_i2s_set_fmt(struct snd_soc_dai *dai, mask = I2S_CTRL_MASTER_EN_MASK; switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_BC_FC: val = 0; break; - case SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_BP_FP: val = I2S_CTRL_MASTER_EN; break; default: -- 2.7.4

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[linux-next:master] BUILD REGRESSION 8cb8311e95e3bb58bd84d6350365f14a718faa6d
by kernel test robot 16 Jun '22

16 Jun '22

tree/branch: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master branch HEAD: 8cb8311e95e3bb58bd84d6350365f14a718faa6d Add linux-next specific files for 20220525 Error/Warning reports: https://lore.kernel.org/linux-mm/202204291924.vTGZmerI-lkp@intel.com https://lore.kernel.org/linux-mm/202205031017.4TwMan3l-lkp@intel.com https://lore.kernel.org/linux-mm/202205041248.WgCwPcEV-lkp@intel.com https://lore.kernel.org/linux-mm/202205150051.3RzuooAG-lkp@intel.com https://lore.kernel.org/linux-mm/202205150117.sd6HzBVm-lkp@intel.com https://lore.kernel.org/lkml/202205100617.5UUm3Uet-lkp@intel.com https://lore.kernel.org/llvm/202205251645.gusu3spL-lkp@intel.com Error/Warning: (recently discovered and may have been fixed) drivers/gpu/drm/amd/amdgpu/amdgpu_discovery.c:1364:5: warning: no previous prototype for 'amdgpu_discovery_get_mall_info' [-Wmissing-prototypes] drivers/gpu/drm/amd/amdgpu/soc21.c:171:6: warning: no previous prototype for 'soc21_grbm_select' [-Wmissing-prototypes] drivers/gpu/drm/solomon/ssd130x-spi.c:154:35: warning: 'ssd130x_spi_table' defined but not used [-Wunused-const-variable=] drivers/net/wireless/intel/iwlwifi/pcie/trans.c:1093:9: warning: 'CAUSE' macro redefined [-Wmacro-redefined] drivers/video/fbdev/omap/hwa742.c:492:5: warning: no previous prototype for 'hwa742_update_window_async' [-Wmissing-prototypes] fs/buffer.c:2254:5: warning: stack frame size (2144) exceeds limit (1024) in 'block_read_full_folio' [-Wframe-larger-than] fs/ntfs/aops.c:378:12: warning: stack frame size (2216) exceeds limit (1024) in 'ntfs_read_folio' [-Wframe-larger-than] Unverified Error/Warning (likely false positive, please contact us if interested): .__mulsi3.o.cmd: No such file or directory Makefile:686: arch/h8300/Makefile: No such file or directory Makefile:765: arch/h8300/Makefile: No such file or directory arch/Kconfig:10: can't open file "arch/h8300/Kconfig" arch/riscv/purgatory/kexec-purgatory.c:1860:9: sparse: sparse: trying to concatenate 29720-character string (8191 bytes max) drivers/gpu/drm/bridge/adv7511/adv7511.h:229:17: warning: 'ADV7511_REG_CEC_RX_FRAME_HDR' defined but not used [-Wunused-const-variable=] drivers/gpu/drm/bridge/adv7511/adv7511.h:235:17: warning: 'ADV7511_REG_CEC_RX_FRAME_LEN' defined but not used [-Wunused-const-variable=] drivers/infiniband/hw/hns/hns_roce_hw_v2.c:309:9: sparse: sparse: dubious: x & !y drivers/pinctrl/meson/pinctrl-meson8-pmx.c:60:25: warning: Value stored to 'func' during its initialization is never read [clang-analyzer-deadcode.DeadStores] drivers/staging/vt6655/card.c:758:16: sparse: sparse: cast to restricted __le64 drivers/vhost/vdpa.c:595 vhost_vdpa_unlocked_ioctl() warn: maybe return -EFAULT instead of the bytes remaining? kernel/bpf/helpers.c:1468:29: sparse: sparse: symbol 'bpf_dynptr_from_mem_proto' was not declared. Should it be static? kernel/bpf/helpers.c:1490:29: sparse: sparse: symbol 'bpf_dynptr_from_mem_proto' was not declared. Should it be static? kernel/bpf/helpers.c:1516:29: sparse: sparse: symbol 'bpf_dynptr_read_proto' was not declared. Should it be static? kernel/bpf/helpers.c:1542:29: sparse: sparse: symbol 'bpf_dynptr_write_proto' was not declared. Should it be static? kernel/bpf/helpers.c:1569:29: sparse: sparse: symbol 'bpf_dynptr_data_proto' was not declared. Should it be static? make[1]: *** No rule to make target 'arch/h8300/Makefile'. mm/shmem.c:1948 shmem_getpage_gfp() warn: should '(((1) << 12) / 512) << folio_order(folio)' be a 64 bit type? sound/soc/intel/avs/ipc.c:87:5-24: atomic_dec_and_test variation before object free at line 88. {standard input}:3488: Error: unknown pseudo-op: `.l28' Error/Warning ids grouped by kconfigs: gcc_recent_errors |-- alpha-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- alpha-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | `-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 |-- arc-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | `-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 |-- arm-allmodconfig | |-- arch-arm-mach-omap2-dma.c:Unneeded-variable:errata-Return-on-line | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | |-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 | |-- drivers-video-fbdev-omap-hwa742.c:warning:no-previous-prototype-for-hwa742_update_window_async | `-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_from_mem_proto-was-not-declared.-Should-it-be-static |-- arm-allyesconfig | |-- arch-arm-mach-omap2-dma.c:Unneeded-variable:errata-Return-on-line | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | |-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 | |-- drivers-video-fbdev-omap-hwa742.c:warning:no-previous-prototype-for-hwa742_update_window_async | `-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_from_mem_proto-was-not-declared.-Should-it-be-static |-- arm64-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- arm64-allyesconfig | |-- arch-arm64-kernel-signal.c:sparse:sparse:dereference-of-noderef-expression | |-- arch-arm64-kernel-signal.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-struct-user_ctxs-noderef-__user-user-got-struct-user_ctxs | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-infiniband-hw-hns-hns_roce_hw_v2.c:sparse:sparse:dubious:x-y | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | |-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_from_mem_proto-was-not-declared.-Should-it-be-static | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_read_proto-was-not-declared.-Should-it-be-static | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_write_proto-was-not-declared.-Should-it-be-static | `-- kernel-stackleak.c:sparse:sparse:symbol-stackleak_erase_off_task_stack-was-not-declared.-Should-it-be-static |-- csky-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- csky-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- csky-randconfig-s032-20220524 | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_data_proto-was-not-declared.-Should-it-be-static | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_from_mem_proto-was-not-declared.-Should-it-be-static | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_read_proto-was-not-declared.-Should-it-be-static | `-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_write_proto-was-not-declared.-Should-it-be-static |-- h8300-allmodconfig | |-- Makefile:arch-h8300-Makefile:No-such-file-or-directory | |-- arch-Kconfig:can-t-open-file-arch-h8300-Kconfig | `-- make:No-rule-to-make-target-arch-h8300-Makefile-. |-- h8300-allyesconfig | |-- Makefile:arch-h8300-Makefile:No-such-file-or-directory | |-- arch-Kconfig:can-t-open-file-arch-h8300-Kconfig | `-- make:No-rule-to-make-target-arch-h8300-Makefile-. |-- h8300-buildonly-randconfig-r004-20220524 | |-- Makefile:arch-h8300-Makefile:No-such-file-or-directory | |-- arch-Kconfig:can-t-open-file-arch-h8300-Kconfig | `-- make:No-rule-to-make-target-arch-h8300-Makefile-. |-- h8300-randconfig-r033-20220524 | |-- Makefile:arch-h8300-Makefile:No-such-file-or-directory | |-- arch-Kconfig:can-t-open-file-arch-h8300-Kconfig | `-- make:No-rule-to-make-target-arch-h8300-Makefile-. |-- i386-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_HDR-defined-but-not-used | |-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_LEN-defined-but-not-used | `-- drivers-gpu-drm-solomon-ssd13-spi.c:warning:ssd13_spi_table-defined-but-not-used |-- i386-randconfig-a012 | |-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_HDR-defined-but-not-used | `-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_LEN-defined-but-not-used |-- i386-randconfig-a014 | |-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_HDR-defined-but-not-used | `-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_LEN-defined-but-not-used |-- i386-randconfig-m021 | `-- mm-shmem.c-shmem_getpage_gfp()-warn:should-((()-)-)-folio_order(folio)-be-a-bit-type |-- ia64-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- ia64-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- ia64-randconfig-r036-20220524 | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- m68k-allyesconfig | |-- drivers-block-paride-bpck.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-comm.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-dstr.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-epat.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-epia.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-friq.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-frpw.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-kbic.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-on26.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-block-paride-ppc6lnx.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-comedi-drivers-aio_aio12_8.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-comedi-drivers-das16m1.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-comedi-drivers-ni_at_ao.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-comedi-drivers-ni_daq_700.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-net-ethernet-apne.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-net-ethernet-xircom-xirc2ps_cs.c:sparse:sparse:cast-to-restricted-__le16 | |-- drivers-tty-ipwireless-hardware.c:sparse:sparse:incorrect-type-in-initializer-(different-base-types)-expected-restricted-__le16-usertype-raw_data-got-int | `-- drivers-tty-ipwireless-hardware.c:sparse:sparse:incorrect-type-in-initializer-(different-base-types)-expected-unsigned-short-unused-usertype-__v-got-restricted-__le16-assigned-usertype-raw_data |-- microblaze-randconfig-m031-20220524 | `-- drivers-vhost-vdpa.c-vhost_vdpa_unlocked_ioctl()-warn:maybe-return-EFAULT-instead-of-the-bytes-remaining |-- mips-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | `-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 |-- mips-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | |-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 | `-- sound-soc-intel-avs-ipc.c:atomic_dec_and_test-variation-before-object-free-at-line-. |-- openrisc-randconfig-s032-20220524 | `-- __mulsi3.o.cmd:No-such-file-or-directory |-- parisc-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- parisc-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- powerpc-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | `-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 |-- powerpc-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | `-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 |-- riscv-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- riscv-allyesconfig | |-- arch-riscv-kernel-machine_kexec.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-void-const-got-void-noderef-__user-buf | |-- arch-riscv-purgatory-kexec-purgatory.c:sparse:sparse:trying-to-concatenate-character-string-(-bytes-max) | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-infiniband-hw-hns-hns_roce_hw_v2.c:sparse:sparse:dubious:x-y | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | |-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 | |-- kernel-fork.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-struct-atomic_t-usertype-lock-got-struct-atomic_t-noderef-__rcu | `-- kernel-seccomp.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-struct-atomic_t-usertype-lock-got-struct-atomic_t-noderef-__rcu |-- riscv-randconfig-r042-20220524 | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- s390-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-infiniband-hw-hns-hns_roce_hw_v2.c:sparse:sparse:dubious:x-y | `-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int |-- s390-randconfig-r014-20220524 | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select |-- sh-buildonly-randconfig-r003-20220524 | `-- standard-input:Error:unknown-pseudo-op:l28 |-- sparc-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-infiniband-hw-hns-hns_roce_hw_v2.c:sparse:sparse:dubious:x-y | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | |-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_from_mem_proto-was-not-declared.-Should-it-be-static | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_read_proto-was-not-declared.-Should-it-be-static | `-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_write_proto-was-not-declared.-Should-it-be-static |-- sparc-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-infiniband-hw-hns-hns_roce_hw_v2.c:sparse:sparse:dubious:x-y | |-- drivers-pci-pci.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-pci_power_t-assigned-usertype-state-got-int | |-- drivers-staging-vt6655-card.c:sparse:sparse:cast-to-restricted-__le64 | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_from_mem_proto-was-not-declared.-Should-it-be-static | |-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_read_proto-was-not-declared.-Should-it-be-static | `-- kernel-bpf-helpers.c:sparse:sparse:symbol-bpf_dynptr_write_proto-was-not-declared.-Should-it-be-static |-- x86_64-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | |-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select | |-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_HDR-defined-but-not-used | |-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_LEN-defined-but-not-used | `-- drivers-gpu-drm-solomon-ssd13-spi.c:warning:ssd13_spi_table-defined-but-not-used |-- x86_64-randconfig-a011 | |-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_HDR-defined-but-not-used | `-- drivers-gpu-drm-bridge-adv7511-adv7511.h:warning:ADV7511_REG_CEC_RX_FRAME_LEN-defined-but-not-used |-- xtensa-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info | `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select `-- xtensa-allyesconfig |-- drivers-gpu-drm-amd-amdgpu-amdgpu_discovery.c:warning:no-previous-prototype-for-amdgpu_discovery_get_mall_info `-- drivers-gpu-drm-amd-amdgpu-soc21.c:warning:no-previous-prototype-for-soc21_grbm_select clang_recent_errors |-- arm-randconfig-c002-20220524 | `-- drivers-pinctrl-meson-pinctrl-meson8-pmx.c:warning:Value-stored-to-func-during-its-initialization-is-never-read-clang-analyzer-deadcode.DeadStores |-- hexagon-randconfig-r011-20220524 | `-- fs-buffer.c:warning:stack-frame-size-()-exceeds-limit-()-in-block_read_full_folio |-- hexagon-randconfig-r035-20220524 | |-- fs-buffer.c:warning:stack-frame-size-()-exceeds-limit-()-in-block_read_full_folio | `-- fs-ntfs-aops.c:warning:stack-frame-size-()-exceeds-limit-()-in-ntfs_read_folio `-- mips-randconfig-r022-20220524 `-- drivers-net-wireless-intel-iwlwifi-pcie-trans.c:warning:CAUSE-macro-redefined elapsed time: 858m configs tested: 94 configs skipped: 3 gcc tested configs: arm allmodconfig arm allyesconfig arm64 allyesconfig arm defconfig arm64 defconfig mips allyesconfig riscv allyesconfig um x86_64_defconfig riscv allmodconfig um i386_defconfig mips allmodconfig powerpc allmodconfig s390 allmodconfig m68k allmodconfig powerpc allyesconfig s390 allyesconfig m68k allyesconfig sparc allyesconfig parisc allyesconfig sh allmodconfig h8300 allyesconfig arc allyesconfig alpha allyesconfig nios2 allyesconfig m68k mac_defconfig arc nsimosci_hs_smp_defconfig sh ul2_defconfig mips maltaup_xpa_defconfig xtensa iss_defconfig ia64 defconfig ia64 allmodconfig ia64 allyesconfig m68k defconfig nios2 defconfig alpha defconfig csky defconfig xtensa allyesconfig arc defconfig parisc defconfig parisc64 defconfig s390 defconfig i386 debian-10.3-kselftests i386 debian-10.3 i386 defconfig i386 allyesconfig sparc defconfig powerpc allnoconfig i386 randconfig-a001 i386 randconfig-a003 i386 randconfig-a005 x86_64 randconfig-a013 x86_64 randconfig-a011 x86_64 randconfig-a015 i386 randconfig-a014 i386 randconfig-a012 riscv randconfig-r042-20220524 arc randconfig-r043-20220524 s390 randconfig-r044-20220524 x86_64 randconfig-a004 x86_64 randconfig-a002 x86_64 randconfig-a006 riscv allnoconfig riscv nommu_k210_defconfig riscv rv32_defconfig riscv nommu_virt_defconfig riscv defconfig x86_64 rhel-8.3-func x86_64 rhel-8.3-kselftests x86_64 rhel-8.3-kunit x86_64 rhel-8.3-syz x86_64 defconfig x86_64 kexec x86_64 rhel-8.3 x86_64 allyesconfig clang tested configs: mips bcm63xx_defconfig mips lemote2f_defconfig mips tb0287_defconfig arm ep93xx_defconfig arm am200epdkit_defconfig powerpc tqm5200_defconfig i386 randconfig-a002 i386 randconfig-a006 i386 randconfig-a004 x86_64 randconfig-a012 x86_64 randconfig-a014 x86_64 randconfig-a016 i386 randconfig-a013 i386 randconfig-a011 i386 randconfig-a015 hexagon randconfig-r045-20220524 hexagon randconfig-r041-20220524 x86_64 randconfig-a005 x86_64 randconfig-a001 x86_64 randconfig-a003 -- 0-DAY CI Kernel Test Service https://01.org/lkp

7 10

[PATCH] ASoC: SOF: mediatek: Fix error code in probe
by Dan Carpenter 15 Jun '22

15 Jun '22

This should return PTR_ERR() instead of IS_ERR(). Fixes: e0100bfd383c ("ASoC: SOF: mediatek: Add mt8186 ipc support") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/sof/mediatek/mt8186/mt8186.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index 3333a0634e29..e006532caf2f 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -392,7 +392,7 @@ static int mt8186_dsp_probe(struct snd_sof_dev *sdev) PLATFORM_DEVID_NONE, pdev, sizeof(*pdev)); if (IS_ERR(priv->ipc_dev)) { - ret = IS_ERR(priv->ipc_dev); + ret = PTR_ERR(priv->ipc_dev); dev_err(sdev->dev, "failed to create mtk-adsp-ipc device\n"); goto err_adsp_off; } -- 2.35.1

5 4

USB-Audio: no Audient iD4 profile for USB ID 2708:0003
by GitHub issues - opened 15 Jun '22

15 Jun '22

alsa-project/alsa-ucm-conf issue #170 was opened from tiran: PR https://github.com/alsa-project/alsa-ucm-conf/pull/151 added a profile for Audient iD4 audio interface with USB ID ``2708:0009``. I own a version of the device with ``2708:0003``. The configuration for ``2708:0009`` does not work for version ``2708:0003``. I hacked the config file from ``alsa-ucm-1.2.7-1.fc36.noarch`` package and ``systemctl --user restart pipewire pipewire-pulse``. After restart the audio interface no longer shows up in KDE. ``pw-dump`` shows only a disabled/off profile for the device. @robertgzr @perexg Also see https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1119 [alsactl.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8910772/al… [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8910773/lsus… [pw-dump.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8910774/pw… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/170 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/2] ASoC: SOF: Intel: Add support for MeteorLake
by Bard Liao 15 Jun '22

15 Jun '22

Add platform abstraction for the Meteor Lake platform. This platform has significant differences compared to the TGL/ADL generation: it relies on new hardware using the code name 'ACE' and only supports the INTEL_IPC4 protocol and firmware architecture based on the Zephyr RTOS Bard Liao (2): ASoC: SOC: Intel: introduce cl_init callback ASoC: SOF: Intel: Add support for MeteorLake (MTL) include/linux/soundwire/sdw_intel.h | 2 + sound/soc/sof/intel/Kconfig | 16 + sound/soc/sof/intel/Makefile | 4 +- sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 2 + sound/soc/sof/intel/hda-loader.c | 15 +- sound/soc/sof/intel/hda.h | 4 + sound/soc/sof/intel/icl.c | 1 + sound/soc/sof/intel/mtl.c | 800 ++++++++++++++++++++++++++++ sound/soc/sof/intel/mtl.h | 76 +++ sound/soc/sof/intel/pci-mtl.c | 71 +++ sound/soc/sof/intel/shim.h | 2 + sound/soc/sof/intel/tgl.c | 4 + 13 files changed, 994 insertions(+), 4 deletions(-) create mode 100644 sound/soc/sof/intel/mtl.c create mode 100644 sound/soc/sof/intel/mtl.h create mode 100644 sound/soc/sof/intel/pci-mtl.c -- 2.17.1

2 3

[PATCH] ASoC: tegra: Fix MBDRC bypass mode check
by Sameer Pujar 15 Jun '22

15 Jun '22

MBDRC supports different modes of operation. There is no configuration required for bypass mode. The hw_params() call does not filter bypass mode correctly and it leads to following Smatch static checker warning: sound/soc/tegra/tegra210_mbdrc.c:778 tegra210_mbdrc_hw_params() warn: bitwise AND condition is false here Fix this condition by using proper mode mask and just return for bypass mode. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Fixes: 7358a803c778 ("ASoC: tegra: Add Tegra210 based OPE driver") Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/soc/tegra/tegra210_mbdrc.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra210_mbdrc.c b/sound/soc/tegra/tegra210_mbdrc.c index 7d9da33..d786daa 100644 --- a/sound/soc/tegra/tegra210_mbdrc.c +++ b/sound/soc/tegra/tegra210_mbdrc.c @@ -775,7 +775,9 @@ int tegra210_mbdrc_hw_params(struct snd_soc_component *cmpnt) regmap_read(ope->mbdrc_regmap, TEGRA210_MBDRC_CFG, &val); - if (val & TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS) + val &= TEGRA210_MBDRC_CFG_MBDRC_MODE_MASK; + + if (val == TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS) return 0; for (i = 0; i < MBDRC_NUM_BAND; i++) { -- 2.7.4

2 1

[PATCH v2 0/2] Add pinctrl support adsp based platforms
by Srinivasa Rao Mandadapu 15 Jun '22

15 Jun '22

This patch set is to add pinctrl support adsp enabled sc7280 platforms. Changes Since V1: -- Update commit message. Srinivasa Rao Mandadapu (2): dt-bindings: pinctrl: qcom: sc7280: Add compatible string for adsp based platforms pinctrl: qcom: sc7280: Add lpi pinctrl variant data for adsp based targets .../bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 4 +++- drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 14 ++++++++++++++ 2 files changed, 17 insertions(+), 1 deletion(-) -- 2.7.4

3 8

[v1] ASoC: rockchip: i2s: switch BCLK to GPIO
by Judy Hsiao 15 Jun '22

15 Jun '22

We discoverd that the state of BCLK on, LRCLK off and SD_MODE on may cause the speaker melting issue. Removing LRCLK while BCLK is present can cause unexpected output behavior including a large DC output voltage as described in the Max98357a datasheet. In order to: 1. prevent BCLK from turning on by other component. 2. keep BCLK and LRCLK being present at the same time This patch switches BCLK to GPIO func before LRCLK output, and configures BCLK func back during LRCLK is output. Without this fix, BCLK is turned on 11 ms earlier than LRCK by the da7219. With this fix, BCLK is turned on only 0.4 ms earlier than LRCK by the rockchip codec. Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- sound/soc/rockchip/rockchip_i2s.c | 162 ++++++++++++++++++++++++------ 1 file changed, 130 insertions(+), 32 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 47a3971a9ce1..cecaadd54b48 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -13,6 +13,7 @@ #include <linux/of_gpio.h> #include <linux/of_device.h> #include <linux/clk.h> +#include <linux/pinctrl/consumer.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/spinlock.h> @@ -54,8 +55,40 @@ struct rk_i2s_dev { const struct rk_i2s_pins *pins; unsigned int bclk_ratio; spinlock_t lock; /* tx/rx lock */ + struct pinctrl *pinctrl; + struct pinctrl_state *bclk_on; + struct pinctrl_state *bclk_off; }; +static int i2s_pinctrl_select_bclk_on(struct rk_i2s_dev *i2s) +{ + int ret = 0; + + if (!IS_ERR(i2s->pinctrl) && !IS_ERR_OR_NULL(i2s->bclk_on)) + ret = pinctrl_select_state(i2s->pinctrl, + i2s->bclk_on); + + if (ret) + dev_err(i2s->dev, "bclk enable failed %d\n", ret); + + return ret; +} + +static int i2s_pinctrl_select_bclk_off(struct rk_i2s_dev *i2s) +{ + + int ret = 0; + + if (!IS_ERR(i2s->pinctrl) && !IS_ERR_OR_NULL(i2s->bclk_off)) + ret = pinctrl_select_state(i2s->pinctrl, + i2s->bclk_off); + + if (ret) + dev_err(i2s->dev, "bclk disable failed %d\n", ret); + + return ret; +} + static int i2s_runtime_suspend(struct device *dev) { struct rk_i2s_dev *i2s = dev_get_drvdata(dev); @@ -92,38 +125,49 @@ static inline struct rk_i2s_dev *to_info(struct snd_soc_dai *dai) return snd_soc_dai_get_drvdata(dai); } -static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) +static int rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) { unsigned int val = 0; int retry = 10; + int ret = 0; spin_lock(&i2s->lock); if (on) { - regmap_update_bits(i2s->regmap, I2S_DMACR, - I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); + if (ret < 0) + goto end; - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | I2S_XFER_RXS_START, - I2S_XFER_TXS_START | I2S_XFER_RXS_START); + ret = regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); + if (ret < 0) + goto end; i2s->tx_start = true; } else { i2s->tx_start = false; - regmap_update_bits(i2s->regmap, I2S_DMACR, - I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); + if (ret < 0) + goto end; if (!i2s->rx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | - I2S_XFER_RXS_START, - I2S_XFER_TXS_STOP | - I2S_XFER_RXS_STOP); + ret = regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | + I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | + I2S_XFER_RXS_STOP); + if (ret < 0) + goto end; udelay(150); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_RXC, - I2S_CLR_TXC | I2S_CLR_RXC); + ret = regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); + if (ret < 0) + goto end; regmap_read(i2s->regmap, I2S_CLR, &val); @@ -138,44 +182,57 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) } } } - spin_unlock(&i2s->lock); +end: + spin_unlock(&i2s->lock); + if (ret < 0) + dev_err(i2s->dev, "lrclk update failed\n"); + + return ret; } -static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) +static int rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) { unsigned int val = 0; int retry = 10; + int ret = 0; spin_lock(&i2s->lock); if (on) { - regmap_update_bits(i2s->regmap, I2S_DMACR, + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); + if (ret < 0) + goto end; - regmap_update_bits(i2s->regmap, I2S_XFER, + ret = regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START | I2S_XFER_RXS_START, I2S_XFER_TXS_START | I2S_XFER_RXS_START); + if (ret < 0) + goto end; i2s->rx_start = true; } else { i2s->rx_start = false; - regmap_update_bits(i2s->regmap, I2S_DMACR, + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); + if (ret < 0) + goto end; if (!i2s->tx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, + ret = regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START | I2S_XFER_RXS_START, I2S_XFER_TXS_STOP | I2S_XFER_RXS_STOP); - + if (ret < 0) + goto end; udelay(150); - regmap_update_bits(i2s->regmap, I2S_CLR, + ret = regmap_update_bits(i2s->regmap, I2S_CLR, I2S_CLR_TXC | I2S_CLR_RXC, I2S_CLR_TXC | I2S_CLR_RXC); - + if (ret < 0) + goto end; regmap_read(i2s->regmap, I2S_CLR, &val); - /* Should wait for clear operation to finish */ while (val) { regmap_read(i2s->regmap, I2S_CLR, &val); @@ -187,7 +244,12 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) } } } +end: spin_unlock(&i2s->lock); + if (ret < 0) + dev_err(i2s->dev, "lrclk update failed\n"); + + return ret; } static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, @@ -425,17 +487,26 @@ static int rockchip_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - rockchip_snd_rxctrl(i2s, 1); + ret = rockchip_snd_rxctrl(i2s, 1); else - rockchip_snd_txctrl(i2s, 1); + ret = rockchip_snd_txctrl(i2s, 1); + /* Do not turn on bclk if lrclk open fails. */ + if (ret < 0) + return ret; + i2s_pinctrl_select_bclk_on(i2s); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - rockchip_snd_rxctrl(i2s, 0); - else - rockchip_snd_txctrl(i2s, 0); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (!i2s->tx_start) + i2s_pinctrl_select_bclk_off(i2s); + ret = rockchip_snd_rxctrl(i2s, 0); + } else { + if (!i2s->rx_start) + i2s_pinctrl_select_bclk_off(i2s); + ret = rockchip_snd_txctrl(i2s, 0); + } break; default: ret = -EINVAL; @@ -736,6 +807,33 @@ static int rockchip_i2s_probe(struct platform_device *pdev) } i2s->bclk_ratio = 64; + i2s->pinctrl = devm_pinctrl_get(&pdev->dev); + if (IS_ERR(i2s->pinctrl)) + dev_err(&pdev->dev, "failed to find i2s pinctrl\n"); + + i2s->bclk_on = pinctrl_lookup_state(i2s->pinctrl, + "bclk_on"); + if (IS_ERR_OR_NULL(i2s->bclk_on)) + dev_err(&pdev->dev, "failed to find i2s default state\n"); + else + dev_dbg(&pdev->dev, "find i2s bclk state\n"); + + i2s->bclk_off = pinctrl_lookup_state(i2s->pinctrl, + "bclk_off"); + if (IS_ERR_OR_NULL(i2s->bclk_off)) + dev_err(&pdev->dev, "failed to find i2s gpio state\n"); + else + dev_dbg(&pdev->dev, "find i2s bclk_off state\n"); + + i2s_pinctrl_select_bclk_off(i2s); + + i2s->playback_dma_data.addr = res->start + I2S_TXDR; + i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->playback_dma_data.maxburst = 4; + + i2s->capture_dma_data.addr = res->start + I2S_RXDR; + i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->capture_dma_data.maxburst = 4; dev_set_drvdata(&pdev->dev, i2s); -- 2.36.1.476.g0c4daa206d-goog

2 1

ACPI REV override for Dell XPS 13 9343
by Cezary Rojewski 15 Jun '22

15 Jun '22

Hello, It's been a while since catpt-driver [1] has been introduced to provide full support for Broadwell (BDW) machines with Intel DSP. For BDW, audio devices can make use of DSP only in I2S mode. In 2015 Rafael and Dominik provided quirk [2] for Dell XPS 13 9343. Given the description: _For example, based on what ACPI exports as the supported revision, Dell XPS 13 (2015) configures its audio device to either work in HDA mode or in I2S mode, where the former is supposed to be used on Linux until the latter is fully supported (in the kernel as well as in user space)._ It's clear that such configuration was not fully supported back then. I believe now it is. Perhaps it is time to let the quirk in mention go? By that I mean just the relevant entry, not the ACPI_REV_OVERRIDE_POSSIBLE functionality as a whole. [1]: https://lore.kernel.org/all/20200929141247.8058-1-cezary.rojewski@intel.com/ [2]: https://lore.kernel.org/alsa-devel/5115175.tbCl1JQVaV@vostro.rjw.lan/ Regards, Czarek

3 5

[bug report] ASoC: tegra: Add Tegra210 based OPE driver
by Dan Carpenter 15 Jun '22

15 Jun '22

Hello Sameer Pujar, The patch 7358a803c778: "ASoC: tegra: Add Tegra210 based OPE driver" from Jun 3, 2022, leads to the following Smatch static checker warning: sound/soc/tegra/tegra210_mbdrc.c:778 tegra210_mbdrc_hw_params() warn: bitwise AND condition is false here sound/soc/tegra/tegra210_mbdrc.c 769 int tegra210_mbdrc_hw_params(struct snd_soc_component *cmpnt) 770 { 771 struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); 772 const struct tegra210_mbdrc_config *conf = &mbdrc_init_config; 773 u32 val = 0; 774 unsigned int i; 775 776 regmap_read(ope->mbdrc_regmap, TEGRA210_MBDRC_CFG, &val); 777 --> 778 if (val & TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS is zero so this can't be true. #define TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS (0 << TEGRA210_MBDRC_CFG_MBDRC_MODE_SHIFT) The common false positive with this warning is that the macro is something which depends on the .config and in that case I just add it to the list in smatch_data/kernel.unconstant_macros. But in this case the macro is just always zero... Is there a plan to make it configurable or something? 779 return 0; 780 781 for (i = 0; i < MBDRC_NUM_BAND; i++) { 782 const struct tegra210_mbdrc_band_params *params = 783 &conf->band_params[i]; 784 785 u32 reg_off = i * TEGRA210_MBDRC_FILTER_PARAM_STRIDE; 786 787 tegra210_mbdrc_write_ram(ope->mbdrc_regmap, 788 reg_off + TEGRA210_MBDRC_CFG_RAM_CTRL, 789 reg_off + TEGRA210_MBDRC_CFG_RAM_DATA, 790 0, (u32 *)&params->biquad_params[0], 791 TEGRA210_MBDRC_MAX_BIQUAD_STAGES * 5); 792 } 793 return 0; 794 } regards, dan carpenter

2 3

pcm: Add error handler for `fgets`
by GitHub pull_request - opened 15 Jun '22

15 Jun '22

alsa-project/alsa-lib pull request #238 was opened from Korilakkuma: `fgets` function returns `NULL` if error occured. Therefore, end loop in the case. Request URL : https://github.com/alsa-project/alsa-lib/pull/238 Patch URL : https://github.com/alsa-project/alsa-lib/pull/238.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Any way to fuse static libasound with alsa-plugins so no dlopens are performed?
by GitHub issues - opened 14 Jun '22

14 Jun '22

alsa-project/alsa-plugins issue #45 was opened from MIvanchev: See title. I have a project that might benefit from such a mix. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/45 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ASoC: cs35l45: Add endianness flag in snd_soc_component_driver
by Charles Keepax 14 Jun '22

14 Jun '22

The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. Fixes: 0d463d016000 ("ASoC: cs35l45: Add driver for Cirrus Logic CS35L45 Smart Amp") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l45.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index c94edfce4b720..d15b3b77c7eb0 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -500,6 +500,8 @@ static const struct snd_soc_component_driver cs35l45_component = { .num_controls = ARRAY_SIZE(cs35l45_controls), .name = "cs35l45", + + .endianness = 1, }; static int __maybe_unused cs35l45_runtime_suspend(struct device *dev) -- 2.30.2

2 1

[PATCH] ASoC: ops: Fix boolean/integer detection for simple controls
by Mark Brown 14 Jun '22

14 Jun '22

The standard snd_soc_info_volsw() detects if a control is a volume control and needs to be reported as an integer even if it only has two values by looking for the string " Volume" in the control name. This results in false positives if the control has a name like "HP Volume Ramp Switch" since any " Volume" is matched, not just a trailing one. Fix this by making sure that we only match at the end of the control name. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index e11109dcaa50..0267e39de2a8 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -176,13 +176,21 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + const char *vol_string = NULL; int max; max = uinfo->value.integer.max = mc->max - mc->min; if (mc->platform_max && mc->platform_max < max) max = mc->platform_max; - if (max == 1 && !strstr(kcontrol->id.name, " Volume")) + /* Even two value controls ending in Volume should always be integer */ + if (max == 1) { + vol_string = strstr(kcontrol->id.name, " Volume"); + if (vol_string && strcmp(vol_string, " Volume")) + vol_string = NULL; + } + + if (!vol_string) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; -- 2.30.2

1 1

Re: [PATCH] ASoC: ti: omap-mcbsp: duplicate sysfs failure after PROBE_DEFER
by Mark Brown 14 Jun '22

14 Jun '22

On Tue, Jun 14, 2022 at 10:59:31AM -0500, David Owens wrote: > The call to sdma_pcm_platform_register() can return PROBE_DEFER, leading > to omap_mcbsp_init() being called multiple times. sysfs node creation > fails in subsequent calls to omap_mcbsp_init(), which prevents > the driver from ever successfully probing. The resulting errors can be > seen during boot: > > [ 1.749328] sysfs: cannot create duplicate filename '/devices/platform/68000000.ocp/49022000.mcbsp/max_tx_thres' > [ 1.759643] CPU: 0 PID: 6 Comm: kworker/u2:0 Not tainted 5.18.0-yocto-standard #1 > [ 1.767181] Hardware name: Generic OMAP36xx (Flattened Device Tree) > [ 1.773498] Workqueue: events_unbound deferred_probe_work_func > [ 1.779449] unwind_backtrace from show_stack+0x10/0x14 > [ 1.784729] show_stack from sysfs_warn_dup+0x4c/0x60 Please think hard before including complete backtraces in upstream reports, they are very large and contain almost no useful information relative to their size so often obscure the relevant content in your message. If part of the backtrace is usefully illustrative (it often is for search engines if nothing else) then it's usually better to pull out the relevant sections. > +++ b/sound/soc/ti/omap-mcbsp.c > @@ -1403,6 +1403,10 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) > mcbsp->dev = &pdev->dev; > platform_set_drvdata(pdev, mcbsp); > > + ret = sdma_pcm_platform_register(&pdev->dev, "tx", "rx"); > + if (ret) > + return ret; > + > ret = omap_mcbsp_init(pdev); > if (ret) > return ret; > @@ -1412,13 +1416,9 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) > omap_mcbsp_dai.capture.formats = SNDRV_PCM_FMTBIT_S16_LE; > } > > - ret = devm_snd_soc_register_component(&pdev->dev, > + return devm_snd_soc_register_component(&pdev->dev, > &omap_mcbsp_component, > &omap_mcbsp_dai, 1); > - if (ret) > - return ret; > - > - return sdma_pcm_platform_register(&pdev->dev, "tx", "rx"); > } It's not clear to me how this fixes the problem, your commit message doesn't mention how? I was expecting to see more error handling paths being added to unwind the sysfs allocation, or a conversion to devm. As things stand it's not clear to me that the error won't persist in the case where we defer registering the component.

1 0

[PATCH 0/2] ASoC/SoundWire: Intel: add sdw BE dai trigger
by Bard Liao 14 Jun '22

14 Jun '22

For SOF IPC4, we need to set pipeline state in BE DAI trigger. @Vinod, Could you ack if the soundwire patch looks good to you? And we can go through the ASoC tree since the change is mainly on ASoC. Bard Liao (2): soundwire: Intel: add trigger callback ASoC: SOF: Intel: add trigger callback into sdw_callback drivers/soundwire/intel.c | 8 ++++++++ include/linux/soundwire/sdw_intel.h | 1 + sound/soc/sof/intel/hda-dai.c | 15 ++++++++++++--- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/intel/hda.h | 1 + 5 files changed, 23 insertions(+), 4 deletions(-) -- 2.17.1

3 6

[PATCH] ASoC: SOF: ipc4-topology: add SoundWire/ALH aggregation support
by Bard Liao 14 Jun '22

14 Jun '22

Some SoundWire hardware topologies rely on different amplifiers or capture devices connected on different links. These devices need to be 'aggregated', remain synchronized and be handled as a single logical device. In the IPC3 solution, the aggregation for amplifiers was handled by a firmware 'demux' component. In the IPC4 solution, the demux component is not needed, the gateway component can handle multiple ALH/DMA transfers at the same time. This change makes the topology slightly more complicated in that only one ALH DAI will be connected in the topology with the gateway. The other DAIs that are part of the 'aggregated' dailink are not shown in the DAPM graph as connected to the gateway, but they will however be activated thanks to a feature in soc-dapm.c where events are forwarded to all DAIs in the dailink (see soc_dapm_stream_event). The topology also sets the same stream name for all widgets, dais and dailinks, so a search for the stream name helps identify cases where SoundWire/ALH aggregation is needed. Signed-off-by: Bard Liao <yung-chuan.liao(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/ipc4-topology.c | 68 ++++++++++++++++++++++++++++++++--- sound/soc/sof/ipc4-topology.h | 11 ++++++ 2 files changed, 74 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index cb0f0823b8eb..3c949298e007 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -19,6 +19,8 @@ #define SOF_IPC4_GAIN_PARAM_ID 0 #define SOF_IPC4_TPLG_ABI_SIZE 6 +static DEFINE_IDA(alh_group_ida); + static const struct sof_topology_token ipc4_sched_tokens[] = { {SOF_TKN_SCHED_LP_MODE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, offsetof(struct sof_ipc4_pipeline, lp_mode)} @@ -478,7 +480,9 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) switch (ipc4_copier->dai_type) { case SOF_DAI_INTEL_ALH: { + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); struct sof_ipc4_alh_configuration_blob *blob; + struct snd_sof_widget *w; blob = kzalloc(sizeof(*blob), GFP_KERNEL); if (!blob) { @@ -486,6 +490,14 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) goto err; } + list_for_each_entry(w, &sdev->widget_list, list) { + if (w->widget->sname && + strcmp(w->widget->sname, swidget->widget->sname)) + continue; + + blob->alh_cfg.count++; + } + ipc4_copier->copier_config = (uint32_t *)blob; ipc4_copier->data.gtw_cfg.config_length = sizeof(*blob) >> 2; break; @@ -844,6 +856,17 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget) struct snd_sof_dai *dai = swidget->private; ipc4_copier = dai->private; + if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) { + struct sof_ipc4_alh_configuration_blob *blob; + unsigned int group_id; + + blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config; + if (blob->alh_cfg.count > 1) { + group_id = SOF_IPC4_NODE_INDEX(ipc4_copier->data.gtw_cfg.node_id) - + ALH_MULTI_GTW_BASE; + ida_free(&alh_group_ida, group_id); + } + } } if (ipc4_copier) { @@ -973,6 +996,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, struct sof_ipc4_copier_data *copier_data; struct snd_pcm_hw_params *ref_params; struct sof_ipc4_copier *ipc4_copier; + struct snd_sof_dai *dai; struct snd_mask *fmt; int out_sample_valid_bits; size_t ref_audio_fmt_size; @@ -1022,7 +1046,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, case snd_soc_dapm_dai_in: case snd_soc_dapm_dai_out: { - struct snd_sof_dai *dai = swidget->private; + dai = swidget->private; ipc4_copier = (struct sof_ipc4_copier *)dai->private; copier_data = &ipc4_copier->data; @@ -1077,22 +1101,56 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, */ if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) { struct sof_ipc4_alh_configuration_blob *blob; + struct sof_ipc4_copier_data *alh_data; + struct sof_ipc4_copier *alh_copier; + struct snd_sof_widget *w; + u32 ch_mask = 0; u32 ch_map; int i; blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config; - /* TODO: add aggregation mode support */ - blob->alh_cfg.count = 1; - blob->alh_cfg.mapping[0].alh_id = copier_data->gtw_cfg.node_id; + blob->gw_attr.lp_buffer_alloc = 0; /* Get channel_mask from ch_map */ ch_map = copier_data->base_config.audio_fmt.ch_map; for (i = 0; ch_map; i++) { if ((ch_map & 0xf) != 0xf) - blob->alh_cfg.mapping[0].channel_mask |= BIT(i); + ch_mask |= BIT(i); ch_map >>= 4; } + + /* + * Set each gtw_cfg.node_id to blob->alh_cfg.mapping[] + * for all widgets with the same stream name + */ + i = 0; + list_for_each_entry(w, &sdev->widget_list, list) { + if (w->widget->sname && + strcmp(w->widget->sname, swidget->widget->sname)) + continue; + + dai = w->private; + alh_copier = (struct sof_ipc4_copier *)dai->private; + alh_data = &alh_copier->data; + blob->alh_cfg.mapping[i].alh_id = alh_data->gtw_cfg.node_id; + blob->alh_cfg.mapping[i].channel_mask = ch_mask; + i++; + } + if (blob->alh_cfg.count > 1) { + int group_id; + + group_id = ida_alloc_max(&alh_group_ida, ALH_MULTI_GTW_COUNT, + GFP_KERNEL); + + if (group_id < 0) + return group_id; + + /* add multi-gateway base */ + group_id += ALH_MULTI_GTW_BASE; + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; + copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(group_id); + } } } } diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 1a9c0627bae9..3bc2fe38c733 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -42,6 +42,17 @@ #define ALH_MAX_NUMBER_OF_GTW 16 +/* + * The base of multi-gateways. Multi-gateways addressing starts from + * ALH_MULTI_GTW_BASE and there are ALH_MULTI_GTW_COUNT multi-sources + * and ALH_MULTI_GTW_COUNT multi-sinks available. + * Addressing is continuous from ALH_MULTI_GTW_BASE to + * ALH_MULTI_GTW_BASE + ALH_MULTI_GTW_COUNT - 1. + */ +#define ALH_MULTI_GTW_BASE 0x50 +/* A magic number from FW */ +#define ALH_MULTI_GTW_COUNT 8 + /** * struct sof_ipc4_pipeline - pipeline config data * @priority: Priority of this pipeline -- 2.17.1

2 1

[PATCH] ALSA: pcm: Test for "silence" field in struct "pcm_format_data"
by Fabio M. De Francesco 14 Jun '22

14 Jun '22

Syzbot reports "KASAN: null-ptr-deref Write in snd_pcm_format_set_silence".[1] It is due to missing validation of the "silence" field of struct "pcm_format_data" in "pcm_formats" array. Add a test for valid "pat" and, if it is not so, return -EINVAL. [1] https://lore.kernel.org/lkml/000000000000d188ef05dc2c7279@google.com/ Reported-and-tested-by: syzbot+205eb15961852c2c5974(a)syzkaller.appspotmail.com Signed-off-by: Fabio M. De Francesco <fmdefrancesco(a)gmail.com> --- I wasn't able to figure out the commit for the "Fixes:" tag. If this patch is good, can someone please help with providing this missing information? sound/core/pcm_misc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 4866aed97aac..5588b6a1ee8b 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -433,7 +433,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int return 0; width = pcm_formats[(INT)format].phys; /* physical width */ pat = pcm_formats[(INT)format].silence; - if (! width) + if (!width || !pat) return -EINVAL; /* signed or 1 byte data */ if (pcm_formats[(INT)format].signd == 1 || width <= 8) { -- 2.34.1

3 8

seq: add boxed structure for event
by GitHub pull_request - opened 14 Jun '22

14 Jun '22

alsa-project/alsa-gobject pull request #75 was opened from takaswie: ``` Current implementation provides ALSASeq.EventCntr to handle batch of sequencer events. Parameters of each event are set/get by public methods of it with index of event. But it's necesarrily convenient. This patchset adds boxed structure to express each of the event. Some public methods are rewritten according to it, with code refactoring. ALSASeq.EventCntr is largely rewritten with serializer/deserializer for flattened event buffer. Takashi Sakamoto (12): update README seq: event-cntr/event-data-queue: obsolete timestamp functions by adding alternatives seq: tstamp: remove unused boxed structure seq: remove-filter: remove unused error arguments seq: enums: rename to tstamp mode seq: port-info: rename properties for time stamp seq: enums: obsolete ALSASeqSubscribeFlag enumerations seq: event: add boxed structure for event seq: user-client: add public API to schedule list of events seq: user-client: rewrite public API to schedule single event seq: event-cntr: rewrite as boxed structure seq: event-cntr: add internal API to serialize list of events README.rst | 18 +- samples/seq | 96 +- src/seq/alsaseq-enum-types.h | 49 +- src/seq/alsaseq.h | 3 +- src/seq/alsaseq.map | 133 +-- src/seq/event-cntr.c | 1709 ++-------------------------------- src/seq/event-cntr.h | 115 +-- src/seq/event-data-queue.c | 52 +- src/seq/event-data-queue.h | 13 +- src/seq/event.c | 1687 +++++++++++++++++++++++++++++++++ src/seq/event.h | 112 +++ src/seq/meson.build | 4 +- src/seq/port-info.c | 118 ++- src/seq/privates.h | 5 + src/seq/remove-filter.c | 31 +- src/seq/remove-filter.h | 17 +- src/seq/subscribe-data.c | 115 ++- src/seq/tstamp.c | 81 -- src/seq/tstamp.h | 24 - src/seq/user-client.c | 153 ++- src/seq/user-client.h | 18 +- tests/alsaseq-enums | 20 +- tests/alsaseq-event-cntr | 60 -- tests/alsaseq-port-info | 4 +- tests/alsaseq-subscribe-data | 4 +- tests/alsaseq-user-client | 1 + tests/meson.build | 1 - 27 files changed, 2452 insertions(+), 2191 deletions(-) create mode 100644 src/seq/event.c create mode 100644 src/seq/event.h delete mode 100644 src/seq/tstamp.c delete mode 100644 src/seq/tstamp.h delete mode 100644 tests/alsaseq-event-cntr ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/75 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/75.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH] ASoC: SOF: Compile and runtime IPC version selection
by Peter Ujfalusi 14 Jun '22

14 Jun '22

The new IPC4 version is only supported by Intel platforms, iMX, AMD and MediaTek only uses the standard SOF IPC. There is no need for these platforms to build kernel support for IPC4 as it is just dead code for them. SND_SOC_SOF_IPC3 and SND_SOC_SOF_INTEL_IPC4 is introduced to allow compile time selection and exclusion of IPC implementations. To avoid randconfig failures add also support for runtime selection of the IPC ops in ipc.c based on sdev->pdata->ipc_type Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/Kconfig | 7 +++++++ sound/soc/sof/Makefile | 16 ++++++++++++---- sound/soc/sof/amd/Kconfig | 1 + sound/soc/sof/imx/Kconfig | 1 + sound/soc/sof/intel/Kconfig | 11 +++++++++++ sound/soc/sof/ipc.c | 24 ++++++++++++++++++------ sound/soc/sof/mediatek/Kconfig | 1 + 7 files changed, 51 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 4542868cd730..e90f173d067c 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -252,6 +252,13 @@ config SND_SOC_SOF_PROBE_WORK_QUEUE When selected, the probe is handled in two steps, for example to avoid lockdeps if request_module is used in the probe. +# Supported IPC versions +config SND_SOC_SOF_IPC3 + bool + +config SND_SOC_SOF_INTEL_IPC4 + bool + source "sound/soc/sof/amd/Kconfig" source "sound/soc/sof/imx/Kconfig" source "sound/soc/sof/intel/Kconfig" diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 2fa8088707a8..9a74ed116ed9 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -1,10 +1,18 @@ # SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ - control.o trace.o iomem-utils.o sof-audio.o stream-ipc.o\ - ipc3-topology.o ipc3-control.o ipc3.o ipc3-pcm.o ipc3-loader.o\ - ipc3-dtrace.o\ - ipc4.o ipc4-loader.o ipc4-topology.o ipc4-control.o ipc4-pcm.o + control.o trace.o iomem-utils.o sof-audio.o stream-ipc.o + +# IPC implementations +ifneq ($(CONFIG_SND_SOC_SOF_IPC3),) +snd-sof-objs += ipc3.o ipc3-loader.o ipc3-topology.o ipc3-control.o ipc3-pcm.o\ + ipc3-dtrace.o +endif +ifneq ($(CONFIG_SND_SOC_SOF_INTEL_IPC4),) +snd-sof-objs += ipc4.o ipc4-loader.o ipc4-topology.o ipc4-control.o ipc4-pcm.o +endif + +# SOF client support ifneq ($(CONFIG_SND_SOC_SOF_CLIENT),) snd-sof-objs += sof-client.o endif diff --git a/sound/soc/sof/amd/Kconfig b/sound/soc/sof/amd/Kconfig index 085232e04582..190c85d57047 100644 --- a/sound/soc/sof/amd/Kconfig +++ b/sound/soc/sof/amd/Kconfig @@ -17,6 +17,7 @@ if SND_SOC_SOF_AMD_TOPLEVEL config SND_SOC_SOF_AMD_COMMON tristate select SND_SOC_SOF + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_PCI_DEV select SND_AMD_ACP_CONFIG select SND_SOC_ACPI if ACPI diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 9b8d5bb1e449..cc6e695f913a 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -15,6 +15,7 @@ config SND_SOC_SOF_IMX_COMMON tristate select SND_SOC_SOF_OF_DEV select SND_SOC_SOF + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_XTENSA select SND_SOC_SOF_COMPRESS help diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 0def2aa5581d..80cdc3788bbe 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -40,6 +40,7 @@ if SND_SOC_SOF_ACPI config SND_SOC_SOF_BAYTRAIL tristate "SOF support for Baytrail, Braswell and Cherrytrail" default SND_SOC_SOF_ACPI + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_INTEL_ATOM_HIFI_EP select SND_SOC_SOF_ACPI_DEV @@ -60,6 +61,7 @@ config SND_SOC_SOF_BAYTRAIL config SND_SOC_SOF_BROADWELL tristate "SOF support for Broadwell" default SND_SOC_SOF_ACPI + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_INTEL_HIFI_EP_IPC select SND_SOC_SOF_ACPI_DEV @@ -85,6 +87,7 @@ config SND_SOC_SOF_MERRIFIELD tristate "SOF support for Tangier/Merrifield" default SND_SOC_SOF_PCI select SND_SOC_SOF_PCI_DEV + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_INTEL_ATOM_HIFI_EP help This adds support for Sound Open Firmware for Intel(R) platforms @@ -95,6 +98,8 @@ config SND_SOC_SOF_MERRIFIELD config SND_SOC_SOF_INTEL_APL tristate select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_IPC3 + select SND_SOC_SOF_INTEL_IPC4 config SND_SOC_SOF_APOLLOLAKE tristate "SOF support for Apollolake" @@ -120,6 +125,8 @@ config SND_SOC_SOF_INTEL_CNL tristate select SND_SOC_SOF_HDA_COMMON select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE + select SND_SOC_SOF_IPC3 + select SND_SOC_SOF_INTEL_IPC4 config SND_SOC_SOF_CANNONLAKE tristate "SOF support for Cannonlake" @@ -154,6 +161,8 @@ config SND_SOC_SOF_INTEL_ICL tristate select SND_SOC_SOF_HDA_COMMON select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE + select SND_SOC_SOF_IPC3 + select SND_SOC_SOF_INTEL_IPC4 config SND_SOC_SOF_ICELAKE tristate "SOF support for Icelake" @@ -179,6 +188,8 @@ config SND_SOC_SOF_INTEL_TGL tristate select SND_SOC_SOF_HDA_COMMON select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE + select SND_SOC_SOF_IPC3 + select SND_SOC_SOF_INTEL_IPC4 config SND_SOC_SOF_TIGERLAKE tristate "SOF support for Tigerlake" diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index c5aef5fc056b..6ed3f9b6a0c4 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -155,12 +155,22 @@ struct snd_sof_ipc *snd_sof_ipc_init(struct snd_sof_dev *sdev) init_waitqueue_head(&msg->waitq); - /* - * Use IPC3 ops as it is the only available version now. With the addition of new IPC - * versions, this will need to be modified to use the selected version at runtime. - */ - ipc->ops = &ipc3_ops; - ops = ipc->ops; + switch (sdev->pdata->ipc_type) { +#if defined(CONFIG_SND_SOC_SOF_IPC3) + case SOF_IPC: + ops = &ipc3_ops; + break; +#endif +#if defined(CONFIG_SND_SOC_SOF_INTEL_IPC4) + case SOF_INTEL_IPC4: + ops = &ipc4_ops; + break; +#endif + default: + dev_err(sdev->dev, "Not supported IPC version: %d\n", + sdev->pdata->ipc_type); + return NULL; + } /* check for mandatory ops */ if (!ops->tx_msg || !ops->rx_msg || !ops->set_get_data || !ops->get_reply) { @@ -190,6 +200,8 @@ struct snd_sof_ipc *snd_sof_ipc_init(struct snd_sof_dev *sdev) return NULL; } + ipc->ops = ops; + return ipc; } EXPORT_SYMBOL(snd_sof_ipc_init); diff --git a/sound/soc/sof/mediatek/Kconfig b/sound/soc/sof/mediatek/Kconfig index a149dd1b3f44..4a2eddf6009a 100644 --- a/sound/soc/sof/mediatek/Kconfig +++ b/sound/soc/sof/mediatek/Kconfig @@ -15,6 +15,7 @@ config SND_SOC_SOF_MTK_COMMON tristate select SND_SOC_SOF_OF_DEV select SND_SOC_SOF + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_XTENSA select SND_SOC_SOF_COMPRESS help -- 2.36.1

2 1

[PATCH] ASoC: SOF: topology: add code to parse config params for ACPDMIC dai
by Peter Ujfalusi 14 Jun '22

14 Jun '22

From: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> Add sof_ipc_dai_acpdmic_params and tokens to parse dmic channels and rate params from topology file Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- include/sound/sof/dai-amd.h | 7 +++++++ include/sound/sof/dai.h | 2 +- include/uapi/sound/sof/tokens.h | 4 ++++ sound/soc/sof/ipc3-pcm.c | 8 ++++---- sound/soc/sof/ipc3-topology.c | 25 +++++++++++++++++++------ sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/topology.c | 4 ++++ 7 files changed, 40 insertions(+), 11 deletions(-) diff --git a/include/sound/sof/dai-amd.h b/include/sound/sof/dai-amd.h index 90d09dbdd709..92f45c180b7c 100644 --- a/include/sound/sof/dai-amd.h +++ b/include/sound/sof/dai-amd.h @@ -18,4 +18,11 @@ struct sof_ipc_dai_acp_params { uint32_t fsync_rate; /* FSYNC frequency in Hz */ uint32_t tdm_slots; } __packed; + +/* ACPDMIC Configuration Request - SOF_IPC_DAI_AMD_CONFIG */ +struct sof_ipc_dai_acpdmic_params { + uint32_t pdm_rate; + uint32_t pdm_ch; +} __packed; + #endif diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h index a818a0f0a226..21d98f31a9ca 100644 --- a/include/sound/sof/dai.h +++ b/include/sound/sof/dai.h @@ -111,7 +111,7 @@ struct sof_ipc_dai_config { struct sof_ipc_dai_sai_params sai; struct sof_ipc_dai_acp_params acpbt; struct sof_ipc_dai_acp_params acpsp; - struct sof_ipc_dai_acp_params acpdmic; + struct sof_ipc_dai_acpdmic_params acpdmic; struct sof_ipc_dai_mtk_afe_params afe; }; } __packed; diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h index f7b2019065ad..5caf75cadaf8 100644 --- a/include/uapi/sound/sof/tokens.h +++ b/include/uapi/sound/sof/tokens.h @@ -157,6 +157,10 @@ /* MIXER */ #define SOF_TKN_MIXER_TYPE 1700 +/* ACPDMIC */ +#define SOF_TKN_AMD_ACPDMIC_RATE 1800 +#define SOF_TKN_AMD_ACPDMIC_CH 1801 + /* CAVS AUDIO FORMAT */ #define SOF_TKN_CAVS_AUDIO_FORMAT_IN_RATE 1900 #define SOF_TKN_CAVS_AUDIO_FORMAT_IN_BIT_DEPTH 1901 diff --git a/sound/soc/sof/ipc3-pcm.c b/sound/soc/sof/ipc3-pcm.c index c8774a891d6f..b97e63d3724a 100644 --- a/sound/soc/sof/ipc3-pcm.c +++ b/sound/soc/sof/ipc3-pcm.c @@ -344,10 +344,10 @@ static int sof_ipc3_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, channels->min, channels->max); break; case SOF_DAI_AMD_DMIC: - rate->min = private->dai_config->acpdmic.fsync_rate; - rate->max = private->dai_config->acpdmic.fsync_rate; - channels->min = private->dai_config->acpdmic.tdm_slots; - channels->max = private->dai_config->acpdmic.tdm_slots; + rate->min = private->dai_config->acpdmic.pdm_rate; + rate->max = private->dai_config->acpdmic.pdm_rate; + channels->min = private->dai_config->acpdmic.pdm_ch; + channels->max = private->dai_config->acpdmic.pdm_ch; dev_dbg(component->dev, "AMD_DMIC rate_min: %d rate_max: %d\n", rate->min, rate->max); diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index d20e71be8b8e..28d3c1414572 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -266,6 +266,16 @@ static const struct sof_topology_token afe_tokens[] = { offsetof(struct sof_ipc_dai_mtk_afe_params, format)}, }; +/* ACPDMIC */ +static const struct sof_topology_token acpdmic_tokens[] = { + {SOF_TKN_AMD_ACPDMIC_RATE, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_acpdmic_params, pdm_rate)}, + {SOF_TKN_AMD_ACPDMIC_CH, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_acpdmic_params, pdm_ch)}, +}; + /* Core tokens */ static const struct sof_topology_token core_tokens[] = { {SOF_TKN_COMP_CORE_ID, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, @@ -300,6 +310,7 @@ static const struct sof_token_info ipc3_token_list[SOF_TOKEN_COUNT] = { [SOF_ESAI_TOKENS] = {"ESAI tokens", esai_tokens, ARRAY_SIZE(esai_tokens)}, [SOF_SAI_TOKENS] = {"SAI tokens", sai_tokens, ARRAY_SIZE(sai_tokens)}, [SOF_AFE_TOKENS] = {"AFE tokens", afe_tokens, ARRAY_SIZE(afe_tokens)}, + [SOF_ACPDMIC_TOKENS] = {"ACPDMIC tokens", acpdmic_tokens, ARRAY_SIZE(acpdmic_tokens)}, }; /** @@ -1120,20 +1131,22 @@ static int sof_link_acp_dmic_load(struct snd_soc_component *scomp, struct snd_so struct snd_soc_tplg_hw_config *hw_config = slink->hw_configs; struct sof_dai_private_data *private = dai->private; u32 size = sizeof(*config); + int ret; /* handle master/slave and inverted clocks */ sof_dai_set_format(hw_config, config); - /* init IPC */ - memset(&config->acpdmic, 0, sizeof(config->acpdmic)); config->hdr.size = size; - config->acpdmic.fsync_rate = le32_to_cpu(hw_config->fsync_rate); - config->acpdmic.tdm_slots = le32_to_cpu(hw_config->tdm_slots); + /* parse the required set of ACPDMIC tokens based on num_hw_cfgs */ + ret = sof_update_ipc_object(scomp, &config->acpdmic, SOF_ACPDMIC_TOKENS, slink->tuples, + slink->num_tuples, size, slink->num_hw_configs); + if (ret < 0) + return ret; dev_info(scomp->dev, "ACP_DMIC config ACP%d channel %d rate %d\n", - config->dai_index, config->acpdmic.tdm_slots, - config->acpdmic.fsync_rate); + config->dai_index, config->acpdmic.pdm_ch, + config->acpdmic.pdm_rate); dai->number_configs = 1; dai->current_config = 0; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 79486266081f..4284ea2f3a1f 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -236,6 +236,7 @@ enum sof_tokens { SOF_AUDIO_FMT_NUM_TOKENS, SOF_COPIER_FORMAT_TOKENS, SOF_GAIN_TOKENS, + SOF_ACPDMIC_TOKENS, /* this should be the last */ SOF_TOKEN_COUNT, diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 1893c590f2f0..7e54eb1bf77b 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1739,6 +1739,10 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, struct snd_ token_id = SOF_AFE_TOKENS; num_tuples += token_list[SOF_AFE_TOKENS].count; break; + case SOF_DAI_AMD_DMIC: + token_id = SOF_ACPDMIC_TOKENS; + num_tuples += token_list[SOF_ACPDMIC_TOKENS].count; + break; default: break; } -- 2.36.1

2 1

[PATCH 1/2] ASoC: sun8i-codec: Partial revert to fix clock specifiers
by Charles Keepax 14 Jun '22

14 Jun '22

Recent updates accidentally updated the clock producer/consumer specifiers on this device as part of refactoring the CPU side of the DAI links. However, this device sits on the CODEC side and shouldn't have been updated. Partially revert the changes keeping the switch to the new clock terminology but going back to the CODEC defines. Fixes: 7cc3965fde74 ("ASoC: sunxi: Update to use set_fmt_new callback") Reported-by: Samuel Holland <samuel(a)sholland.org> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/sunxi/sun8i-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 90d74a2d53f38..f797c535f2983 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -287,10 +287,10 @@ static int sun8i_codec_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* clock masters */ switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_BP_FP: /* Codec slave, DAI master */ + case SND_SOC_DAIFMT_CBC_CFC: /* Codec slave, DAI master */ value = 0x1; break; - case SND_SOC_DAIFMT_BC_FC: /* Codec Master, DAI slave */ + case SND_SOC_DAIFMT_CBP_CFP: /* Codec Master, DAI slave */ value = 0x0; break; default: -- 2.30.2

3 4

[PATCH] ASoC: SOF: reduce default verbosity of IPC logs
by Pierre-Louis Bossart 14 Jun '22

14 Jun '22

We currently log the initiation of an IPC as well at its success. [ 3906.106987] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc tx: 0x80010000: GLB_DAI_MSG: CONFIG [ 3906.107189] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc tx succeeded: 0x80010000: GLB_DAI_MSG: CONFIG This is overkill in most cases, we already have a message thrown in case of errors and have tracepoints enabled to check for IPC duration. The only case where this might be useful is to check if there is an interleaved IPC RX. Add a flag and only print those logs if enabled. In addition, the DMA_POSITION_UPDATE for traces brings limited information in most cases and pollutes the logs for no good reason. [ 3906.322256] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3906.322308] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx done: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3906.822261] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3906.822319] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx done: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3907.822261] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3907.822319] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx done: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3908.822251] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3908.822309] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx done: 0x90020000: GLB_TRACE_MSG: DMA_POSITION This information is only helpful when debugging the trace support, not when using the trace. Add a flag to only print DMA position update logs if enabled. 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/sof/ipc3.c | 5 ++++- sound/soc/sof/sof-priv.h | 6 ++++++ 2 files changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index dff5feaad3703..7608a3b0aeb21 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -147,6 +147,8 @@ static void ipc3_log_header(struct device *dev, u8 *text, u32 cmd) case SOF_IPC_TRACE_DMA_PARAMS: str2 = "DMA_PARAMS"; break; case SOF_IPC_TRACE_DMA_POSITION: + if (!sof_debug_check_flag(SOF_DBG_PRINT_DMA_POSITION_UPDATE_LOGS)) + return; str2 = "DMA_POSITION"; break; case SOF_IPC_TRACE_DMA_PARAMS_EXT: str2 = "DMA_PARAMS_EXT"; break; @@ -299,7 +301,8 @@ static int ipc3_wait_tx_done(struct snd_sof_ipc *ipc, void *reply_data) "ipc tx error for %#x (msg/reply size: %d/%zu): %d\n", hdr->cmd, hdr->size, msg->reply_size, ret); } else { - ipc3_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd); + if (sof_debug_check_flag(SOF_DBG_PRINT_IPC_SUCCESS_LOGS)) + ipc3_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd); if (msg->reply_size) /* copy the data returned from DSP */ memcpy(reply_data, msg->reply_data, diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 9d7f53ff9c70e..afbb37ec841ea 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -37,6 +37,12 @@ #define SOF_DBG_IGNORE_D3_PERSISTENT BIT(7) /* ignore the DSP D3 persistent capability * and always download firmware upon D3 exit */ +#define SOF_DBG_PRINT_DMA_POSITION_UPDATE_LOGS BIT(8) /* print DMA position updates + * in dmesg logs + */ +#define SOF_DBG_PRINT_IPC_SUCCESS_LOGS BIT(9) /* print IPC success + * in dmesg logs + */ /* Flag definitions used for controlling the DSP dump behavior */ #define SOF_DBG_DUMP_REGS BIT(0) -- 2.34.1

2 1

[PATCH] ASoC: SOF: Intel: hda-dai: enhance debug messages
by Pierre-Louis Bossart 14 Jun '22

14 Jun '22

The same message was added twice for dai and link_dma, remove the latter one and add dai name and direction to better understand problematic sequences. 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/sof/intel/hda-dai.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 9823230d2ef4a..cb3d253558fe3 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -257,7 +257,6 @@ static int hda_link_dma_trigger(struct snd_pcm_substream *substream, int cmd) struct hdac_ext_stream *hext_stream = snd_soc_dai_get_dma_data(cpu_dai, substream); int ret; - dev_dbg(cpu_dai->dev, "%s: cmd=%d\n", __func__, cmd); if (!hext_stream) return 0; @@ -408,13 +407,15 @@ static int ipc3_hda_dai_trigger(struct snd_pcm_substream *substream, struct snd_soc_dapm_widget *w; int ret; + dev_dbg(dai->dev, "%s: cmd=%d dai %s direction %d\n", __func__, cmd, + dai->name, substream->stream); + ret = hda_link_dma_trigger(substream, cmd); if (ret < 0) return ret; w = snd_soc_dai_get_widget(dai, substream->stream); - dev_dbg(dai->dev, "%s: cmd=%d\n", __func__, cmd); switch (cmd) { case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: -- 2.34.1

2 1

[PATCH] ASoC: intel: sof_sdw: add RT711 SDCA card for MTL platform
by Pierre-Louis Bossart 14 Jun '22

14 Jun '22

From: Yong Zhi <yong.zhi(a)intel.com> Enable on-board rt711 based sound card for MTL RVP. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Uday M Bhat <uday.m.bhat(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/sof_sdw.c | 8 +++++ .../intel/common/soc-acpi-intel-mtl-match.c | 31 +++++++++++++++++++ 2 files changed, 39 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 77af3a7655c44..d23846572543b 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -315,6 +315,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { RT711_JD2 | SOF_SDW_FOUR_SPK), }, + /* MeteorLake devices */ + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Intel_mtlrvp"), + }, + .driver_data = (void *)(RT711_JD1 | SOF_SDW_TGL_HDMI), + }, {} }; diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index cc594b27e03be..74d3b82f8d35c 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -15,6 +15,31 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_machines); +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, +}; + +static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { + { + .adr = 0x000030025D071101ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt711" + } +}; + +static const struct snd_soc_acpi_link_adr mtl_rvp[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_sdca_0_adr), + .adr_d = rt711_sdca_0_adr, + }, + {} +}; + /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { /* mockup tests need to be first */ @@ -36,6 +61,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-mtl-rt715-rt711-rt1308-mono.tplg", }, + { + .link_mask = BIT(0), + .links = mtl_rvp, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt711.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_sdw_machines); -- 2.34.1

2 1

[PATCH 1/2] ASoC: dt-bindings: fsl, mqs: Add compatible string for i.MX93 platform
by Shengjiu Wang 14 Jun '22

14 Jun '22

Add compatible string "fsl,imx93-mqs" for i.MX93 platform Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,mqs.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.txt b/Documentation/devicetree/bindings/sound/fsl,mqs.txt index 40353fc30255..d66284b8bef2 100644 --- a/Documentation/devicetree/bindings/sound/fsl,mqs.txt +++ b/Documentation/devicetree/bindings/sound/fsl,mqs.txt @@ -2,7 +2,7 @@ fsl,mqs audio CODEC Required properties: - compatible : Must contain one of "fsl,imx6sx-mqs", "fsl,codec-mqs" - "fsl,imx8qm-mqs", "fsl,imx8qxp-mqs". + "fsl,imx8qm-mqs", "fsl,imx8qxp-mqs", "fsl,imx93-mqs". - clocks : A list of phandles + clock-specifiers, one for each entry in clock-names - clock-names : "mclk" - must required. -- 2.17.1

2 2

[PATCH 0/2] soundwire: Intel: add trigger callback
by Bard Liao 14 Jun '22

14 Jun '22

The .trigger callback will be used in ASoC. Hi Vinod, Could you provied a tag so that Mark can take the tag to his tree? Bard Liao (1): soundwire: Intel: add trigger callback Pierre-Louis Bossart (1): soundwire: intel: uniquify debug message drivers/soundwire/intel.c | 36 ++++++++++++++++++----------- include/linux/soundwire/sdw_intel.h | 1 + 2 files changed, 23 insertions(+), 14 deletions(-) -- 2.17.1

4 11

[PATCH] ALSA: hda/realtek: Apply fixup for Lenovo Yoga Duet 7 properly
by Takashi Iwai 14 Jun '22

14 Jun '22

It turned out that Lenovo shipped two completely different products with the very same PCI SSID, where both require different quirks; namely, Lenovo C940 has already the fixup for its speaker (ALC298_FIXUP_LENOVO_SPK_VOLUME) with the PCI SSID 17aa:3818, while Yoga Duet 7 has also the very same PCI SSID but requires a different quirk, ALC287_FIXUP_YOGA7_14TIL_SPEAKERS. Fortunately, both are with different codecs (C940 with ALC298 and Duet 7 with ALC287), hence we can apply different fixes by checking the codec ID. This patch implements that special fixup function. For easier handling, the internal function for applying a specific fixup entry is exported as __snd_hda_apply_fixup(), so that it can be called from the codec driver. The rest is simply calling it with a different fixup ID depending on the codec ID. Reported-by: Hans de Goede <hdegoede(a)redhat.com> Tested-by: nikitashvets(a)flyium.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/5ca147d1-3a2d-60c6-c491-8aa844183222@redhat.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_auto_parser.c | 7 ++++--- sound/pci/hda/hda_local.h | 1 + sound/pci/hda/patch_realtek.c | 24 +++++++++++++++++++++++- 3 files changed, 28 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index cd1db943b7e0..7c6b1fe8dfcc 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -819,7 +819,7 @@ static void set_pin_targets(struct hda_codec *codec, snd_hda_set_pin_ctl_cache(codec, cfg->nid, cfg->val); } -static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) +void __snd_hda_apply_fixup(struct hda_codec *codec, int id, int action, int depth) { const char *modelname = codec->fixup_name; @@ -829,7 +829,7 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) if (++depth > 10) break; if (fix->chained_before) - apply_fixup(codec, fix->chain_id, action, depth + 1); + __snd_hda_apply_fixup(codec, fix->chain_id, action, depth + 1); switch (fix->type) { case HDA_FIXUP_PINS: @@ -870,6 +870,7 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) id = fix->chain_id; } } +EXPORT_SYMBOL_GPL(__snd_hda_apply_fixup); /** * snd_hda_apply_fixup - Apply the fixup chain with the given action @@ -879,7 +880,7 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) void snd_hda_apply_fixup(struct hda_codec *codec, int action) { if (codec->fixup_list) - apply_fixup(codec, codec->fixup_id, action, 0); + __snd_hda_apply_fixup(codec, codec->fixup_id, action, 0); } EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index aca592651870..682dca2057db 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -348,6 +348,7 @@ void snd_hda_apply_verbs(struct hda_codec *codec); void snd_hda_apply_pincfgs(struct hda_codec *codec, const struct hda_pintbl *cfg); void snd_hda_apply_fixup(struct hda_codec *codec, int action); +void __snd_hda_apply_fixup(struct hda_codec *codec, int id, int action, int depth); void snd_hda_pick_fixup(struct hda_codec *codec, const struct hda_model_fixup *models, const struct snd_pci_quirk *quirk, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9b6ee775ee3f..b937f63d0d09 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7004,6 +7004,7 @@ enum { ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, + ALC298_FIXUP_LENOVO_C940_DUET7, ALC287_FIXUP_13S_GEN2_SPEAKERS, ALC256_FIXUP_SET_COEF_DEFAULTS, ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, @@ -7022,6 +7023,23 @@ enum { ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE, }; +/* A special fixup for Lenovo C940 and Yoga Duet 7; + * both have the very same PCI SSID, and we need to apply different fixups + * depending on the codec ID + */ +static void alc298_fixup_lenovo_c940_duet7(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + int id; + + if (codec->core.vendor_id == 0x10ec0298) + id = ALC298_FIXUP_LENOVO_SPK_VOLUME; /* C940 */ + else + id = ALC287_FIXUP_YOGA7_14ITL_SPEAKERS; /* Duet 7 */ + __snd_hda_apply_fixup(codec, id, action, 0); +} + static const struct hda_fixup alc269_fixups[] = { [ALC269_FIXUP_GPIO2] = { .type = HDA_FIXUP_FUNC, @@ -8721,6 +8739,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE, }, + [ALC298_FIXUP_LENOVO_C940_DUET7] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_lenovo_c940_duet7, + }, [ALC287_FIXUP_13S_GEN2_SPEAKERS] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -9274,7 +9296,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), - SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), + SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940 / Yoga Duet 7", ALC298_FIXUP_LENOVO_C940_DUET7), SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3820, "Yoga Duet 7 13ITL6", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3824, "Legion Y9000X 2020", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), -- 2.35.3

1 0

How to add 2 different SND_PCI_QUIRKs for 2 models with the same PCI subsys vend + prod ids?
by Hans de Goede 14 Jun '22

14 Jun '22

Hi All, There is a problem with the sound on the Lenovo Yoga Duet 7 13IML05, which requires a SND_PCI_QUIRK to fix. But there already is an existing quirk for another Lenovo laptop using the same PCI subsys vend + prod ids. For more details see: https://wiki.archlinux.org/title/Lenovo_Yoga_Duet_7_13IML05#Audio So I guess this means that we need to add a way to also include a DMI match for SND_PCI_QUIRKs ? Maybe add a: const struct dmi_systemid *dmi_ids; member to struct snd_pci_quirk and a new SND_PCI_QUIRK_DMI macro to set this? Regards, Hans

2 5

[PATCH AUTOSEL 4.9 01/12] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0d9c4a57301b..f733f6b42b53 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -141,7 +141,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -368,7 +368,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 4

[PATCH AUTOSEL 4.14 01/14] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0d9c4a57301b..f733f6b42b53 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -141,7 +141,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -368,7 +368,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 4

[PATCH AUTOSEL 4.19 09/18] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ] Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 02c557e1f779..c5b0b56d9c94 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -697,7 +697,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 08/18] ASoC: es8328: Fix event generation for deemphasis control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 8259610c2ec01c5cbfb61882ae176aabacac9c19 ] Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 3afa163f7652..dcb01889e177 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -165,13 +165,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 07/18] ASoC: wm8962: Fix suspend while playing music
by Sasha Levin 14 Jun '22

14 Jun '22

From: Adam Ford <aford173(a)gmail.com> [ Upstream commit d1f5272c0f7d2e53c6f2480f46725442776f5f78 ] If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220526182129.538472-1-aford173@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index dde015fd70a4..3f75cb3209ff 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3861,6 +3861,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 05/18] ASoC: cs42l56: Correct typo in minimum level for SX volume controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit a8928ada9b96944cadd8b65d191e33199fd38782 ] A couple of the SX volume controls specify 0x84 as the lowest volume value, however the correct value from the datasheet is 0x44. The datasheet don't include spaces in the value it displays as binary so this was almost certainly just a typo reading 1000100. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-6-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 04f89b751304..deaad703a7db 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -403,9 +403,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), -- 2.35.1

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[PATCH AUTOSEL 4.19 04/18] ASoC: cs42l52: Correct TLV for Bypass Volume
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91e90c712fade0b69cdff7cc6512f6099bd18ae5 ] The Bypass Volume is accidentally using a -6dB minimum TLV rather than the correct -60dB minimum. Add a new TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-5-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 9a3180e71bd8..de311299432b 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -141,6 +141,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -355,7 +357,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 03/18] ASoC: cs53l30: Correct number of volume levels on SX controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 7fbd6dd68127927e844912a16741016d432a0737 ] This driver specified the maximum value rather than the number of volume levels on the SX controls, this is incorrect, so correct them. Reported-by: David Rhodes <david.rhodes(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 8995ea45b4ca..86e93904b001 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -351,22 +351,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 02/18] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 3d83c1be1292..9a3180e71bd8 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -141,7 +141,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -368,7 +368,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 18/23] ALSA: hda/realtek - Add HW8326 support
by Sasha Levin 14 Jun '22

14 Jun '22

From: huangwenhui <huangwenhuia(a)uniontech.com> [ Upstream commit 527f4643e03c298c1e3321cfa27866b1374a55e1 ] Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> Link: https://lore.kernel.org/r/20220608082357.26898-1-huangwenhuia@uniontech.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index b84e12f4f804..489f996d86bc 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -656,6 +656,7 @@ static struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 78b5a0f22a41..57da27cd7498 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -382,6 +382,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -519,6 +520,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3184,6 +3186,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3212,6 +3215,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4674,6 +4678,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -4789,6 +4794,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -4939,6 +4945,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5038,6 +5045,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5152,6 +5160,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5248,6 +5257,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5542,6 +5552,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6145,6 +6156,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -9150,6 +9162,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -10592,6 +10605,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 12/23] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ] Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 13672928da99..aedfa6b2895b 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -791,7 +791,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 11/23] ASoC: es8328: Fix event generation for deemphasis control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 8259610c2ec01c5cbfb61882ae176aabacac9c19 ] Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index fdf64c29f563..4117ab6e9b6f 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 10/23] ASoC: wm8962: Fix suspend while playing music
by Sasha Levin 14 Jun '22

14 Jun '22

From: Adam Ford <aford173(a)gmail.com> [ Upstream commit d1f5272c0f7d2e53c6f2480f46725442776f5f78 ] If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220526182129.538472-1-aford173@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d9d59f45833f..ebaee468057b 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3854,6 +3854,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 08/23] ASoC: cs42l56: Correct typo in minimum level for SX volume controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit a8928ada9b96944cadd8b65d191e33199fd38782 ] A couple of the SX volume controls specify 0x84 as the lowest volume value, however the correct value from the datasheet is 0x44. The datasheet don't include spaces in the value it displays as binary so this was almost certainly just a typo reading 1000100. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-6-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 51d7a87ab4c3..8be7d83f0ce9 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 07/23] ASoC: cs42l52: Correct TLV for Bypass Volume
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91e90c712fade0b69cdff7cc6512f6099bd18ae5 ] The Bypass Volume is accidentally using a -6dB minimum TLV rather than the correct -60dB minimum. Add a new TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-5-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0daa3b4c68ca..6c054b357205 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,6 +137,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 06/23] ASoC: cs53l30: Correct number of volume levels on SX controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 7fbd6dd68127927e844912a16741016d432a0737 ] This driver specified the maximum value rather than the number of volume levels on the SX controls, this is incorrect, so correct them. Reported-by: David Rhodes <david.rhodes(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index ed22361b35c1..a5a383b92305 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -347,22 +347,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 05/23] ASoC: cs35l36: Update digital volume TLV
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 5005a2345825eb8346546d99bfe669f73111b5c5 ] The digital volume TLV specifies the step as 0.25dB but the actual step of the control is 0.125dB. Update the TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l36.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index e9b5f76f27a8..aa32b8c26578 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 04/23] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 2ea4cba3be2a..0daa3b4c68ca 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,7 +137,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -364,7 +364,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 02/23] ASoC: nau8822: Add operation for internal PLL off and on
by Sasha Levin 14 Jun '22

14 Jun '22

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit aeca8a3295022bcec46697f16e098140423d8463 ] We tried to enable the audio on an imx6sx EVB with the codec nau8822, after setting the internal PLL fractional parameters, the audio still couldn't work and the there was no sdma irq at all. After checking with the section "8.1.1 Phase Locked Loop (PLL) Design Example" of "NAU88C22 Datasheet Rev 0.6", we found we need to turn off the PLL before programming fractional parameters and turn on the PLL after programming. After this change, the audio driver could record and play sound and the sdma's irq is triggered when playing or recording. Cc: David Lin <ctlin0(a)nuvoton.com> Cc: John Hsu <kchsu0(a)nuvoton.com> Cc: Seven Li <wtli(a)nuvoton.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20220530040151.95221-2-hui.wang@canonical.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8822.c | 4 ++++ sound/soc/codecs/nau8822.h | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 78db3bd0b3bc..cd163978792e 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -740,6 +740,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, pll_param->pre_factor); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | @@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 21/29] ALSA: hda/realtek - Add HW8326 support
by Sasha Levin 14 Jun '22

14 Jun '22

From: huangwenhui <huangwenhuia(a)uniontech.com> [ Upstream commit 527f4643e03c298c1e3321cfa27866b1374a55e1 ] Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> Link: https://lore.kernel.org/r/20220608082357.26898-1-huangwenhuia@uniontech.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..b7e5032b61c9 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -660,6 +660,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 71a9462e8f6e..e067759561df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -439,6 +439,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -576,6 +577,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3252,6 +3254,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3280,6 +3283,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4849,6 +4853,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -4964,6 +4969,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5114,6 +5120,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5213,6 +5220,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5327,6 +5335,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5428,6 +5437,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5722,6 +5732,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6325,6 +6336,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -9812,6 +9824,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11254,6 +11267,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 14/29] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ] Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 51d95437e0fd..10189f44af28 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -800,7 +800,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 13/29] ASoC: es8328: Fix event generation for deemphasis control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 8259610c2ec01c5cbfb61882ae176aabacac9c19 ] Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 7e26231a596a..081b5f189632 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 12/29] ASoC: wm8962: Fix suspend while playing music
by Sasha Levin 14 Jun '22

14 Jun '22

From: Adam Ford <aford173(a)gmail.com> [ Upstream commit d1f5272c0f7d2e53c6f2480f46725442776f5f78 ] If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220526182129.538472-1-aford173@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 0bd3bbc2aacf..38651022e3d5 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3864,6 +3864,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 09/29] ASoC: cs42l51: Correct minimum value for SX volume control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit fcb3b5a58926d16d9a338841b74af06d4c29be15 ] The minimum value for the PGA Volume is given as 0x1A, however the values from there to 0x19 are all the same volume and this is not represented in the TLV structure. The number of volumes given is correct so this leads to all the volumes being shifted. Move the minimum value up to 0x19 to fix this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-7-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index c61b17dc2af8..fc6a2bc311b4 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -146,7 +146,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0xA0, 96, adc_att_tlv), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL, - 0, 0x1A, 30, pga_tlv), + 0, 0x19, 30, pga_tlv), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 08/29] ASoC: cs42l56: Correct typo in minimum level for SX volume controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit a8928ada9b96944cadd8b65d191e33199fd38782 ] A couple of the SX volume controls specify 0x84 as the lowest volume value, however the correct value from the datasheet is 0x44. The datasheet don't include spaces in the value it displays as binary so this was almost certainly just a typo reading 1000100. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-6-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 06dcfae9dfe7..d41e03193106 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), -- 2.35.1

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[PATCH AUTOSEL 5.10 07/29] ASoC: cs42l52: Correct TLV for Bypass Volume
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91e90c712fade0b69cdff7cc6512f6099bd18ae5 ] The Bypass Volume is accidentally using a -6dB minimum TLV rather than the correct -60dB minimum. Add a new TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-5-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 750e6c923512..38223641bdf6 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,6 +137,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), -- 2.35.1

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[PATCH AUTOSEL 5.10 06/29] ASoC: cs53l30: Correct number of volume levels on SX controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 7fbd6dd68127927e844912a16741016d432a0737 ] This driver specified the maximum value rather than the number of volume levels on the SX controls, this is incorrect, so correct them. Reported-by: David Rhodes <david.rhodes(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index ed22361b35c1..a5a383b92305 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -347,22 +347,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 05/29] ASoC: cs35l36: Update digital volume TLV
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 5005a2345825eb8346546d99bfe669f73111b5c5 ] The digital volume TLV specifies the step as 0.25dB but the actual step of the control is 0.125dB. Update the TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l36.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index e9b5f76f27a8..aa32b8c26578 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 04/29] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index f772628f233e..750e6c923512 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,7 +137,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -364,7 +364,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 02/29] ASoC: nau8822: Add operation for internal PLL off and on
by Sasha Levin 14 Jun '22

14 Jun '22

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit aeca8a3295022bcec46697f16e098140423d8463 ] We tried to enable the audio on an imx6sx EVB with the codec nau8822, after setting the internal PLL fractional parameters, the audio still couldn't work and the there was no sdma irq at all. After checking with the section "8.1.1 Phase Locked Loop (PLL) Design Example" of "NAU88C22 Datasheet Rev 0.6", we found we need to turn off the PLL before programming fractional parameters and turn on the PLL after programming. After this change, the audio driver could record and play sound and the sdma's irq is triggered when playing or recording. Cc: David Lin <ctlin0(a)nuvoton.com> Cc: John Hsu <kchsu0(a)nuvoton.com> Cc: Seven Li <wtli(a)nuvoton.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20220530040151.95221-2-hui.wang@canonical.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8822.c | 4 ++++ sound/soc/codecs/nau8822.h | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 609aeeb27818..d831959d8ff7 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -740,6 +740,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, pll_param->pre_factor); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | @@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 27/41] ALSA: hda/realtek - Add HW8326 support
by Sasha Levin 14 Jun '22

14 Jun '22

From: huangwenhui <huangwenhuia(a)uniontech.com> [ Upstream commit 527f4643e03c298c1e3321cfa27866b1374a55e1 ] Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> Link: https://lore.kernel.org/r/20220608082357.26898-1-huangwenhuia@uniontech.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..b7e5032b61c9 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -660,6 +660,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2b5a610744f6..a2dd23b3a6a3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -438,6 +438,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -575,6 +576,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3242,6 +3244,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3270,6 +3273,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4905,6 +4909,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5020,6 +5025,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5170,6 +5176,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5269,6 +5276,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5383,6 +5391,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5484,6 +5493,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5778,6 +5788,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6380,6 +6391,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -9878,6 +9890,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11328,6 +11341,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 18/41] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ] Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f7c800927cb2..08fc1a025b1a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -794,7 +794,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 17/41] ASoC: es8328: Fix event generation for deemphasis control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 8259610c2ec01c5cbfb61882ae176aabacac9c19 ] Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 9632afc2d4d6..ca3b1c00fa78 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 16/41] ASoC: wm8962: Fix suspend while playing music
by Sasha Levin 14 Jun '22

14 Jun '22

From: Adam Ford <aford173(a)gmail.com> [ Upstream commit d1f5272c0f7d2e53c6f2480f46725442776f5f78 ] If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220526182129.538472-1-aford173@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index ba16bdf9e478..a5a4ae4440cc 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3867,6 +3867,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.35.1

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[PATCH AUTOSEL 5.15 12/41] ASoC: cs42l51: Correct minimum value for SX volume control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit fcb3b5a58926d16d9a338841b74af06d4c29be15 ] The minimum value for the PGA Volume is given as 0x1A, however the values from there to 0x19 are all the same volume and this is not represented in the TLV structure. The number of volumes given is correct so this leads to all the volumes being shifted. Move the minimum value up to 0x19 to fix this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-7-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index c61b17dc2af8..fc6a2bc311b4 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -146,7 +146,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0xA0, 96, adc_att_tlv), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL, - 0, 0x1A, 30, pga_tlv), + 0, 0x19, 30, pga_tlv), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 11/41] ASoC: cs42l56: Correct typo in minimum level for SX volume controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit a8928ada9b96944cadd8b65d191e33199fd38782 ] A couple of the SX volume controls specify 0x84 as the lowest volume value, however the correct value from the datasheet is 0x44. The datasheet don't include spaces in the value it displays as binary so this was almost certainly just a typo reading 1000100. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-6-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 3cf8a0b4478c..b39c25409c23 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), -- 2.35.1

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[PATCH AUTOSEL 5.15 10/41] ASoC: cs42l52: Correct TLV for Bypass Volume
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91e90c712fade0b69cdff7cc6512f6099bd18ae5 ] The Bypass Volume is accidentally using a -6dB minimum TLV rather than the correct -60dB minimum. Add a new TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-5-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 785caba3f653..c19ad3c24702 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,6 +137,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 09/41] ASoC: cs53l30: Correct number of volume levels on SX controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 7fbd6dd68127927e844912a16741016d432a0737 ] This driver specified the maximum value rather than the number of volume levels on the SX controls, this is incorrect, so correct them. Reported-by: David Rhodes <david.rhodes(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index f2087bd38dbc..c2912ad3851b 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -348,22 +348,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 08/41] ASoC: cs35l36: Update digital volume TLV
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 5005a2345825eb8346546d99bfe669f73111b5c5 ] The digital volume TLV specifies the step as 0.25dB but the actual step of the control is 0.125dB. Update the TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l36.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index d83c1b318c1c..0accdb45ed72 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 07/41] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 80161151b3f2..785caba3f653 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,7 +137,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -364,7 +364,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 02/41] ASoC: nau8822: Add operation for internal PLL off and on
by Sasha Levin 14 Jun '22

14 Jun '22

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit aeca8a3295022bcec46697f16e098140423d8463 ] We tried to enable the audio on an imx6sx EVB with the codec nau8822, after setting the internal PLL fractional parameters, the audio still couldn't work and the there was no sdma irq at all. After checking with the section "8.1.1 Phase Locked Loop (PLL) Design Example" of "NAU88C22 Datasheet Rev 0.6", we found we need to turn off the PLL before programming fractional parameters and turn on the PLL after programming. After this change, the audio driver could record and play sound and the sdma's irq is triggered when playing or recording. Cc: David Lin <ctlin0(a)nuvoton.com> Cc: John Hsu <kchsu0(a)nuvoton.com> Cc: Seven Li <wtli(a)nuvoton.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20220530040151.95221-2-hui.wang@canonical.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8822.c | 4 ++++ sound/soc/codecs/nau8822.h | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 58123390c7a3..b436e532993d 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -740,6 +740,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, pll_param->pre_factor); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | @@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 27/43] ALSA: hda/realtek - Add HW8326 support
by Sasha Levin 14 Jun '22

14 Jun '22

From: huangwenhui <huangwenhuia(a)uniontech.com> [ Upstream commit 527f4643e03c298c1e3321cfa27866b1374a55e1 ] Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> Link: https://lore.kernel.org/r/20220608082357.26898-1-huangwenhuia@uniontech.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..b7e5032b61c9 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -660,6 +660,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 112ecc256b14..7e7e517d32c6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10092,6 +10104,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11542,6 +11555,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 18/43] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ] Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0582585236a2..93677b9f3a68 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -328,7 +328,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.35.1

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