January 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: da7219: Fix pole orientation detection on OMTP headsets when playing music
by David Rau 09 Feb '23

09 Feb '23

The OMTP pin define headsets can be mis-detected as line out instead of OMTP, causing obvious issues with audio quality. This patch is to put increased resistances within the device at a suitable point. To solve this issue better, the new mechanism setup ground switches with conditional delay control and these allow for more stabile detection process to operate as intended. This conditional delay control will not impact the hardware process but use extra system resource. This commit improves control of ground switches in the AAD logic. Signed-off-by: David Rau <david.rau.zg(a)renesas.com> --- sound/soc/codecs/da7219-aad.c | 42 ++++++++++++++++++++++++++++++----- sound/soc/codecs/da7219-aad.h | 1 + 2 files changed, 37 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index bba73c44c219..08200ec259f9 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -352,9 +352,14 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 events[DA7219_AAD_IRQ_REG_MAX]; - u8 statusa; + u8 statusa, srm_st; int i, report = 0, mask = 0; + srm_st = snd_soc_component_read(component, DA7219_PLL_SRM_STS) & DA7219_PLL_SRM_STS_MCLK; + msleep(da7219_aad->gnd_switch_delay * ((srm_st == 0x0) ? 2 : 1) - 4); + /* Enable ground switch */ + snd_soc_component_update_bits(component, 0xFB, 0x01, 0x01); + /* Read current IRQ events */ regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, events, DA7219_AAD_IRQ_REG_MAX); @@ -454,8 +459,8 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) snd_soc_dapm_disable_pin(dapm, "Mic Bias"); snd_soc_dapm_sync(dapm); - /* Enable ground switch */ - snd_soc_component_update_bits(component, 0xFB, 0x01, 0x01); + /* Disable ground switch */ + snd_soc_component_update_bits(component, 0xFB, 0x01, 0x00); } } @@ -831,6 +836,32 @@ static void da7219_aad_handle_pdata(struct snd_soc_component *component) } } +static void da7219_aad_handle_gnd_switch_time(struct snd_soc_component *component) +{ + struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); + struct da7219_aad_priv *da7219_aad = da7219->aad; + u8 jack_det; + + jack_det = snd_soc_component_read(component, DA7219_ACCDET_CONFIG_2) + & DA7219_JACK_DETECT_RATE_MASK; + switch (jack_det) { + case 0x00: + da7219_aad->gnd_switch_delay = 32; + break; + case 0x10: + da7219_aad->gnd_switch_delay = 64; + break; + case 0x20: + da7219_aad->gnd_switch_delay = 128; + break; + case 0x30: + da7219_aad->gnd_switch_delay = 256; + break; + default: + da7219_aad->gnd_switch_delay = 32; + break; + } +} /* * Suspend/Resume @@ -908,9 +939,6 @@ int da7219_aad_init(struct snd_soc_component *component) snd_soc_component_update_bits(component, DA7219_ACCDET_CONFIG_1, DA7219_BUTTON_CONFIG_MASK, 0); - /* Enable ground switch */ - snd_soc_component_update_bits(component, 0xFB, 0x01, 0x01); - INIT_WORK(&da7219_aad->btn_det_work, da7219_aad_btn_det_work); INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work); @@ -928,6 +956,8 @@ int da7219_aad_init(struct snd_soc_component *component) regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A, &mask, DA7219_AAD_IRQ_REG_MAX); + da7219_aad_handle_gnd_switch_time(component); + return 0; } EXPORT_SYMBOL_GPL(da7219_aad_init); diff --git a/sound/soc/codecs/da7219-aad.h b/sound/soc/codecs/da7219-aad.h index f48a12012ef3..21fdf53095cc 100644 --- a/sound/soc/codecs/da7219-aad.h +++ b/sound/soc/codecs/da7219-aad.h @@ -187,6 +187,7 @@ enum da7219_aad_event_regs { struct da7219_aad_priv { struct snd_soc_component *component; int irq; + int gnd_switch_delay; u8 micbias_pulse_lvl; u32 micbias_pulse_time; -- 2.17.1

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Re: Firewire Phase24 Motu help
by Takashi Sakamoto 07 Feb '23

07 Feb '23

Hi, (C.C.ed to administrator of alsa-devel mailing list) On Mon, Jan 30, 2023 at 12:40:06PM +0100, M. Armsby wrote: > Hi Takashi, > I'm writing direct because I cannot seem to get a post on AlsaDevel > list. I've realized your two posts, but no reaction since they have no message body: * Firewire ALSA Motu and Phase24 * https://lore.kernel.org/alsa-devel/sokos1-r6xj933yn42y-hmy3gsrfgsma-mnwe9a-… * Firewire Motu ALSA (Re-post) * https://lore.kernel.org/alsa-devel/v6inx6-fwmr0r-kvajcd-kfs8u9l6m2qn2cnogd-… Jaroslav, would I ask your opinion for the problem which perhaps dropped the message body from his messages? > I need to repair my two Phase X24 (defekt caps?) and I'm having > problems with Traveler and other Motu interfaces on Linux. > Thanks for any hints or where I should post. > Here's a copy of my post: > Hello everyone! > Thank you for all the hard work on this. > Especially Mr. Takashi Sakamoto. > I'm an experienced studio engineer and was a Beta tester for Samplitude > for years. > Not a programmer and have very little experience with Linux. > So please excuse me blundering in on this professional programmer forum > but I'm hoping my experience will help somehow to find bugs. I'll > provide any information I can or test following instructions. > I really would love to completely change over to Linux but I need > reliable drivers. > How can I help? > Here's my experience so far. > Around November I started testing Linux for audio as I have > occasionally over the last years. I was pleased to find Manjaro and > Sparky would detect Traveler Mk1 and Mk3 and to run so efficiently > compared to Win10. > But things would freeze. > Reaper has a long standing Midi bug so it was difficult at first to see > why my laptop would freeze on occasion. I dissabled hardware midi in > the software. > Then I updated Sparky beginning of December and to my surprise Firewire > didn't work anymore. The Firewire stack must have been removed. > I use: > Dell Laptops 4310, 6510, M4800 > Tested platforms Ardour and Reaper using Pianoteq 8 > No JACK > Manjaro Kernel from 6.0 rc4? and 6.1.1 work for a limited time. > Sparky only works with special kernels Xanmod1 6.1.7-x64v2 but the > audio is completely distorted. > CachyOS Kernel 6.2 rc4 sounds good. > Common problem is freezing or loosing audio. Ardour is less sensitive > than Reaper but Reaper is far more efficient on CPU > (20% compared to 90%and more with one instance of Pianoteq 7 or 8) > Motu Traveler MK3 and MK1 wrong channel assignment from 1+2 (3+4 are > actually 1+2) and freezing or just loosing audio. > Motu 896Mk3 Hybrid not recognised in firewire or USB mode. > Shame cause its the best. > Motu 8pre working similarly. > Phase 24 working much better but still freezes or looses audio after > long period. > Tested in Ardour and Reaper. > I hope that's not too much at once. > Martin Armsby > M. Armsby Thanks for the report including many issues. I apologize to trouble you. In my opinion, the issues are not so easy since each application uses ALSA PCM device by slightly different ways. So we need to investigate the issue step by step to take relevant longer time than easy issues. Before discussing about the issue, I need to ask a help about MOTU devices. As you realized, 896 mk3 is neither supported yet, FireWire only nor Hybrid, in IEEE 1394 bus nor Universal Serial Bus. For example, you can see no entry for the device in module device table[1] in the file which consists of snd-firewire-motu driver. The reason is apparent; no one provides information enough to support it. Then I would like you to help me to integrate for it in the case in IEEE 1394 bus. I maintain a remote repository for my collection of configuration ROM image which includes identity information of each node in IEEE 1394 bus[2]. The information is useful both to make the entry of module device table and to make hardware database[3] in systemd project. When 896 mk3 Hybrid is connected to IEEE 1394 bus and only connected to the bus, Linux FireWire subsystem detects it as '/dev/fw1'. Then the rom image is exported to an node on sysfs. You can retrieve the content by the command line below: $ cat /sys/bus/firewire/devices/fw1/config_rom > motu-896mk3hybrid.img If retrieved successfully, please send it to me. I don't mind to receive it to private message, mailing list, any upload services. [1] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/sou… [2] https://github.com/takaswie/am-config-roms [3] https://github.com/systemd/systemd/blob/main/hwdb.d/80-ieee1394-unit-functi… Thanks Takashi Sakamoto

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[PATCH 00/19] Add soundwire support for Pink Sardine platform
by Vijendar Mukunda 06 Feb '23

06 Feb '23

Pink Sardine(ps) platform is based on ACP6.3 Architecture. ACP6.3 IP has two soundwire controller instance support. This patchset add support for Soundwire controller on Pink Sardine platform. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Vijendar Mukunda (19): ASoC: amd: ps: create platform devices based on acp config soundwire: amd: Add support for AMD Master driver soundwire: amd: register sdw controller dai ops soundwire: amd: enable build for AMD soundwire master driver soundwire: amd: add soundwire interrupt handling ASoC: amd: ps: add support for soundwire interrupts in acp pci driver ASoC: amd: ps: add soundwire dma driver for pink sardine platform ASoC: amd: ps: add soundwire dma driver dma ops ASoC: amd: ps: add support for Soundwire DMA interrupts ASoC: amd: ps: enable Soundwire DMA driver build ASoC: amd: update comments in Kconfig file ASoC: amd: ps: Add soundwire specific checks in pci driver in pm ops. ASoC: amd: ps: add support for runtime pm ops for soundwire dma driver soundwire: amd: add runtime pm ops for AMD master driver soundwire: amd: add startup and shutdown dai ops soundwire: amd: handle wake enable interrupt soundwire: amd: add pm_prepare callback and pm ops support ASoC: amd: ps: implement system level pm ops for soundwire dma driver ASoC: amd: ps: increase runtime suspend delay drivers/soundwire/Kconfig | 9 + drivers/soundwire/Makefile | 4 + drivers/soundwire/amd_master.c | 1734 +++++++++++++++++++++++++++++ drivers/soundwire/amd_master.h | 284 +++++ include/linux/soundwire/sdw_amd.h | 65 ++ sound/soc/amd/Kconfig | 3 +- sound/soc/amd/ps/Makefile | 2 + sound/soc/amd/ps/acp63.h | 98 +- sound/soc/amd/ps/pci-ps.c | 383 ++++++- sound/soc/amd/ps/ps-sdw-dma.c | 728 ++++++++++++ 10 files changed, 3287 insertions(+), 23 deletions(-) create mode 100644 drivers/soundwire/amd_master.c create mode 100644 drivers/soundwire/amd_master.h create mode 100644 include/linux/soundwire/sdw_amd.h create mode 100644 sound/soc/amd/ps/ps-sdw-dma.c -- 2.34.1

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[PATCH][next] ASoC: uapi: Replace zero-length arrays with __DECLARE_FLEX_ARRAY() helper
by Gustavo A. R. Silva 03 Feb '23

03 Feb '23

Zero-length arrays are deprecated and we are moving towards adopting C99 flexible-array members, instead. So, replace zero-length arrays declarations in anonymous union with the new __DECLARE_FLEX_ARRAY() helper macro. This helper allows for flexible-array members in unions. Link: https://github.com/KSPP/linux/issues/193 Link: https://github.com/KSPP/linux/issues/227 Link: https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- include/uapi/sound/asoc.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 053949287ce8..dd8ad790fc15 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -226,9 +226,9 @@ struct snd_soc_tplg_vendor_array { __le32 type; /* SND_SOC_TPLG_TUPLE_TYPE_ */ __le32 num_elems; /* number of elements in array */ union { - struct snd_soc_tplg_vendor_uuid_elem uuid[0]; - struct snd_soc_tplg_vendor_value_elem value[0]; - struct snd_soc_tplg_vendor_string_elem string[0]; + __DECLARE_FLEX_ARRAY(struct snd_soc_tplg_vendor_uuid_elem, uuid); + __DECLARE_FLEX_ARRAY(struct snd_soc_tplg_vendor_value_elem, value); + __DECLARE_FLEX_ARRAY(struct snd_soc_tplg_vendor_string_elem, string); }; } __attribute__((packed)); -- 2.34.1

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Is there any way to install alsa-lib with the suid bit set?
by Stefan Alexe 03 Feb '23

03 Feb '23

I am trying to only use a static /dev. I have opened an issue about this on GitHub and got it to work only as the root user. This resembles my experience with xorg. In that case I had to use the suid use flag. Is there an equivalent in this situation?

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[PATCH v3 0/6] Fix default DMIC gain on AMD PDM drivers
by Mario Limonciello 02 Feb '23

02 Feb '23

It's been reported that a number of laptops have a low volume level from the digital microphone compared to Windows. AMD offers a register that can adjust the gain for PDM which is not configured at maximum gain by default. To fix this change the default for all 3 drivers to raise the gain but also offer a module parameter. The module parameter can be used for debugging if the gain is too high on a given laptop. This is intentionally split into multiple patches for default and parameter so that if the default really does behave better universally we can bring it back to stable too later. v2->v3: * Use clamp and clear properly Mario Limonciello (6): ASoC: amd: yc: Adjust the gain for PDM DMIC ASoC: amd: yc: Add a module parameter to influence pdm_gain ASoC: amd: renoir: Adjust the gain for PDM DMIC ASoC: amd: renoir: Add a module parameter to influence pdm_gain ASoC: amd: ps: Adjust the gain for PDM DMIC ASoC: amd: ps: Add a module parameter to influence pdm_gain sound/soc/amd/ps/acp63.h | 2 +- sound/soc/amd/ps/ps-pdm-dma.c | 8 +++++++- sound/soc/amd/renoir/acp3x-pdm-dma.c | 8 +++++++- sound/soc/amd/renoir/rn_acp3x.h | 2 +- sound/soc/amd/yc/acp6x-pdm-dma.c | 8 +++++++- sound/soc/amd/yc/acp6x.h | 2 +- 6 files changed, 24 insertions(+), 6 deletions(-) -- 2.34.1

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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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[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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