May 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: hda/cs_dsp_ctl: Use private_free for control cleanup
by Richard Fitzgerald 08 May '24

08 May '24

Use the control private_free callback to free the associated data block. This ensures that the memory won't leak, whatever way the control gets destroyed. The original implementation didn't actually remove the ALSA controls in hda_cs_dsp_control_remove(). It only freed the internal tracking structure. This meant it was possible to remove/unload the amp driver while leaving its ALSA controls still present in the soundcard. Obviously attempting to access them could cause segfaults or at least dereferencing stale pointers. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 3233b978af23 ("ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls") --- This replaces my earlier "keep it simple" attempt at cleanup. I decided that implementing private_free is the only sensible way to fix the cleanup. --- sound/pci/hda/hda_cs_dsp_ctl.c | 47 ++++++++++++++++++++++------------ 1 file changed, 31 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/hda_cs_dsp_ctl.c b/sound/pci/hda/hda_cs_dsp_ctl.c index 463ca06036bf..9db45d7c17e5 100644 --- a/sound/pci/hda/hda_cs_dsp_ctl.c +++ b/sound/pci/hda/hda_cs_dsp_ctl.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <sound/soc.h> +#include <linux/cleanup.h> #include <linux/firmware/cirrus/cs_dsp.h> #include <linux/firmware/cirrus/wmfw.h> #include "hda_cs_dsp_ctl.h" @@ -97,11 +98,23 @@ static unsigned int wmfw_convert_flags(unsigned int in) return out; } -static void hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char *name) +static void hda_cs_dsp_free_kcontrol(struct snd_kcontrol *kctl) { + struct hda_cs_dsp_coeff_ctl *ctl = (struct hda_cs_dsp_coeff_ctl *)snd_kcontrol_chip(kctl); struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl; + + /* NULL priv to prevent a double-free in hda_cs_dsp_control_remove() */ + cs_ctl->priv = NULL; + kfree(ctl); +} + +static void hda_cs_dsp_add_kcontrol(struct cs_dsp_coeff_ctl *cs_ctl, + const struct hda_cs_dsp_ctl_info *info, + const char *name) +{ struct snd_kcontrol_new kcontrol = {0}; struct snd_kcontrol *kctl; + struct hda_cs_dsp_coeff_ctl *ctl __free(kfree) = NULL; int ret = 0; if (cs_ctl->len > ADSP_MAX_STD_CTRL_SIZE) { @@ -110,6 +123,13 @@ static void hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char return; } + ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); + if (!ctl) + return; + + ctl->cs_ctl = cs_ctl; + ctl->card = info->card; + kcontrol.name = name; kcontrol.info = hda_cs_dsp_coeff_info; kcontrol.iface = SNDRV_CTL_ELEM_IFACE_MIXER; @@ -117,20 +137,22 @@ static void hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char kcontrol.get = hda_cs_dsp_coeff_get; kcontrol.put = hda_cs_dsp_coeff_put; - /* Save ctl inside private_data, ctl is owned by cs_dsp, - * and will be freed when cs_dsp removes the control */ kctl = snd_ctl_new1(&kcontrol, (void *)ctl); if (!kctl) return; - ret = snd_ctl_add(ctl->card, kctl); + kctl->private_free = hda_cs_dsp_free_kcontrol; + ctl->kctl = kctl; + + /* snd_ctl_add() calls our private_free on error, which will kfree(ctl) */ + cs_ctl->priv = no_free_ptr(ctl); + ret = snd_ctl_add(info->card, kctl); if (ret) { dev_err(cs_ctl->dsp->dev, "Failed to add KControl %s = %d\n", kcontrol.name, ret); return; } dev_dbg(cs_ctl->dsp->dev, "Added KControl: %s\n", kcontrol.name); - ctl->kctl = kctl; } static void hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, @@ -138,7 +160,6 @@ static void hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, { struct cs_dsp *cs_dsp = cs_ctl->dsp; char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - struct hda_cs_dsp_coeff_ctl *ctl; const char *region_name; int ret; @@ -163,15 +184,7 @@ static void hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, " %.*s", cs_ctl->subname_len - skip, cs_ctl->subname + skip); } - ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); - if (!ctl) - return; - - ctl->cs_ctl = cs_ctl; - ctl->card = info->card; - cs_ctl->priv = ctl; - - hda_cs_dsp_add_kcontrol(ctl, name); + hda_cs_dsp_add_kcontrol(cs_ctl, info, name); } void hda_cs_dsp_add_controls(struct cs_dsp *dsp, const struct hda_cs_dsp_ctl_info *info) @@ -203,7 +216,9 @@ void hda_cs_dsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) { struct hda_cs_dsp_coeff_ctl *ctl = cs_ctl->priv; - kfree(ctl); + /* ctl and kctl may already have been removed by ALSA private_free */ + if (ctl && ctl->kctl) + snd_ctl_remove(ctl->card, ctl->kctl); } EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_control_remove, SND_HDA_CS_DSP_CONTROLS); -- 2.39.2

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[PATCH v1 1/1] ASoC: atmel: tse850-pcm5142: Remove unused of_gpio.h
by Andy Shevchenko 08 May '24

08 May '24

of_gpio.h is deprecated and subject to remove. The driver doesn't use it, simply remove the unused header. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/atmel/tse850-pcm5142.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 611da23325d3..5d208e0b4b90 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -35,10 +35,9 @@ // of the (filtered) output from the PCM5142 codec. #include <linux/clk.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> -#include <linux/of_gpio.h> #include <linux/regulator/consumer.h> #include <sound/soc.h> -- 2.43.0.rc1.1336.g36b5255a03ac

2 1

[PATCH v2 0/4] ASoC: qcom: x1e80100: Correct channel mapping
by Krzysztof Kozlowski 08 May '24

08 May '24

Hi, First patch is a build dependency. Changes in v2: - Re-work most of the idea according to Srini comments: set channel mapping for backend DAIs, not frontend. - Patch #1: no changes - Patch #2 is entirely replaced - now channel mapping is implemented in q6apm-lpass-dais. - Patch #3: rework to new approach, but most of the code stays. - Patch #4: rework significantly, because only backend DAIs is now affected. - Link to v1: https://lore.kernel.org/r/20240507-asoc-x1e80100-4-channel-mapping-v1-0-b12… Description =========== X1E80100 CRD is the first board, which comes with four speakers, so we still keep fixing and adding missing pieces. The board has speaker arranged as left front+back and then right front+back. Using default channel mapping causes front right speaker to play left back stream. Adjust the channel maps for frontend DAIs to fix stereo and four-channel playback. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: Constify channel mapping array arguments in set_channel_map() ASoC: qcom: q6apm-lpass-dais: Implement proper channel mapping ASoC: qcom: qdsp6: Set channel mapping instead of fixed defaults ASoC: qcom: x1e80100: Correct channel mapping include/sound/cs35l41.h | 4 ++-- include/sound/soc-dai.h | 8 ++++---- sound/soc/codecs/adau7118.c | 6 ++++-- sound/soc/codecs/cs35l41-lib.c | 4 ++-- sound/soc/codecs/cs35l41.c | 3 ++- sound/soc/codecs/max98504.c | 6 ++++-- sound/soc/codecs/wcd9335.c | 6 ++++-- sound/soc/codecs/wcd934x.c | 6 ++++-- sound/soc/qcom/qdsp6/audioreach.c | 30 +++++++----------------------- sound/soc/qcom/qdsp6/audioreach.h | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 16 ++++++++++------ sound/soc/qcom/qdsp6/q6apm-dai.c | 2 ++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 21 +++++++++++++-------- sound/soc/qcom/x1e80100.c | 18 ++++++++++++++++++ sound/soc/soc-dai.c | 4 ++-- 15 files changed, 79 insertions(+), 57 deletions(-) --- base-commit: 2b84edefcad14934796fad37b16512b6a2ca467e change-id: 20240507-asoc-x1e80100-4-channel-mapping-ea5f02b9e678 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

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[PATCH v1] ALSA: hda/realtek: Add support for ASUS Zenbook 2024 HN7306W
by Stefan Binding 08 May '24

08 May '24

This laptop uses CS35L41 HDA with 2 amps using I2C and Internal Boost. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.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 8502b30d5d5d..8c157158f1cb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10276,6 +10276,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALC285_FIXUP_CS35L56_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALC285_FIXUP_CS35L56_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1ed3, "ASUS HN7306W", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), -- 2.34.1

2 1

[PATCH 0/4] ASoC: Constify static snd_pcm_hardware
by Krzysztof Kozlowski 08 May '24

08 May '24

No dependencies. Static 'struct snd_pcm_hardware' is not modified by few drivers and its copy is passed to the core, so it can be made const for increased code safety. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: qcom: Constify static snd_pcm_hardware ASoC: fsl: Constify static snd_pcm_hardware ASoC: meson: Constify static snd_pcm_hardware ASoC: uniphier: Constify static snd_pcm_hardware sound/soc/fsl/imx-pcm-rpmsg.c | 2 +- sound/soc/meson/aiu-fifo-i2s.c | 2 +- sound/soc/meson/aiu-fifo-spdif.c | 2 +- sound/soc/meson/aiu-fifo.h | 2 +- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/qcom/qdsp6/q6apm-dai.c | 4 ++-- sound/soc/qcom/qdsp6/q6asm-dai.c | 2 +- sound/soc/uniphier/aio-dma.c | 2 +- 8 files changed, 9 insertions(+), 9 deletions(-) --- base-commit: ebbaba914053d8bcadd7d64eb8d2555299509a27 change-id: 20240429-n-asoc-const-snd-pcm-hardware-f0a5ade885ca Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

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[PATCH AUTOSEL 6.1 01/25] ASoC: Intel: bytcr_rt5640: Apply Asus T100TA quirk to Asus T100TAM too
by Sasha Levin 08 May '24

08 May '24

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit e50729d742ec364895f1c389c32315984a987aa5 ] The Asus T100TA quirk has been using an exact match on a product-name of "T100TA" but there are also T100TAM variants with a slightly higher clocked CPU and a metal backside which need the same quirk. Sort the existing T100TA (stereo speakers) below the more specific T100TAF (mono speaker) quirk and switch from exact matching to substring matching so that the T100TA quirk will also match on the T100TAM models. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://msgid.link/r/20240407191559.21596-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 094445036c20f..d6ef8e850412b 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -636,28 +636,30 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_USE_AMCR0F28), }, { + /* Asus T100TAF, unlike other T100TA* models this one has a mono speaker */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { + /* Asus T100TA and T100TAM, must come after T100TAF (mono spk) match */ .matches = { - DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | - BYT_RT5640_MONO_SPEAKER | - BYT_RT5640_DIFF_MIC | - BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { -- 2.43.0

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[PATCH AUTOSEL 6.6 01/43] ASoC: Intel: bytcr_rt5640: Apply Asus T100TA quirk to Asus T100TAM too
by Sasha Levin 08 May '24

08 May '24

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit e50729d742ec364895f1c389c32315984a987aa5 ] The Asus T100TA quirk has been using an exact match on a product-name of "T100TA" but there are also T100TAM variants with a slightly higher clocked CPU and a metal backside which need the same quirk. Sort the existing T100TA (stereo speakers) below the more specific T100TAF (mono speaker) quirk and switch from exact matching to substring matching so that the T100TA quirk will also match on the T100TAM models. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://msgid.link/r/20240407191559.21596-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index e609249cc38d5..651408c6f399d 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -636,28 +636,30 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_USE_AMCR0F28), }, { + /* Asus T100TAF, unlike other T100TA* models this one has a mono speaker */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { + /* Asus T100TA and T100TAM, must come after T100TAF (mono spk) match */ .matches = { - DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | - BYT_RT5640_MONO_SPEAKER | - BYT_RT5640_DIFF_MIC | - BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { -- 2.43.0

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[PATCH AUTOSEL 6.8 01/52] ASoC: Intel: bytcr_rt5640: Apply Asus T100TA quirk to Asus T100TAM too
by Sasha Levin 08 May '24

08 May '24

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit e50729d742ec364895f1c389c32315984a987aa5 ] The Asus T100TA quirk has been using an exact match on a product-name of "T100TA" but there are also T100TAM variants with a slightly higher clocked CPU and a metal backside which need the same quirk. Sort the existing T100TA (stereo speakers) below the more specific T100TAF (mono speaker) quirk and switch from exact matching to substring matching so that the T100TA quirk will also match on the T100TAM models. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://msgid.link/r/20240407191559.21596-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 05f38d1f7d824..b41a1147f1c34 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -636,28 +636,30 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_USE_AMCR0F28), }, { + /* Asus T100TAF, unlike other T100TA* models this one has a mono speaker */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { + /* Asus T100TA and T100TAM, must come after T100TAF (mono spk) match */ .matches = { - DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | - BYT_RT5640_MONO_SPEAKER | - BYT_RT5640_DIFF_MIC | - BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { -- 2.43.0

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[PATCH 0/4] ASoC: qcom: x1e80100: Correct channel mapping
by Krzysztof Kozlowski 07 May '24

07 May '24

Hi, First patch is a build dependency. Description =========== X1E80100 CRD is the first board, which comes with four speakers, so we still keep fixing and adding missing pieces. The board has speaker arranged as left front+back and then right front+back. Using default channel mapping causes front right speaker to play left back stream. Adjust the channel maps for frontend DAIs to fix stereo and four-channel playback. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: Constify channel mapping array arguments in set_channel_map() ASoC: qcom: q6dsp: Implement proper channel mapping in Audioreach ASoC: qcom: q6dsp: Set channel mapping instead of fixed defaults ASoC: qcom: x1e80100: Correct channel mapping include/sound/cs35l41.h | 4 ++-- include/sound/soc-dai.h | 8 +++---- sound/soc/codecs/adau7118.c | 6 ++++-- sound/soc/codecs/cs35l41-lib.c | 4 ++-- sound/soc/codecs/cs35l41.c | 3 ++- sound/soc/codecs/max98504.c | 6 ++++-- sound/soc/codecs/wcd9335.c | 6 ++++-- sound/soc/codecs/wcd934x.c | 6 ++++-- sound/soc/qcom/qdsp6/audioreach.c | 14 +++++-------- sound/soc/qcom/qdsp6/audioreach.h | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 16 ++++++++------ sound/soc/qcom/qdsp6/q6apm-dai.c | 12 +++++++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 6 ++++-- sound/soc/qcom/qdsp6/q6apm.c | 28 ++++++++++++++++++++++++- sound/soc/qcom/qdsp6/q6apm.h | 8 +++++++ sound/soc/qcom/qdsp6/topology.c | 12 +++++++++++ sound/soc/qcom/x1e80100.c | 37 +++++++++++++++++++++++++++++++-- sound/soc/soc-dai.c | 4 ++-- 18 files changed, 142 insertions(+), 39 deletions(-) --- base-commit: c5e512ffe106f751c61e5a036560f522e58eadcd change-id: 20240507-asoc-x1e80100-4-channel-mapping-ea5f02b9e678 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

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[PATCH 0/3] ASoC: samsung: midas-audio: Add GPIO-based headset jack detection
by Artur Weber 07 May '24

07 May '24

Many of Samsung's Exynos 4 devices share the same midas-audio driver to handle the codec setup. While most of these devices, including the Midas itself, use the jack detection provided by the WM8994 driver, other devices such as the Samsung Galaxy Tab 3 8.0 (lt01) use two GPIOs and an ADC channel to determine jack insertion, the jack's type, and button presses (for headsets with volume up/down/play buttons). In the downstream kernel, this behavior is implemented in the sec-jack driver[1], and the per-device settings are configured in *-jack.c files in the mach folder (see e.g. the Tab 3's implementation[2]). This patchset implements this mechanism in the midas_wm1811.c driver, and adds new DTS options to allow for its configuration. It also enables jack detection for the Samsung Galaxy Tab 3 8.0. A very similar mechanism was already present in the aries_wm8994.c driver[3]; this implementation heavily borrows from it, though there are a few extra cleanups as well. Signed-off-by: Artur Weber <aweber.kernel(a)gmail.com> [1] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [2] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [3] https://github.com/torvalds/linux/blob/master/sound/soc/samsung/aries_wm899… --- Artur Weber (3): ASoC: dt-bindings: samsung,midas-audio: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Add GPIO-based headset jack detection ARM: dts: samsung: exynos4212-tab3: Fix headset mic, add jack detection .../bindings/sound/samsung,midas-audio.yaml | 30 +++ arch/arm/boot/dts/samsung/exynos4212-tab3.dtsi | 23 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/midas_wm1811.c | 265 ++++++++++++++++++++- 4 files changed, 305 insertions(+), 15 deletions(-) --- base-commit: e67572cd2204894179d89bd7b984072f19313b03 change-id: 20240502-midas-wm1811-gpio-jack-b10226b17ecc Best regards, -- Artur Weber <aweber.kernel(a)gmail.com>

2 8

[PATCH v2] ASoC: Intel: avs: boards: Properly name input device
by Amadeusz Sławiński 07 May '24

07 May '24

Machine boards expose input device for use with userspace. Current name in some cases is incorrect, fix it. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- v2: - Rebase on top of correct tree. --- sound/soc/intel/avs/boards/es8336.c | 2 +- sound/soc/intel/avs/boards/nau8825.c | 2 +- sound/soc/intel/avs/boards/rt274.c | 3 ++- sound/soc/intel/avs/boards/rt286.c | 4 ++-- sound/soc/intel/avs/boards/rt298.c | 4 ++-- 5 files changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c index ff3bd15132694..f4919a6abc278 100644 --- a/sound/soc/intel/avs/boards/es8336.c +++ b/sound/soc/intel/avs/boards/es8336.c @@ -113,7 +113,7 @@ static int avs_es8336_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, &data->jack, pins, num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index 3164745b15167..eef718c6b8ad9 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -96,7 +96,7 @@ static int avs_nau8825_codec_init(struct snd_soc_pcm_runtime *runtime) * 4 buttons here map to the google Reference headset. * The use of these buttons can be decided by the user space. */ - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 | + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, jack, pins, num_pins); if (ret) diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c index e3aa28780df5a..51ebc790c6b4d 100644 --- a/sound/soc/intel/avs/boards/rt274.c +++ b/sound/soc/intel/avs/boards/rt274.c @@ -102,7 +102,8 @@ static int avs_rt274_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET, jack, pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET, jack, pins, + num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c index d24316fc4db9c..67fd362962c40 100644 --- a/sound/soc/intel/avs/boards/rt286.c +++ b/sound/soc/intel/avs/boards/rt286.c @@ -63,8 +63,8 @@ static int avs_rt286_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, - pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, + jack, pins, num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c index 7fd28544f7869..1003e8689eacc 100644 --- a/sound/soc/intel/avs/boards/rt298.c +++ b/sound/soc/intel/avs/boards/rt298.c @@ -74,8 +74,8 @@ static int avs_rt298_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, - pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, + jack, pins, num_pins); if (ret) return ret; -- 2.34.1

2 1

[PATCH 0/4] ASoC: use 'time_left' instead of 'timeout' with wait_for_*() functions
by Wolfram Sang 07 May '24

07 May '24

There is a confusing pattern in the kernel to use a variable named 'timeout' to store the result of wait_for_*() functions causing patterns like: timeout = wait_for_completion_timeout(...) if (!timeout) return -ETIMEDOUT; with all kinds of permutations. Use 'time_left' as a variable to make the code obvious and self explaining. This is part of a tree-wide series. The rest of the patches can be found here (some parts may still be WIP): git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git i2c/time_left Because these patches are generated, I audit them before sending. This is why I will send series step by step. Build bot is happy with these patches, though. No functional changes intended. Wolfram Sang (4): ASoC: codecs: wm8962: use 'time_left' variable with wait_for_completion_timeout() ASoC: codecs: wm8993: use 'time_left' variable with wait_for_completion_timeout() ASoC: codecs: wm8994: use 'time_left' variable with wait_for_completion_timeout() ASoC: codecs: wm8996: use 'time_left' variable with wait_for_completion_timeout() sound/soc/codecs/wm8962.c | 12 ++++++------ sound/soc/codecs/wm8993.c | 12 ++++++------ sound/soc/codecs/wm8994.c | 8 ++++---- sound/soc/codecs/wm8996.c | 14 +++++++------- 4 files changed, 23 insertions(+), 23 deletions(-) -- 2.43.0

3 6

[PATCH 00/13] ASoC: Use snd_soc_substream_to_rtd() for accessing private_data
by Krzysztof Kozlowski 07 May '24

07 May '24

Hi, Do not open-code snd_soc_substream_to_rtd() when accessing snd_pcm_substream->private_data. This makes code more consistent with rest of ASoC and allows in the future to move the field to any other place or add additional checks in snd_soc_substream_to_rtd(). Best regards, Krzysztof --- Krzysztof Kozlowski (13): ASoC: qcom: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: tegra: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: ti: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: arm: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: amd: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: fsl: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: img: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: kirkwood: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: loongson: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: mediatek: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: meson: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: samsung: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: sunxi: Use snd_soc_substream_to_rtd() for accessing private_data sound/arm/pxa2xx-pcm-lib.c | 4 ++-- sound/soc/amd/acp/acp-mach-common.c | 2 +- sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/fsl/imx-card.c | 6 +++--- sound/soc/fsl/imx-hdmi.c | 2 +- sound/soc/fsl/imx-pcm-rpmsg.c | 6 +++--- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/loongson/loongson_card.c | 2 +- sound/soc/loongson/loongson_dma.c | 2 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 4 ++-- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 14 +++++++------- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 2 +- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 8 ++++---- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 6 +++--- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 10 +++++----- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 4 ++-- sound/soc/meson/aiu-fifo.c | 2 +- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/qcom/apq8016_sbc.c | 4 ++-- sound/soc/qcom/qdsp6/q6apm-dai.c | 2 +- sound/soc/qcom/sc7180.c | 10 +++++----- sound/soc/qcom/sc7280.c | 12 ++++++------ sound/soc/qcom/sc8280xp.c | 8 ++++---- sound/soc/qcom/sdw.c | 8 ++++---- sound/soc/qcom/sm8250.c | 10 +++++----- sound/soc/qcom/x1e80100.c | 8 ++++---- sound/soc/samsung/midas_wm1811.c | 2 +- sound/soc/sunxi/sun50i-dmic.c | 2 +- sound/soc/tegra/tegra_asoc_machine.c | 2 +- sound/soc/tegra/tegra_pcm.c | 6 +++--- sound/soc/ti/omap-hdmi.c | 2 +- 35 files changed, 82 insertions(+), 82 deletions(-) --- base-commit: 82415cf72c7e224be7a6496f3a53c0b365c2fe9d change-id: 20240430-asoc-snd-substream-clean-924b717d8f54 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

3 15

[PATCH] ASoC: qcom: audioreach: Correct mapping of back speakers
by Krzysztof Kozlowski 07 May '24

07 May '24

Qualcomm DSP, according to downstream sources, expects back speakers to be mapped as "back", not "surround". The surround is used only for 8+ speakers configuration. Reported-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Fixes: 3c5fcb20e07e ("ASoC: qcom: audioreach: Add 4 channel support") Cc: <stable(a)vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/qcom/qdsp6/audioreach.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c index 5291deac0a0b..c655f0213723 100644 --- a/sound/soc/qcom/qdsp6/audioreach.c +++ b/sound/soc/qcom/qdsp6/audioreach.c @@ -277,8 +277,8 @@ static void audioreach_set_channel_mapping(u8 *ch_map, int num_channels) } else if (num_channels == 4) { ch_map[0] = PCM_CHANNEL_FL; ch_map[1] = PCM_CHANNEL_FR; - ch_map[2] = PCM_CHANNEL_LS; - ch_map[3] = PCM_CHANNEL_RS; + ch_map[2] = PCM_CHANNEL_LB; + ch_map[3] = PCM_CHANNEL_RB; } } @@ -851,8 +851,8 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph, } else if (num_channels == 4) { media_format->channel_mapping[0] = PCM_CHANNEL_FL; media_format->channel_mapping[1] = PCM_CHANNEL_FR; - media_format->channel_mapping[2] = PCM_CHANNEL_LS; - media_format->channel_mapping[3] = PCM_CHANNEL_RS; + media_format->channel_mapping[2] = PCM_CHANNEL_LB; + media_format->channel_mapping[3] = PCM_CHANNEL_RB; } rc = q6apm_send_cmd_sync(graph->apm, pkt, 0); -- 2.43.0

1 1

[PATCH v3 0/7] ASoC: codecs: wcd937x: add wcd937x audio codec support
by Mohammad Rafi Shaik 07 May '24

07 May '24

This patchset adds support for Qualcomm WCD9370/WCD9375 codec. Qualcomm WCD9370/WCD9375 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 3 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 6 DMICs and MBHC. For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on QCM6490 IDP device. Changes since v2: - Used common qcom,wcd93xx-common.yaml. removed duplicate properties. - Merged bindings patches "v2-0001" and "v2-0003" in single patch for easy review. - Fixed dt binding check errors. - Added missing "qcom,wcd9375-codec" in v3-0001 dt binding patch. - Added constraints on values in v3-0001 binding patch as suggested by Krzysztof - Fix the typo mistake in v2 cover letter Changes since v1: - Split the patch per driver for easier review as suggested by Krzysztof - Used devm_gpiod_get api to get reset gpio as suggested by Krzysztof Prasad Kumpatla (7): ASoC: dt-bindings: document wcd937x Audio Codec ASoC: codecs: wcd937x: add wcd937x codec driver ASoC: codecs: wcd937x-sdw: add SoundWire driver ASoC: codecs: wcd937x: add basic controls ASoC: codecs: wcd937x: add playback dapm widgets ASoC: codecs: wcd937x: add capture dapm widgets ASoC: codecs: wcd937x: add audio routing and Kconfig .../bindings/sound/qcom,wcd937x-sdw.yaml | 78 + .../bindings/sound/qcom,wcd937x.yaml | 58 + sound/soc/codecs/Kconfig | 20 + sound/soc/codecs/Makefile | 7 + sound/soc/codecs/wcd937x-sdw.c | 1147 +++++++ sound/soc/codecs/wcd937x.c | 3036 +++++++++++++++++ sound/soc/codecs/wcd937x.h | 653 ++++ 7 files changed, 4999 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml create mode 100644 sound/soc/codecs/wcd937x-sdw.c create mode 100644 sound/soc/codecs/wcd937x.c create mode 100644 sound/soc/codecs/wcd937x.h base-commit: 5e4f84f18c4ee9b0ccdc19e39b7de41df21699dd -- 2.25.1

4 13

[PATCH v1] slimbus: qcom-ngd-ctrl: Add stream disable support
by Viken Dadhaniya 07 May '24

07 May '24

Currently slimbus driver doesn't support stream disable callback, it only supports stream enable callback. In slimbus usecase, client is switching to new frequency with same channel and calling enable stream callback for new frequency but DSP subsystem is crashing as we are switching to new frequency with same channel without disabling stream for older frequency. Ideally, before switching to another frequency, client should call disable stream callback and then enable stream for newer frequency. Hence add support to disable stream via qcom_slim_ngd_disable_stream(). Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 70 +++++++++++++++++++++++++++++++++ drivers/slimbus/slimbus.h | 13 ++++++ 2 files changed, 83 insertions(+) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index e0b21f0f79c1..d952827d2e12 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1,6 +1,7 @@ // SPDX-License-Identifier: GPL-2.0 // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved. // Copyright (c) 2018, Linaro Limited +// Copyright (c) 2024 Qualcomm Innovation Center, Inc. All rights reserved. #include <linux/irq.h> #include <linux/kernel.h> @@ -1084,6 +1085,74 @@ static int qcom_slim_ngd_enable_stream(struct slim_stream_runtime *rt) return ret; } +static int qcom_slim_ngd_disable_stream(struct slim_stream_runtime *rt) +{ + struct slim_device *sdev = rt->dev; + struct slim_controller *ctrl = sdev->ctrl; + struct slim_val_inf msg = {0}; + u8 wbuf[SLIM_MSGQ_BUF_LEN]; + u8 rbuf[SLIM_MSGQ_BUF_LEN]; + struct slim_msg_txn txn = {0,}; + int i, ret; + + txn.mt = SLIM_MSG_MT_DEST_REFERRED_USER; + txn.dt = SLIM_MSG_DEST_LOGICALADDR; + txn.la = SLIM_LA_MGR; + txn.ec = 0; + txn.msg = &msg; + txn.msg->num_bytes = 0; + txn.msg->wbuf = wbuf; + txn.msg->rbuf = rbuf; + + for (i = 0; i < rt->num_ports; i++) { + struct slim_port *port = &rt->ports[i]; + + if (txn.msg->num_bytes == 0) { + wbuf[txn.msg->num_bytes++] = (u8)(SLIM_CH_REMOVE << 6) + | (sdev->laddr & 0x1f); + + ret = slim_alloc_txn_tid(ctrl, &txn); + if (ret) { + dev_err(&sdev->dev, "Fail to allocate TID\n"); + return -ENXIO; + } + wbuf[txn.msg->num_bytes++] = txn.tid; + } + wbuf[txn.msg->num_bytes++] = port->ch.id; + } + + txn.mc = SLIM_USR_MC_CHAN_CTRL; + txn.rl = txn.msg->num_bytes + 4; + ret = qcom_slim_ngd_xfer_msg_sync(ctrl, &txn); + if (ret) { + slim_free_txn_tid(ctrl, &txn); + dev_err(&sdev->dev, "TX timed out:MC:0x%x,mt:0x%x ret:%d\n", + txn.mc, txn.mt, ret); + return ret; + } + + txn.mc = SLIM_USR_MC_RECONFIG_NOW; + txn.msg->num_bytes = 2; + wbuf[1] = sdev->laddr; + txn.rl = txn.msg->num_bytes + 4; + + ret = slim_alloc_txn_tid(ctrl, &txn); + if (ret) { + dev_err(ctrl->dev, "Fail to allocate TID ret:%d\n", ret); + return ret; + } + + wbuf[0] = txn.tid; + ret = qcom_slim_ngd_xfer_msg_sync(ctrl, &txn); + if (ret) { + slim_free_txn_tid(ctrl, &txn); + dev_err(&sdev->dev, "TX timed out:MC:0x%x,mt:0x%x ret:%d\n", + txn.mc, txn.mt, ret); + } + + return ret; +} + static int qcom_slim_ngd_get_laddr(struct slim_controller *ctrl, struct slim_eaddr *ea, u8 *laddr) { @@ -1642,6 +1711,7 @@ static int qcom_slim_ngd_ctrl_probe(struct platform_device *pdev) ctrl->ctrl.clkgear = SLIM_MAX_CLK_GEAR; ctrl->ctrl.get_laddr = qcom_slim_ngd_get_laddr; ctrl->ctrl.enable_stream = qcom_slim_ngd_enable_stream; + ctrl->ctrl.disable_stream = qcom_slim_ngd_disable_stream; ctrl->ctrl.xfer_msg = qcom_slim_ngd_xfer_msg; ctrl->ctrl.wakeup = NULL; ctrl->state = QCOM_SLIM_NGD_CTRL_DOWN; diff --git a/drivers/slimbus/slimbus.h b/drivers/slimbus/slimbus.h index 00a7f112574b..21543d125614 100644 --- a/drivers/slimbus/slimbus.h +++ b/drivers/slimbus/slimbus.h @@ -1,6 +1,7 @@ /* SPDX-License-Identifier: GPL-2.0 */ /* * Copyright (c) 2011-2017, The Linux Foundation + * Copyright (c) 2024 Qualcomm Innovation Center, Inc. All rights reserved. */ #ifndef _DRIVERS_SLIMBUS_H @@ -316,6 +317,18 @@ enum slim_transport_protocol { SLIM_PROTO_EXT_HALF_DUP, }; +/* + * enum slim_ch_control: Channel control. + * Activate will schedule channel and/or group of channels in the TDM frame. + * Suspend will keep the schedule but data-transfer won't happen. + * Remove will remove the channel/group from the TDM frame. + */ +enum slim_ch_control { + SLIM_CH_ACTIVATE, + SLIM_CH_SUSPEND, + SLIM_CH_REMOVE, +}; + /** * struct slim_stream_runtime - SLIMbus stream runtime instance * -- 2.17.1

2 1

[PATCH] [RFC PATCH] ALSA: usb-audio: endpoint: Prevent NULL pointer deference in snd_usb_endpoint_close
by Rajashekar kuruva 07 May '24

07 May '24

When multiple plug-in and plug-out events occur, there is a risk of encountering a NULL pointer dereference leading to a kernel panic during a headset use-case. this issue arises in the snd_usb_endpoint_close function To avoid check if ep->iface_ref is not null before decrementing its opened count. If ep->iface_ref is null, we skip the decrement and the subsequent logic. Signed-off-by: Rajashekar kuruva <quic_kuruva(a)quicinc.com> --- sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 8f65349a06d3..0e3101b7e392 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -950,7 +950,7 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, usb_audio_dbg(chip, "Closing EP 0x%x (count %d)\n", ep->ep_num, ep->opened); - if (!--ep->iface_ref->opened && + if (ep->iface_ref && !--ep->iface_ref->opened && !(chip->quirk_flags & QUIRK_FLAG_IFACE_SKIP_CLOSE)) endpoint_set_interface(chip, ep, false); -- 2.25.1

3 9

[PATCH] ASoC: qcom: q6apm-dai: drop unused 'q6apm_dai_rtd' fields
by Krzysztof Kozlowski 07 May '24

07 May '24

Remove few unused fields from 'struct q6apm_dai_rtd'. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/qcom/qdsp6/q6apm-dai.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 00bbd291be5c..cee23d4bda6f 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -70,14 +70,10 @@ struct q6apm_dai_rtd { unsigned int bytes_received; unsigned int copied_total; uint16_t bits_per_sample; - uint16_t source; /* Encoding source bit mask */ - uint16_t session_id; bool next_track; enum stream_state state; struct q6apm_graph *graph; spinlock_t lock; - uint32_t initial_samples_drop; - uint32_t trailing_samples_drop; bool notify_on_drain; }; @@ -720,14 +716,12 @@ static int q6apm_dai_compr_set_metadata(struct snd_soc_component *component, switch (metadata->key) { case SNDRV_COMPRESS_ENCODER_PADDING: - prtd->trailing_samples_drop = metadata->value[0]; q6apm_remove_trailing_silence(component->dev, prtd->graph, - prtd->trailing_samples_drop); + metadata->value[0]); break; case SNDRV_COMPRESS_ENCODER_DELAY: - prtd->initial_samples_drop = metadata->value[0]; q6apm_remove_initial_silence(component->dev, prtd->graph, - prtd->initial_samples_drop); + metadata->value[0]); break; default: ret = -EINVAL; -- 2.43.0

2 1

[PATCH v2] ASoC: tas2781: Fix a warning reported by robot kernel test
by Shenghao Ding 07 May '24

07 May '24

Fix a warning reported by robot kernel test that 'fw_entry' in function 'tas2781_load_calibration' is used uninitialized with compiler sh4-linux-gcc (GCC) 13.2.0, an update of copyright and a correction of the comments. Fixes: ef3bcde75d06 ("ASoc: tas2781: Add tas2781 driver") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Describe more what was actually an issue reported by robot kernel test. - Correct the subject suffix, ALSA: ASoc/tas2781 --> ASoC: tas2781. v1: - Update the copyright year to 2024 - tasdevice-fmw.c --> tas2781-fmwlib.c - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202405021200.YHInjV43-lkp@intel.com/ --- sound/soc/codecs/tas2781-fmwlib.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 45760fe19523..a6be81adcb83 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1,8 +1,8 @@ // SPDX-License-Identifier: GPL-2.0 // -// tasdevice-fmw.c -- TASDEVICE firmware support +// tas2781-fmwlib.c -- TASDEVICE firmware support // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -1878,7 +1878,7 @@ int tas2781_load_calibration(void *context, char *file_name, { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; struct tasdevice *tasdev = &(tas_priv->tasdevice[i]); - const struct firmware *fw_entry; + const struct firmware *fw_entry = NULL; struct tasdevice_fw *tas_fmw; struct firmware fmw; int offset = 0; -- 2.34.1

3 2

[PATCH 0/7] ASoC: SOF: Intel: remove circular dependency for SoundWire BPT
by Pierre-Louis Bossart 07 May '24

07 May '24

The SoundWire BPT support will rely on the HDaudio DMA. This exposes a circular dependency module dependency which has to be resolved by splitting common parts used by HDaudio and SoundWire parts, and 'generic' parts used by HDaudio only. This patchset does not change any functionality, it just moves code around, exposes symbols that are used in the new module. The code has been in use for more than one kernel cycle already so it really shouldn't break any existing platforms. The main issue with such code moves is that it makes backports or fixes more complicated. That's the main reason why we held back these patches until we were reasonably confident on the maturity of MTL and LNL drivers. Pierre-Louis Bossart (7): ASoC: SOF: Intel: hda-stream: export stream_get_position() helper ASoC: SOF: Intel: regroup all SoundWire/Intel functions in hda.c ASoC: SOF: Intel: start splitting top-level from common parts ASoC: SOF: Intel: move common code from hda.c ASoC: SOF: Intel: move tracepoint creation ASoC: SOF: Intel: remove circular dependency on hda_sdw_process_wakeen() ASoC: SOF: Intel: move hda.c to different module sound/soc/sof/intel/Kconfig | 25 +- sound/soc/sof/intel/Makefile | 23 +- sound/soc/sof/intel/apl.c | 3 - sound/soc/sof/intel/cnl.c | 15 +- sound/soc/sof/intel/hda-bus.c | 2 + sound/soc/sof/intel/hda-common-ops.c | 1 + sound/soc/sof/intel/hda-ctrl.c | 10 + sound/soc/sof/intel/hda-dai.c | 6 + sound/soc/sof/intel/hda-dsp.c | 502 ++++++++++++++++++++++ sound/soc/sof/intel/hda-ipc.c | 115 +++++ sound/soc/sof/intel/hda-loader.c | 45 +- sound/soc/sof/intel/hda-pcm.c | 6 + sound/soc/sof/intel/hda-probes.c | 2 + sound/soc/sof/intel/hda-stream.c | 13 + sound/soc/sof/intel/hda-trace.c | 3 + sound/soc/sof/intel/hda.c | 605 +++------------------------ sound/soc/sof/intel/hda.h | 15 + sound/soc/sof/intel/icl.c | 4 +- sound/soc/sof/intel/lnl.c | 4 +- sound/soc/sof/intel/mtl.c | 21 +- sound/soc/sof/intel/pci-apl.c | 1 + sound/soc/sof/intel/pci-cnl.c | 1 + sound/soc/sof/intel/pci-icl.c | 2 + sound/soc/sof/intel/pci-lnl.c | 3 + sound/soc/sof/intel/pci-mtl.c | 1 + sound/soc/sof/intel/pci-skl.c | 1 + sound/soc/sof/intel/pci-tgl.c | 2 + sound/soc/sof/intel/shim.h | 1 + sound/soc/sof/intel/telemetry.c | 1 + sound/soc/sof/intel/tgl.c | 10 +- sound/soc/sof/intel/tracepoints.c | 5 + 31 files changed, 811 insertions(+), 637 deletions(-) create mode 100644 sound/soc/sof/intel/tracepoints.c -- 2.40.1

2 8

[PATCH 0/3] ASoC: SOF: Intel: HDA/DMIC updates
by Pierre-Louis Bossart 07 May '24

07 May '24

The first patch handles a problematic configuration where the wrong machine driver/topology is used: when the hardware reports an external HDaudio codec the direction is to ignore/discard ACPI SoundWire devices. The last two patch deal with DMIC format configurations and allow users to select S16_LE even if the DMIC and internal copiers only support 24 or 32-bits. The code changes are located in sound/soc/sof/ but in the scope of Intel DAIs. Peter Ujfalusi (2): ASoC: SOF: ipc4-topology: Allow selective update in sof_ipc4_update_hw_params ASoC: SOF: ipc4-topology: Correct DAI copier config and NHLT blob request Pierre-Louis Bossart (1): ASoC: SOF: Intel: discard SoundWire configuration if HDaudio codec is reported sound/soc/sof/intel/hda.c | 9 +- sound/soc/sof/intel/hda.h | 2 + sound/soc/sof/ipc4-topology.c | 245 ++++++++++++++++++++++++++-------- 3 files changed, 196 insertions(+), 60 deletions(-) -- 2.40.1

2 4

[PATCH v3] ALSA: usb-audio: Add sampling rates support for Mbox3
by Manuel Barrio Linares 06 May '24

06 May '24

This adds support for all sample rates supported by the hardware,Digidesign Mbox 3 supports: {44100, 48000, 88200, 96000} Fixes syncing clock issues that presented as pops. To test this, without this patch playing 440hz tone produces pops. Clock is now synced between playback and capture interfaces so no more latency drift issue when using pipewire pro-profile. (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3900) Signed-off-by: Manuel Barrio Linares <mbarriolinares(a)gmail.com> --- v2: Cleaned up code. Explicit __le32 usage. v3: Get rid of sparse warnings sound/usb/quirks-table.h | 38 +++++++++++++-------- sound/usb/quirks.c | 74 +++++++++++++++++++++++++++++++--------- 2 files changed, 81 insertions(+), 31 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 5d72dc8..73abc38 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3013,21 +3013,28 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .fmt_bits = 24, .channels = 4, .iface = 2, .altsetting = 1, .altset_idx = 1, .attributes = 0x00, - .endpoint = 0x01, + .endpoint = USB_RECIP_INTERFACE | USB_DIR_OUT, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, - .rate_max = 48000, - .nr_rates = 1, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, .rate_table = (unsigned int[]) { - 48000 - } + 44100, 48000, 88200, 96000 + }, + .sync_ep = USB_RECIP_INTERFACE | USB_DIR_IN, + .sync_iface = 3, + .sync_altsetting = 1, + .sync_ep_idx = 1, + .implicit_fb = 1, } }, { @@ -3035,22 +3042,25 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .fmt_bits = 24, .channels = 4, .iface = 3, .altsetting = 1, .altset_idx = 1, - .endpoint = 0x81, .attributes = 0x00, + .endpoint = USB_RECIP_INTERFACE | USB_DIR_IN, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, .maxpacksize = 0x009c, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, - .rate_max = 48000, - .nr_rates = 1, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, .rate_table = (unsigned int[]) { - 48000 - } + 44100, 48000, 88200, 96000 + }, + .implicit_fb = 0, } }, { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 09712e6..2f961f0 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -984,21 +984,13 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } -static void mbox3_setup_48_24_magic(struct usb_device *dev) +static void mbox3_setup_defaults(struct usb_device *dev) { /* The Mbox 3 is "little endian" */ /* max volume is: 0x0000. */ /* min volume is: 0x0080 (shown in little endian form) */ - - /* Load 48000Hz rate into buffer */ - u8 com_buff[4] = {0x80, 0xbb, 0x00, 0x00}; - - /* Set 48000Hz sample rate */ - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x01, 0x21, 0x0100, 0x0001, &com_buff, 4); //Is this really needed? - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x01, 0x21, 0x0100, 0x8101, &com_buff, 4); + u8 com_buff[2]; /* Deactivate Tuner */ /* on = 0x01*/ @@ -1008,6 +1000,8 @@ static void mbox3_setup_48_24_magic(struct usb_device *dev) 0x01, 0x21, 0x0003, 0x2001, &com_buff, 1); /* Set clock source to Internal (as opposed to S/PDIF) */ + /* Internal = 0x01*/ + /* S/PDIF = 0x02*/ com_buff[0] = 0x01; snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 1, 0x21, 0x0100, 0x8001, &com_buff, 1); @@ -1113,9 +1107,11 @@ static void mbox3_setup_48_24_magic(struct usb_device *dev) 1, 0x21, 0x0107, 0x4201, &com_buff, 2); /* Toggle allowing host control */ + /* Not needed com_buff[0] = 0x02; snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 3, 0x21, 0x0000, 0x2001, &com_buff, 1); + */ /* Do not dim fx returns */ com_buff[0] = 0x00; @@ -1259,26 +1255,27 @@ static int snd_usb_mbox3_boot_quirk(struct usb_device *dev) descriptor_size = le16_to_cpu(get_cfg_desc(config)->wTotalLength); if (descriptor_size != MBOX3_DESCRIPTOR_SIZE) { - dev_err(&dev->dev, "Invalid descriptor size=%d.\n", descriptor_size); + dev_err(&dev->dev, "MBOX3: Invalid descriptor size=%d.\n", descriptor_size); return -ENODEV; } - dev_dbg(&dev->dev, "device initialised!\n"); + dev_dbg(&dev->dev, "MBOX3: device initialised!\n"); err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, &dev->descriptor, sizeof(dev->descriptor)); config = dev->actconfig; if (err < 0) - dev_dbg(&dev->dev, "error usb_get_descriptor: %d\n", err); + dev_dbg(&dev->dev, "MBOX3: error usb_get_descriptor: %d\n", err); err = usb_reset_configuration(dev); if (err < 0) - dev_dbg(&dev->dev, "error usb_reset_configuration: %d\n", err); - dev_dbg(&dev->dev, "mbox3_boot: new boot length = %d\n", + dev_dbg(&dev->dev, "MBOX3: error usb_reset_configuration: %d\n", err); + + dev_dbg(&dev->dev, "MBOX3: new boot length = %d\n", le16_to_cpu(get_cfg_desc(config)->wTotalLength)); - mbox3_setup_48_24_magic(dev); - dev_info(&dev->dev, "Digidesign Mbox 3: 24bit 48kHz"); + mbox3_setup_defaults(dev); + dev_info(&dev->dev, "MBOX3: Initialized."); return 0; /* Successful boot */ } @@ -1734,6 +1731,46 @@ static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs, return 0; } +static void mbox3_set_format_quirk(struct snd_usb_substream *subs, + const struct audioformat *fmt) +{ + __le32 buff4 = 0; + u8 buff1 = 0x01; + u32 new_rate = subs->data_endpoint->cur_rate; + u32 current_rate; + + // Get current rate from card and check if changing it is needed + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4); + current_rate = le32_to_cpu(buff4); + dev_dbg(&subs->dev->dev, + "MBOX3: Current configured sample rate: %d", current_rate); + if (current_rate == new_rate) { + dev_dbg(&subs->dev->dev, + "MBOX3: No change needed (current rate:%d == new rate:%d)", + current_rate, new_rate); + return; + } + + // Set new rate + dev_info(&subs->dev->dev, + "MBOX3: Changing sample rate to: %d", new_rate); + buff4 = cpu_to_le32(new_rate); + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21, 0x0100, 0x8101, &buff4, 4); + + // Set clock source to Internal + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21, 0x0100, 0x8001, &buff1, 1); + + // Check whether the change was successful + buff4 = 0; + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4); + if (new_rate != le32_to_cpu(buff4)) + dev_warn(&subs->dev->dev, "MBOX3: Couldn't set the sample rate"); +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1755,6 +1792,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x08e4, 0x0163): /* Pioneer DJM-850 */ pioneer_djm_set_format_quirk(subs, 0x0086); break; + case USB_ID(0x0dba, 0x5000): + mbox3_set_format_quirk(subs, fmt); /* Digidesign Mbox 3 */ + break; } } -- 2.45.0

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[PATCH] ASoC: Intel: avs: boards: Properly name input device
by Amadeusz Sławiński 06 May '24

06 May '24

Machine boards expose input device for use with userspace. Current name in some cases is incorrect, fix it. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/intel/avs/boards/es8336.c | 2 +- sound/soc/intel/avs/boards/nau8825.c | 2 +- sound/soc/intel/avs/boards/rt274.c | 3 ++- sound/soc/intel/avs/boards/rt286.c | 4 ++-- sound/soc/intel/avs/boards/rt298.c | 4 ++-- sound/soc/intel/avs/boards/rt5640.c | 3 ++- 6 files changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c index 3d77bc3344702..6440a46e795ec 100644 --- a/sound/soc/intel/avs/boards/es8336.c +++ b/sound/soc/intel/avs/boards/es8336.c @@ -113,7 +113,7 @@ static int avs_es8336_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, &data->jack, pins, num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index 3164745b15167..eef718c6b8ad9 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -96,7 +96,7 @@ static int avs_nau8825_codec_init(struct snd_soc_pcm_runtime *runtime) * 4 buttons here map to the google Reference headset. * The use of these buttons can be decided by the user space. */ - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 | + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, jack, pins, num_pins); if (ret) diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c index e3aa28780df5a..51ebc790c6b4d 100644 --- a/sound/soc/intel/avs/boards/rt274.c +++ b/sound/soc/intel/avs/boards/rt274.c @@ -102,7 +102,8 @@ static int avs_rt274_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET, jack, pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET, jack, pins, + num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c index d24316fc4db9c..67fd362962c40 100644 --- a/sound/soc/intel/avs/boards/rt286.c +++ b/sound/soc/intel/avs/boards/rt286.c @@ -63,8 +63,8 @@ static int avs_rt286_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, - pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, + jack, pins, num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c index 7fd28544f7869..1003e8689eacc 100644 --- a/sound/soc/intel/avs/boards/rt298.c +++ b/sound/soc/intel/avs/boards/rt298.c @@ -74,8 +74,8 @@ static int avs_rt298_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, - pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, + jack, pins, num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/rt5640.c b/sound/soc/intel/avs/boards/rt5640.c index 07cd8db4e6c2b..7c94680c4b3ad 100644 --- a/sound/soc/intel/avs/boards/rt5640.c +++ b/sound/soc/intel/avs/boards/rt5640.c @@ -62,7 +62,8 @@ static int avs_rt5640_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET, jack, pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET, jack, pins, + num_pins); if (ret) return ret; -- 2.34.1

1 0

[PATCH v3 0/3] ALSA: hda/tas2781: Add tas2781 driver for SPI.
by Baojun Xu 06 May '24

06 May '24

This patch was used to add TAS2781 devices on SPI support in sound/pci/hda. It use ACPI node descript about parameters of TAS2781 on SPI, it like: Scope (_SB.PC00.SPI0) { Device (GSPK) { Name (_HID, "TXNW2781") // _HID: Hardware ID Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { SpiSerialBusV2 (...) SpiSerialBusV2 (...) } } } } And in platform/x86/serial-multi-instantiate.c, those spi devices will be added into system as a single SPI device, so TAS2781 SPI driver will probe twice for every single SPI device. And driver will also parser mono DSP firmware binary and RCA binary for itself. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> Baojun Xu (3): ALSA: hda/tas2781: Modification for add tas2781 driver for SPI ALSA: hda/tas2781: Main code of tas2781 driver for SPI ALSA: hda/tas2781: Firmware load for tas2781 driver for SPI drivers/acpi/scan.c | 1 + .../platform/x86/serial-multi-instantiate.c | 10 + sound/pci/hda/Kconfig | 15 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_realtek.c | 13 + sound/pci/hda/tas2781-spi.h | 165 ++ sound/pci/hda/tas2781_hda_spi.c | 1305 ++++++++++ sound/pci/hda/tas2781_spi_fwlib.c | 2272 +++++++++++++++++ 8 files changed, 3783 insertions(+) create mode 100644 sound/pci/hda/tas2781-spi.h create mode 100644 sound/pci/hda/tas2781_hda_spi.c create mode 100644 sound/pci/hda/tas2781_spi_fwlib.c -- 2.40.1

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[PATCH v4 0/3] ALSA: hda/tas2781: Add tas2781 driver for SPI.
by Baojun Xu 06 May '24

06 May '24

This patch was used to add TAS2781 devices on SPI support in sound/pci/hda. It use ACPI node descript about parameters of TAS2781 on SPI, it like: Scope (_SB.PC00.SPI0) { Device (GSPK) { Name (_HID, "TXNW2781") // _HID: Hardware ID Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { SpiSerialBusV2 (...) SpiSerialBusV2 (...) } } } } And in platform/x86/serial-multi-instantiate.c, those spi devices will be added into system as a single SPI device, so TAS2781 SPI driver will probe twice for every single SPI device. And driver will also parser mono DSP firmware binary and RCA binary for itself. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> Baojun Xu (3): ALSA: hda/tas2781: Add tas2781 hda driver based on SPI ALSA: hda/tas2781: Tas2781 hda driver for SPI ALSA: hda/tas2781: Firmware load for tas2781 hda driver based on SPI drivers/acpi/scan.c | 2 + .../platform/x86/serial-multi-instantiate.c | 13 + sound/pci/hda/Kconfig | 14 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_realtek.c | 13 + sound/pci/hda/tas2781-spi.h | 149 ++ sound/pci/hda/tas2781_hda_spi.c | 1240 +++++++++ sound/pci/hda/tas2781_spi_fwlib.c | 2252 +++++++++++++++++ 8 files changed, 3685 insertions(+) create mode 100644 sound/pci/hda/tas2781-spi.h create mode 100644 sound/pci/hda/tas2781_hda_spi.c create mode 100644 sound/pci/hda/tas2781_spi_fwlib.c -- 2.40.1

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