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[PATCH] ALSA: hda: Fix the control element identification for multiple codecs
by Jaroslav Kysela 02 Feb '23

02 Feb '23

Some motherboards have multiple HDA codecs connected to the serial bus. The current code may create multiple mixer controls with the almost identical identification. The current code use id.device field from the control element structure to store the codec address to avoid such clashes for multiple codecs. Unfortunately, the user space do not handle this correctly. For mixer controls, only name and index are used for the identifiers. This patch fixes this problem to compose the index using the codec address as an offset in case, when the control already exists. It is really unlikely that one codec will create 10 similar controls. This patch adds new kernel module parameter 'ctl_dev_id' to allow select the old behaviour, too. The CONFIG_SND_HDA_CTL_DEV_ID Kconfig option sets the default value. BugLink: https://github.com/alsa-project/alsa-lib/issues/294 BugLink: https://github.com/alsa-project/alsa-lib/issues/205 Fixes: 54d174031576 ("[ALSA] hda-codec - Fix connection list parsing") Fixes: 1afe206ab699 ("ALSA: hda - Try to find an empty control index when it's occupied") Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> -- rfc..v1: - added CONFIG_SND_HDA_CTL_DEV_ID Kconfig option --- include/sound/hda_codec.h | 1 + sound/pci/hda/Kconfig | 14 ++++++++++++++ sound/pci/hda/hda_codec.c | 13 ++++++++++--- sound/pci/hda/hda_controller.c | 1 + sound/pci/hda/hda_controller.h | 1 + sound/pci/hda/hda_intel.c | 6 ++++++ 6 files changed, 33 insertions(+), 3 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index eba23daf2c29..bbb7805e85d8 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -259,6 +259,7 @@ struct hda_codec { unsigned int relaxed_resume:1; /* don't resume forcibly for jack */ unsigned int forced_resume:1; /* forced resume for jack */ unsigned int no_stream_clean_at_suspend:1; /* do not clean streams at suspend */ + unsigned int ctl_dev_id:1; /* old control element id build behaviour */ #ifdef CONFIG_PM unsigned long power_on_acct; diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 06d304db4183..886255a03e8b 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -302,6 +302,20 @@ config SND_HDA_INTEL_HDMI_SILENT_STREAM This feature can impact power consumption as resources are kept reserved both at transmitter and receiver. +config SND_HDA_CTL_DEV_ID + bool "Use the device identifier field for controls" + depends on SND_HDA_INTEL + help + Say Y to use the device identifier field for (mixer) + controls (old behaviour until this option is available). + + When enabled, the multiple HDA codecs may set the device + field in control (mixer) element identifiers. The use + of this field is not recommended and defined for mixer controls. + + The old behaviour (Y) is obsolete and will be removed. Consider + to not enable this option. + endif endmenu diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index edd653ece70d..8f20abc036bf 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3390,7 +3390,12 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, kctl = snd_ctl_new1(knew, codec); if (!kctl) return -ENOMEM; - if (addr > 0) + /* Do not use the id.device field for MIXER elements. + * This field is for real device numbers (like PCM) but codecs + * are hidden components from the user space view (unrelated + * to the mixer element identification). + */ + if (addr > 0 && codec->ctl_dev_id) kctl->id.device = addr; if (idx > 0) kctl->id.index = idx; @@ -3401,9 +3406,11 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, * the codec addr; if it still fails (or it's the * primary codec), then try another control index */ - if (!addr && codec->core.addr) + if (!addr && codec->core.addr) { addr = codec->core.addr; - else if (!idx && !knew->index) { + if (!codec->ctl_dev_id) + idx += 10 * addr; + } else if (!idx && !knew->index) { idx = find_empty_mixer_ctl_idx(codec, knew->name, 0); if (idx <= 0) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 0ff286b7b66b..083df287c1a4 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1231,6 +1231,7 @@ int azx_probe_codecs(struct azx *chip, unsigned int max_slots) continue; codec->jackpoll_interval = chip->jackpoll_interval; codec->beep_mode = chip->beep_mode; + codec->ctl_dev_id = chip->ctl_dev_id; codecs++; } } diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index f5bf295eb830..8556031bcd68 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -124,6 +124,7 @@ struct azx { /* HD codec */ int codec_probe_mask; /* copied from probe_mask option */ unsigned int beep_mode; + bool ctl_dev_id; #ifdef CONFIG_SND_HDA_PATCH_LOADER const struct firmware *fw; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 87002670c0c9..048656234280 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -119,6 +119,8 @@ static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif static bool dmic_detect = 1; +static bool ctl_dev_id[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = + CONFIG_SND_HDA_CTL_DEV_ID}; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -157,6 +159,8 @@ module_param(dmic_detect, bool, 0444); MODULE_PARM_DESC(dmic_detect, "Allow DSP driver selection (bypass this driver) " "(0=off, 1=on) (default=1); " "deprecated, use snd-intel-dspcfg.dsp_driver option instead"); +module_param_array(ctl_dev_id, bool, NULL, 0444); +MODULE_PARM_DESC(ctl_dev_id, "Use control device identifier (based on codec address)."); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2278,6 +2282,8 @@ static int azx_probe_continue(struct azx *chip) chip->beep_mode = beep_mode[dev]; #endif + chip->ctl_dev_id = ctl_dev_id[dev]; + /* create codec instances */ if (bus->codec_mask) { err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); -- 2.39.0

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Test - empty posts
by M. Armsby 02 Feb '23

02 Feb '23

1 0

[PATCH] Sound: pci/hda: Add quirk for HP Envy x360 ey0xxx
by Fae 02 Feb '23

02 Feb '23

This patch fixes speaker output on the HP Envy x360 ey0xxx The mic/speaker mute LEDs are still non-functional but they aren't required for basic funtionality Also the quirk is labeled ALC287_FIXUP_CS35L41_I2C_2 however it does not only apply to ALC287 Its function is the generic cs35l41 fixup telling the driver that there's 2 i2c speakers. It isn't actually specific to ALC287 I did have a bit of luck getting the mic mute LED to work but it was to finicky to submit ( I used the ALC245_FIXUP_HP_GPIO_LED quirk ) I had no luck getting the speaker mute LED to work Signed-off-by: Fae <faenkhauser(a)gmail.com> --- sound/pci/hda/cs35l41_hda.c | 3 ++- sound/pci/hda/patch_realtek.c | 1 + 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 129bffb431c22..21d8c2de128ff 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1156,7 +1156,8 @@ static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physd hw_cfg->valid = true; put_device(physdev); - if (strncmp(hid, "CLSA0100", 8) == 0) { + if ((strncmp(hid, "CLSA0100", 8) == 0) || + (strncmp(hid, "CSC3551", 7) == 0)) { hw_cfg->bst_type = CS35L41_EXT_BOOST_NO_VSPK_SWITCH; } else if (strncmp(hid, "CLSA0101", 8) == 0) { hw_cfg->bst_type = CS35L41_EXT_BOOST; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8a57636f622e9..e7053cbc6bb6e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9179,6 +9179,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x841c, "HP Pavilion 15-CK0xx", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x8a31, "HP ENVY x360 2-in-1 Laptop 15-ey0xxx", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), -- 2.37.1

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Re: [Bug 216859] New: PCI bridge to bus boot hang at enumeration
by Bjorn Helgaas 01 Feb '23

01 Feb '23

[+cc sound folks] On Wed, Dec 28, 2022 at 06:02:48AM -0600, Bjorn Helgaas wrote: > On Wed, Dec 28, 2022 at 08:37:52AM +0000, bugzilla-daemon(a)kernel.org wrote: > > https://bugzilla.kernel.org/show_bug.cgi?id=216859 > > > Summary: PCI bridge to bus boot hang at enumeration > > Kernel Version: 6.1-rc1 > > ... > > > With Kernel 6.1-rc1 the enumeration process stopped working for me, > > see attachments. > > > > The enumeration works fine with Kernel 6.0 and below. > > > > Same problem still exists with v6.1. and v6.2.-rc1 > > Thank you very much for your report, Zeno! > > v6.0 works, v6.1-rc1 fails. Would you mind booting v6.1-rc1 with the > "ignore_loglevel initcall_debug" kernel parameters and taking a photo > when it hangs? > > How did you conclude that the hang is related to a PCI bridge? I see > recent PCI messages in the photo, but it looks like the last message > is from NFS, so I'm wondering if I'm missing some context. The v6.0 > dmesg shows several other ntfs, fuse, JFS, etc messages before more > PCI-related things. Anyway, the "initcall_debug" might help us narrow > it down a bit. Thanks very much for the bisection (complete log at [1])! The bisection claims the first bad commit is: 833477fce7a1 ("Merge tag 'sound-6.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound") with parents: 7e6739b9336e ("Merge tag 'drm-next-2022-10-05' of git://anongit.freedesktop.org/drm/drm") 86a4d29e7554 ("Merge tag 'asoc-v6.1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus") Both 7e6739b9336e and 86a4d29e7554 tested "good" during the bisection. There is a minor conflict when merging 86a4d29e7554 into the upstream, but I can't imagine that being resolved incorrectly. Would you mind turning off CONFIG_SOUND in your .config and testing 833477fce7a1 again? I'm a little skeptical that the hang would be sound-related, but I guess it's a place to start. Bjorn [1] https://bugzilla.kernel.org/show_bug.cgi?id=216859#c35

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[PATCH] ASoC: codecs/jz4760: add digital gain controls
by Christophe Branchereau 01 Feb '23

01 Feb '23

Both the DAC and ADC have digital gain controls available for their mixers, which go from -31 to 0db by step of 1db. Signed-off-by: Christophe Branchereau <cbranchereau(a)gmail.com> --- sound/soc/codecs/jz4760.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/jz4760.c b/sound/soc/codecs/jz4760.c index d96a4f6c9183..9df58e23d360 100644 --- a/sound/soc/codecs/jz4760.c +++ b/sound/soc/codecs/jz4760.c @@ -287,6 +287,7 @@ static const DECLARE_TLV_DB_MINMAX_MUTE(dac_tlv, -3100, 100); static const DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0); static const DECLARE_TLV_DB_MINMAX(out_tlv, -2500, 100); static const DECLARE_TLV_DB_SCALE(linein_tlv, -2500, 100, 0); +static const DECLARE_TLV_DB_MINMAX(mixer_tlv, -3100, 0); /* Unconditional controls. */ static const struct snd_kcontrol_new jz4760_codec_snd_controls[] = { @@ -299,6 +300,14 @@ static const struct snd_kcontrol_new jz4760_codec_snd_controls[] = { JZ4760_CODEC_REG_GCR4, JZ4760_CODEC_REG_GCR3, REG_GCR_GAIN_OFFSET, REG_GCR_GAIN_MAX, 1, linein_tlv), + SOC_SINGLE_TLV("Mixer Capture Volume", + JZ4760_CODEC_REG_MIX1, + REG_GCR_GAIN_OFFSET, REG_GCR_GAIN_MAX, 1, mixer_tlv), + + SOC_SINGLE_TLV("Mixer Playback Volume", + JZ4760_CODEC_REG_MIX2, + REG_GCR_GAIN_OFFSET, REG_GCR_GAIN_MAX, 1, mixer_tlv), + SOC_SINGLE("High-Pass Filter Capture Switch", JZ4760_CODEC_REG_CR4, REG_CR4_ADC_HPF_OFFSET, 1, 0), -- 2.39.0

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[PATCH] ALSA: hda/realtek: Fix the speaker output on Samsung Galaxy Book2 Pro 360
by Guillaume Pinot 01 Feb '23

01 Feb '23

Samsung Galaxy Book2 Pro 360 (13" 2022 NP930QED-KA1FR) with codec SSID 144d:ca03 requires the same workaround for enabling the speaker amp like other Samsung models with ALC298 codec. Cc: <stable(a)vger.kernel.org> Signed-off-by: Guillaume Pinot <texitoi(a)texitoi.eu> --- I've tested this fix on my laptop with success. I've took "inspiration" from https://lore.kernel.org/all/20221115170235.18875-1-tiwai@suse.de/ This is my first contribution, so feel free to give me feedbacks! 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 6fab7c8fc19a..c4496206c3e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9521,6 +9521,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xca03, "Samsung Galaxy Book2 Pro 360 (NP930QED)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), -- 2.30.2

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[PATCH] ASoC: dt-bindings: irondevice, sma1303.yaml: Fix about breaking the checks
by Kiseok Jo 01 Feb '23

01 Feb '23

Fix the bindings checks like syntax error. Signed-off-by: Kiseok Jo <kiseok.jo(a)irondevice.com> Reported-by: Rob Herring <robh(a)kernel.org> --- .../bindings/sound/irondevice,sma1303.yaml | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml b/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml index eb87374cc812..162c52606635 100644 --- a/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml +++ b/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml @@ -19,14 +19,13 @@ required: - compatible - reg -additionalProperties: true +additionalProperties: false examples: - | - #include <dt-bindings/gpio/gpio.h> i2c_bus { - sma1303_amp: sma1303@1e { - compatible = "irondevice,sma1303"; - reg = <0x1e>; - }; - }; + amplifier@1e { + compatible = "irondevice,sma1303"; + reg = <0x1e>; + }; + }; base-commit: eeb6b0feb597e8fb88c4e1459323060e17350752 -- 2.20.1

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[PATCH v2] ALSA: pci: lx6464es: fix a debug loop
by Dan Carpenter 01 Feb '23

01 Feb '23

This loop accidentally reuses the "i" iterator for both the inside and the outside loop. The value of MAX_STREAM_BUFFER is 5. I believe that chip->rmh.stat_len is in the 2-12 range. If the value of .stat_len is 4 or more then it will loop exactly one time, but if it's less then it is a forever loop. It looks like it was supposed to combined into one loop where conditions are checked. Fixes: 8e6320064c33 ("ALSA: lx_core: Remove useless #if 0 .. #endif") Signed-off-by: Dan Carpenter <error27(a)gmail.com> --- v2: In the first version I just deleted the outside loop but that was not the correct fix. Combine the conditions instead. sound/pci/lx6464es/lx_core.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index d3f58a3d17fb..b5b0d43bb8dc 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -493,12 +493,11 @@ int lx_buffer_ask(struct lx6464es *chip, u32 pipe, int is_capture, dev_dbg(chip->card->dev, "CMD_08_ASK_BUFFERS: needed %d, freed %d\n", *r_needed, *r_freed); - for (i = 0; i < MAX_STREAM_BUFFER; ++i) { - for (i = 0; i != chip->rmh.stat_len; ++i) - dev_dbg(chip->card->dev, - " stat[%d]: %x, %x\n", i, - chip->rmh.stat[i], - chip->rmh.stat[i] & MASK_DATA_SIZE); + for (i = 0; i < MAX_STREAM_BUFFER && i < chip->rmh.stat_len; + ++i) { + dev_dbg(chip->card->dev, " stat[%d]: %x, %x\n", i, + chip->rmh.stat[i], + chip->rmh.stat[i] & MASK_DATA_SIZE); } } -- 2.35.1

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[PATCH] ALSA: emu10k1: clean up some inconsistent indenting
by Jiapeng Chong 01 Feb '23

01 Feb '23

No functional modification involved. sound/pci/emu10k1/memory.c:446 snd_emu10k1_synth_free() warn: inconsistent indenting. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Link: https://bugzilla.openanolis.cn/show_bug.cgi?id=3903 Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/pci/emu10k1/memory.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index edb3f1763719..6a7406014f81 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -443,7 +443,7 @@ snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *memblk) unmap_memblk(emu, blk); spin_unlock_irqrestore(&emu->memblk_lock, flags); synth_free_pages(emu, blk); - __snd_util_mem_free(hdr, memblk); + __snd_util_mem_free(hdr, memblk); mutex_unlock(&hdr->block_mutex); return 0; } -- 2.20.1.7.g153144c

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[PATCH] ASoC: cs42l42: use helper function
by Kuninori Morimoto 01 Feb '23

01 Feb '23

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current ASoC has many helper function. Link: https://lore.kernel.org/r/20230201111415.67bd4b7e@canb.auug.org.au Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/cs42l42-sdw.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l42-sdw.c b/sound/soc/codecs/cs42l42-sdw.c index 79023268d4c1..7b539ee55499 100644 --- a/sound/soc/codecs/cs42l42-sdw.c +++ b/sound/soc/codecs/cs42l42-sdw.c @@ -155,10 +155,7 @@ static int cs42l42_sdw_dai_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_str if (!sdw_stream) return 0; - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback_dma_data = sdw_stream; - else - dai->capture_dma_data = sdw_stream; + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); return 0; } -- 2.25.1

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