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

[PATCH v2] ASoC: hdmi-codec: Add option for ELD bypass
by Sugar Zhang 14 May '22

14 May '22

This patch allow users to enable "ELD Bypass" who don't care anything from EDID Link Data. Currently, this driver gets ELD(from EDID) to constraint channels and rates. Unfortunately, EDID is not always valid, maybe caused by the fragile HDMI port or cable, in this situation, the max features are limited to 48kHz stereo. So, add this option to allow user to select the manual way to output audio as expected. such as multi-channels LPCM(7.1), or HBR bitstream for these sink devices. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Changes in v2: - Use MACRO SOC_SINGLE_BOOL_EXT to simplify code. Fix event_missing checked by mixer-test. Add suffix "Switch" for "ELD Bypass". sound/soc/codecs/hdmi-codec.c | 32 ++++++++++++++++++++++++++++++-- 1 file changed, 30 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index b07607a..be46fbd 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -275,6 +275,7 @@ struct hdmi_codec_priv { unsigned int chmap_idx; struct mutex lock; bool busy; + bool eld_bypass; struct snd_soc_jack *jack; unsigned int jack_status; u8 iec_status[AES_IEC958_STATUS_SIZE]; @@ -427,6 +428,31 @@ static int hdmi_codec_iec958_mask_get(struct snd_kcontrol *kcontrol, return 0; } +static int hdmi_codec_eld_bypass_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = hcp->eld_bypass; + + return 0; +} + +static int hdmi_codec_eld_bypass_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + if (hcp->eld_bypass == ucontrol->value.integer.value[0]) + return 0; + + hcp->eld_bypass = ucontrol->value.integer.value[0]; + + return 1; +} + static int hdmi_codec_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -447,7 +473,7 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, goto err; } - if (tx && hcp->hcd.ops->get_eld) { + if (tx && !hcp->eld_bypass && hcp->hcd.ops->get_eld) { ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->hcd.data, hcp->eld, sizeof(hcp->eld)); if (ret) @@ -770,6 +796,8 @@ static struct snd_kcontrol_new hdmi_codec_controls[] = { .info = hdmi_eld_ctl_info, .get = hdmi_eld_ctl_get, }, + SOC_SINGLE_BOOL_EXT("ELD Bypass Switch", 0, + hdmi_codec_eld_bypass_get, hdmi_codec_eld_bypass_put), }; static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, @@ -854,7 +882,7 @@ static void plugged_cb(struct device *dev, bool plugged) struct hdmi_codec_priv *hcp = dev_get_drvdata(dev); if (plugged) { - if (hcp->hcd.ops->get_eld) { + if (!hcp->eld_bypass && hcp->hcd.ops->get_eld) { hcp->hcd.ops->get_eld(dev->parent, hcp->hcd.data, hcp->eld, sizeof(hcp->eld)); } -- 2.7.4

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Add "Pro Audio" profile for UMC204HD
by GitHub pull_request - opened 14 May '22

14 May '22

alsa-project/alsa-ucm-conf pull request #153 was opened from brndd: Follow-up to #128 I figured that some users might prefer to keep the channel mapping untouched from what the card defaults to when unconfigured. Providing an alternative profile like this (implementation adapted from #145) for these users seems like a good idea. I believe PipeWire autogenerates profiles like these for unconfigured devices, so that's where the "Pro Audio" nomenclature comes from. Likely a good idea to match that since PipeWire is increasingly the default on most distributions. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/153 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/153.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] sound: sound_core: changed spaces to tabs
by Paul Lemmermann 14 May '22

14 May '22

changed spaces to tabs in sound_core.c according to kernel styling guidelines. found using checkpatch.pl. Signed-off-by: Paul Lemmermann <thepaulodoom(a)thepaulodoom.com> --- sound/sound_core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 3332fe321..28bc038dc 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -290,7 +290,7 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati * completed the removal before their file operations become * invalid. */ - + static void sound_remove_unit(struct sound_unit **list, int unit) { struct sound_unit *p; @@ -395,12 +395,12 @@ int register_sound_special_device(const struct file_operations *fops, int unit, name = "admmidi"; break; default: - { + { __unknown: sprintf(_name, "unknown%d", chain); - if (unit >= SOUND_STEP) - strcat(_name, "-"); - name = _name; + if (unit >= SOUND_STEP) + strcat(_name, "-"); + name = _name; } break; } -- 2.35.1

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Re: [PATCH 19/32] afs: Use mem_to_flex_dup() with struct afs_acl
by David Howells 13 May '22

13 May '22

Kees Cook <keescook(a)chromium.org> wrote: > struct afs_acl { > - u32 size; > - u8 data[]; > + DECLARE_FLEX_ARRAY_ELEMENTS_COUNT(u32, size); > + DECLARE_FLEX_ARRAY_ELEMENTS(u8, data); > }; Oof... That's really quite unpleasant syntax. Is it not possible to have mem_to_flex_dup() and friends work without that? You are telling them the fields they have to fill in. > + struct afs_acl *acl = NULL; > > - acl = kmalloc(sizeof(*acl) + size, GFP_KERNEL); > - if (!acl) { > + if (mem_to_flex_dup(&acl, buffer, size, GFP_KERNEL)) { Please don't do that. Either do: acl = mem_to_flex_dup(buffer, size, GFP_KERNEL); if (!acl) or: acl = mem_to_flex_dup(buffer, size, GFP_KERNEL); if (IS_ERR(acl)) Please especially don't make it that an apparent 'true' return indicates an error. If you absolutely must return the acl pointer through the argument list (presumably because it's actually a macro), make it return false on failure: if (!mem_to_flex_dup(&acl, buffer, size, GFP_KERNEL)) or return and explicitly check for an error code: if (mem_to_flex_dup(&acl, buffer, size, GFP_KERNEL) < 0) or: ret = mem_to_flex_dup(&acl, buffer, size, GFP_KERNEL); if (ret < 0) (or use != 0 rather than < 0) David

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[PATCH 1/2 v2] ASoC: ADAU7118: add bindings for managing pins drive strength
by Dylan Laduranty 13 May '22

13 May '22

This allows users to change SDATA and both PDM clocks pins drive strength during device probing according to their need. Signed-off-by: Dylan Laduranty <dylan.laduranty(a)mesotic.com> --- Changes since v1: split new bindings and documentation into separate patches sound/soc/codecs/adau7118.c | 62 ++++++++++++++++++++++++++++++++++--- 1 file changed, 58 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c index 841229dcbca1..18c1f246f911 100644 --- a/sound/soc/codecs/adau7118.c +++ b/sound/soc/codecs/adau7118.c @@ -29,6 +29,12 @@ FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) #define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) #define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_DS_PDM_CLK0_MASK GENMASK(1, 0) +#define ADAU7118_DS_PDM_CLK0(x) FIELD_PREP(ADAU7118_DS_PDM_CLK0_MASK, x) +#define ADAU7118_DS_PDM_CLK1_MASK GENMASK(3, 2) +#define ADAU7118_DS_PDM_CLK1(x) FIELD_PREP(ADAU7118_DS_PDM_CLK1_MASK, x) +#define ADAU7118_DS_SDATA_MASK GENMASK(5, 4) +#define ADAU7118_DS_SDATA(x) FIELD_PREP(ADAU7118_DS_SDATA_MASK, x) #define ADAU7118_FULL_SOFT_R_MASK BIT(1) #define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) @@ -489,7 +495,7 @@ static int adau7118_regulator_setup(struct adau7118_data *st) static int adau7118_parset_dt(const struct adau7118_data *st) { int ret; - u32 dec_ratio = 0; + u32 val32 = 0; /* 4 inputs */ u32 clk_map[4], regval; @@ -497,9 +503,9 @@ static int adau7118_parset_dt(const struct adau7118_data *st) return 0; ret = device_property_read_u32(st->dev, "adi,decimation-ratio", - &dec_ratio); + &val32); if (!ret) { - switch (dec_ratio) { + switch (val32) { case 64: regval = ADAU7118_DEC_RATIO(0); break; @@ -510,7 +516,7 @@ static int adau7118_parset_dt(const struct adau7118_data *st) regval = ADAU7118_DEC_RATIO(2); break; default: - dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + dev_err(st->dev, "Invalid dec ratio: %u", val32); return -EINVAL; } @@ -537,6 +543,54 @@ static int adau7118_parset_dt(const struct adau7118_data *st) return ret; } + ret = device_property_read_u32(st->dev, "adi,pdm-clk0-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid pdm-clk0-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_PDM_CLK0_MASK, + ADAU7118_DS_PDM_CLK0(val32)); + if (ret) + return ret; + } + + ret = device_property_read_u32(st->dev, "adi,pdm-clk1-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid pdm-clk1-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_PDM_CLK1_MASK, + ADAU7118_DS_PDM_CLK1(val32)); + if (ret) + return ret; + } + + ret = device_property_read_u32(st->dev, "adi,sdata-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid sdata-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_SDATA_MASK, + ADAU7118_DS_SDATA(val32)); + if (ret) + return ret; + } + return 0; } -- 2.17.1

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[PATCH v2] ALSA: hda/realtek: Add quirk for TongFang devices with pop noise
by Werner Sembach 13 May '22

13 May '22

When audio stops playing there is an audible "pop"-noise when using headphones on the TongFang GMxMRxx, GKxNRxx, GMxZGxx, GMxTGxx and GMxAGxx. This quirk fixes this mostly. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: stable(a)vger.kernel.org --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cf531c1efa132..8003bf7441d6d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9290,6 +9290,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_SET_COEF_DEFAULTS), + SND_PCI_QUIRK(0x1d05, 0x1096, "TongFang GMxMRxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1100, "TongFang GKxNRxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1111, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1119, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1129, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1147, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x115c, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), -- 2.25.1

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Re: [PATCH 00/32] Introduce flexible array struct memcpy() helpers
by David Howells 12 May '22

12 May '22

Kees Cook <keescook(a)chromium.org> wrote: > I'm happy to also point out that the conversions (patches 5+) are actually > a net reduction in lines of code: > 49 files changed, 154 insertions(+), 244 deletions(-) That doesn't mean that it's actually code that's clearer to read. I would say that it's actually less clear. In a bunch of places, you've done something like: - e = kmalloc(...); - if (!e) + if (__mem_to_flex_dup(&e, ...)) The problem is that, to me at least, it looks like: - e = kmalloc(...); - if (kmalloc failed) + if (__mem_to_flex_dup(&e, ...) succeeded) David

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[PATCH] ALSA: hda/intel: Add quirk for TongFang devices with pop noise
by Werner Sembach 12 May '22

12 May '22

When audio stops playing and sometimes when it starts playing, there is an audible "pop" noise when using headphones on most Tongfang GMxMxxx, GKxNxxx, GMxZxxx, GMxTxxx, and GMxAxxx devices. Disabling power saving for the Realtek codec fixes this noise. Presumably it is triggered on some power event in the audio circuit. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: stable(a)vger.kernel.org --- sound/pci/hda/hda_intel.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0a83eb6b88b1f..8a1088e057ec0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2201,6 +2201,13 @@ static const struct snd_pci_quirk power_save_denylist[] = { SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugs.launchpad.net/bugs/1821663 */ SND_PCI_QUIRK(0x1631, 0xe017, "Packard Bell NEC IMEDIA 5204", 0), + /* Several TongFang barebones making popping sounds */ + SND_PCI_QUIRK(0x1d05, 0x1096, "TongFang GMxMxxx", 0), + SND_PCI_QUIRK(0x1d05, 0x1100, "TongFang GKxNxxx", 0), + SND_PCI_QUIRK(0x1d05, 0x1111, "TongFang GMxZxxx", 0), + SND_PCI_QUIRK(0x1d05, 0x1129, "TongFang GMxZxxx", 0), + SND_PCI_QUIRK(0x1d05, 0x1147, "TongFang GMxTxxx", 0), + SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAxxx", 0), {} }; #endif /* CONFIG_PM */ -- 2.25.1

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[PATCH -next] ASoC: SOF: amd: add missing platform_device_unregister in acp_pci_rn_probe
by Zheng Bin 12 May '22

12 May '22

acp_pci_rn_probe misses a call platform_device_unregister in error path, this patch fixes that. Signed-off-by: Zheng Bin <zhengbin13(a)huawei.com> --- sound/soc/sof/amd/pci-rn.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index b8910bb7f27c..d5d9bcc2c997 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -101,6 +101,7 @@ static int acp_pci_rn_probe(struct pci_dev *pci, const struct pci_device_id *pci res = devm_kzalloc(&pci->dev, sizeof(struct resource) * ARRAY_SIZE(renoir_res), GFP_KERNEL); if (!res) { sof_pci_remove(pci); + platform_device_unregister(dmic_dev); return -ENOMEM; } -- 2.31.1

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[PATCH] ASoC: samsung: Fix refcount leak in aries_audio_probe
by Miaoqian Lin 12 May '22

12 May '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when done. If extcon_find_edev_by_node() fails, it doesn't call of_node_put() Calling of_node_put() after extcon_find_edev_by_node() to fix this. Fixes: 7a3a7671fa6c ("ASoC: samsung: Add driver for Aries boards") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/samsung/aries_wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c index 5265e546b124..83acbe57b248 100644 --- a/sound/soc/samsung/aries_wm8994.c +++ b/sound/soc/samsung/aries_wm8994.c @@ -585,10 +585,10 @@ static int aries_audio_probe(struct platform_device *pdev) extcon_np = of_parse_phandle(np, "extcon", 0); priv->usb_extcon = extcon_find_edev_by_node(extcon_np); + of_node_put(extcon_np); if (IS_ERR(priv->usb_extcon)) return dev_err_probe(dev, PTR_ERR(priv->usb_extcon), "Failed to get extcon device"); - of_node_put(extcon_np); priv->adc = devm_iio_channel_get(dev, "headset-detect"); if (IS_ERR(priv->adc)) -- 2.25.1

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