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[PATCH 1/2] ALSA: hda/realtek: Accelerate the front MIC charging and discharging of the 897 codec
by wangdich9700＠163.com 04 Aug '25

04 Aug '25

From: wangdicheng <wangdicheng(a)kylinos.cn> The front MIC capacitance of the Inspur board is too large, resulting in a longer charging and discharging time required, which in turn prevents recording for 5 seconds before startup. Therefore, software is used to increase the delay by 2 seconds to avoid this issue. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- 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 877137cb09ac..88675eae9447 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1001,6 +1001,14 @@ static int alc_resume(struct hda_codec *codec) if (!spec->no_depop_delay) msleep(150); /* to avoid pop noise */ codec->patch_ops.init(codec); + + if (codec->core.subsystem_id == 0x10ec1304) { + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x04); + msleep(50); + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x44); + msleep(2000); + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80); + } snd_hda_regmap_sync(codec); hda_call_check_power_status(codec, 0x01); return 0; -- 2.25.1

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[PATCH v1] ASoC: tas2781: Add keyword "init" in profile section
by Shenghao Ding 03 Aug '25

03 Aug '25

Since version 0x105, the keyword 'init' was introduced into the profile, which is used for chip initialization, particularly to store common settings for other non-initialization profiles. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- include/sound/tas2781-dsp.h | 8 ++++++++ sound/soc/codecs/tas2781-fmwlib.c | 12 ++++++++++++ sound/soc/codecs/tas2781-i2c.c | 6 ++++++ 3 files changed, 26 insertions(+) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index c3a9efa73d5d..a21f34c0266e 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -198,6 +198,14 @@ struct tasdevice_rca { int ncfgs; struct tasdevice_config_info **cfg_info; int profile_cfg_id; + /* + * Since version 0x105, the keyword 'init' was introduced into the + * profile, which is used for chip initialization, particularly to + * store common settings for other non-initialization profiles. + * if (init_profile_id < 0) + * No init profile inside the RCA firmware. + */ + int init_profile_id; }; void tasdevice_select_cfg_blk(void *context, int conf_no, diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index c9c1e608ddb7..8baf56237624 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -180,6 +180,16 @@ static struct tasdevice_config_info *tasdevice_add_config( dev_err(tas_priv->dev, "add conf: Out of boundary\n"); goto out; } + /* If in the RCA bin file are several profiles with the + * keyword "init", init_profile_id only store the last + * init profile id. + */ + if (strnstr(&config_data[config_offset], "init", 64)) { + tas_priv->rcabin.init_profile_id = + tas_priv->rcabin.ncfgs - 1; + dev_dbg(tas_priv->dev, "%s: init profile id = %d\n", + __func__, tas_priv->rcabin.init_profile_id); + } config_offset += 64; } @@ -283,6 +293,8 @@ int tasdevice_rca_parser(void *context, const struct firmware *fmw) int i; rca = &(tas_priv->rcabin); + /* Initialize to none */ + rca->init_profile_id = -1; fw_hdr = &(rca->fw_hdr); if (!fmw || !fmw->data) { dev_err(tas_priv->dev, "Failed to read %s\n", diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9f4d965a1335..f8faeffd983d 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -1641,6 +1641,12 @@ static void tasdevice_fw_ready(const struct firmware *fmw, tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; + /* Init common setting for different audio profiles */ + if (tas_priv->rcabin.init_profile_id >= 0) + tasdevice_select_cfg_blk(tas_priv, + tas_priv->rcabin.init_profile_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + #ifdef CONFIG_SND_SOC_TAS2781_ACOUST_I2C if (tas_priv->name_prefix) acoustic_debugfs_node = devm_kasprintf(tas_priv->dev, -- 2.43.0

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[PATCH REQUEST] ALSA: hda/tas2781: Add support for ASUS VivoBook 16 Flip TP3607SA
by Ludd ite 02 Aug '25

02 Aug '25

Summary ======= The ASUS VivoBook 16 Flip TP3607SA laptop contains TI TAS2781 smart amplifiers but they are not supported by the current snd_hda_scodec_tas2781_i2c driver. The driver fails to initialize with "Platform not supported" error, preventing internal speakers from working. Hardware Information ==================== Laptop Model: ASUS VivoBook 16 Flip TP3607SA_TP3607SA BIOS Version: 1.0 PCI Audio Device: Intel Lunar Lake-M HD Audio Controller [8086:a828] (rev 10) Subsystem Vendor: 0x1043 (ASUS) Subsystem Device: 0x3e30 Audio Codec: Realtek ALC294 (HDA:10ec0294,10433e30,00100004) Smart Amplifier: TI TAS2781 on I2C bus (ACPI ID: TIAS2781:00) System Information ================== OS: Ubuntu 25.04 (development) Kernel: 6.14.0-1007-realtime Audio System: PipeWire with SOF (Sound Open Firmware) SOF Firmware: intel/sof-ipc4/lnl/sof-lnl.ri (version 2.12.0.1) Topology: intel/sof-ipc4-tplg/sof-hda-generic-2ch.tplg Current Behavior ================ 1. Audio Detection: SOF firmware loads successfully and creates speaker sink in PipeWire 2. Audio Pipeline: Complete audio pipeline works (volume controls, routing, mixing) 3. TAS2781 Driver: Fails to bind with the following errors: tas2781-hda i2c-TIAS2781:00: Failed to get Speaker id. tas2781-hda i2c-TIAS2781:00: read acpi error, ret: -2 tas2781-hda i2c-TIAS2781:00: error -ENOENT: Platform not supported tas2781-hda i2c-TIAS2781:00: probe with driver tas2781-hda failed with error -2 4. Result: No sound from internal speakers (Bluetooth and HDMI audio work fine) Technical Analysis ================== - I2C Device Present: i2c-TIAS2781:00 device exists and is detected - ACPI Information: Device has ACPI ID TIAS2781: but driver cannot read required configuration - Volume Control: Audio shows active volume bars in pavucontrol, indicating software pipeline works - Driver Status: TAS2781 driver loads but doesn't bind to the device due to platform whitelist Evidence of TAS2781 Hardware ============================= I2C device exists: $ ls /sys/bus/i2c/devices/i2c-TIAS2781:00/ firmware_node modalias name power subsystem Device modalias: $ cat /sys/bus/i2c/devices/i2c-TIAS2781:00/modalias acpi:TIAS2781: PipeWire detects speaker sink: $ pactl list sinks short | grep Speaker 60 alsa_output.pci-0000_00_1f.3-platform-skl_hda_dsp_generic.HiFi__Speaker__sink Expected Behavior ================= The TAS2781 driver should recognize the ASUS VivoBook 16 Flip TP3607SA (subsystem ID 1043:3e30) as a supported platform and successfully initialize the smart amplifiers, enabling internal speaker audio output. Workarounds Attempted ===================== 1. Manual I2C device creation: Created tas2781 devices at addresses 0x38-0x3b, but they fail with communication timeouts (error -121) 2. Driver parameters: Tried various modprobe options, but the driver doesn't expose platform override parameters 3. GPIO manipulation: Attempted to enable GPIO pins that might control amplifier power, without success 4. Kernel versions: Tested multiple 6.14.x kernels including recent versions with November 26, 2024 ASUS TAS2781 improvements Request ======= Please add support for ASUS VivoBook 16 Flip TP3607SA (subsystem ID 1043:3e30) to the TAS2781 driver's supported platform list. This appears to be a matter of adding the device ID to the driver's whitelist and possibly providing the appropriate ACPI configuration parsing. Additional Notes ================ - This is a recent laptop model (2024) with Intel Lunar Lake platform - Other ASUS laptops may have similar TAS2781 configurations - The November 26, 2024 commit "ALSA: hda/tas2781: Add speaker id check for ASUS projects" suggests ongoing work on ASUS TAS2781 support - Hardware detection is working perfectly through SOF firmware Contact Information =================== Available for testing patches or providing additional debugging information. Attachments =========== - Full dmesg output available upon request - ACPI tables dump available if needed - Detailed hardware information via sudo lshw available

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conf/pistachio: fix syntax
by GitHub pull_request - opened 01 Aug '25

01 Aug '25

alsa-project/alsa-lib pull request #470 was opened from blino: It was missing closing brackets since its introduction. Fixes: 4dfa8f08fb83 ("conf/cards: add support for pistachio-card.") Request URL : https://github.com/alsa-project/alsa-lib/pull/470 Patch URL : https://github.com/alsa-project/alsa-lib/pull/470.patch Repository URL: https://github.com/alsa-project/alsa-lib

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retire_capture_urb: Corrected urb data len
by Sean Anderson 01 Aug '25

01 Aug '25

Hi all, I have a "FiiO DigiHug USB Audio" sound card (1852:7022) [3]. I have had no problems with the audio, but I did notice a large number of message like retire_capture_urb: 4992 callbacks suppressed in my dmesg [1]. This is caused by the "Corrected urb data len." warning. The patch adding this warning [2] makes it seem like this warning should be an uncommon occurance. However, based on the number of suppressed callbacks, this seems to be happening at a rate of around 500 Hz. Is this buggy hardware? Or is this a bug in the driver? Does there need to be a quirk? Or perhaps the warning above should be a debug instead? I'm using 5.15.0-46-generic. I know, I know; this is a vendor kernel. However, I went through the commits for sound/usb/pcm.c and didn't see any major differences. I also previously saw this issue on 5.4, so this probably isn't a regression. --Sean [1] I would have liked to post the actual message, but I was unable to figure out how to disable ratelimiting. I tried setting /proc/sys/kernel/printk_ratelimit to 0, /proc/sys/kernel/printk_ratelimit_burst to 5000, and /proc/sys/kernel/printk to 8. However, nothing seemed to have any effect (except that the callbacks suppressed messages were disabled). I find this very strange, given that the callbacks suppressed message uses KERN_WARN, and so does dev_warn_ratelimited. [2] https://lore.kernel.org/all/4B28A659.6070303@ladisch.de/T/ [3] Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x1852 GYROCOM C&C Co., LTD idProduct 0x7022 bcdDevice 0.01 iManufacturer 1 FiiO iProduct 2 DigiHug USB Audio iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0182 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 58 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0012 1x 18 bytes bInterval 32 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 3 FiiO USB DAC-E10 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x003e bInCollection 2 baInterfaceNr(0) 2 baInterfaceNr(1) 3 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 9 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bSourceID 16 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 16 bSourceID 9 bControlSize 1 bmaControls(0) 0x01 Mute Control bmaControls(1) 0x02 Volume Control bmaControls(2) 0x02 Volume Control iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 26 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 6 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 32000 tSamFreq[ 3] 44100 tSamFreq[ 4] 48000 tSamFreq[ 5] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0184 1x 388 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 26 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 6 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 32000 tSamFreq[ 3] 44100 tSamFreq[ 4] 48000 tSamFreq[ 5] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0246 1x 582 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 32000 tSamFreq[ 1] 44100 tSamFreq[ 2] 48000 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0184 1x 388 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 4 Discrete tSamFreq[ 0] 32000 tSamFreq[ 1] 44100 tSamFreq[ 2] 48000 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0246 1x 582 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x2001 IEC1937_AC-3 AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 3 (FORMAT_TYPE_III) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 can't get debug descriptor: Resource temporarily unavailable Device Status: 0x0000 (Bus Powered)

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pcm: add a loop to snd_pcm_avail_delay() to avoid bogus delay values
by GitHub pull_request - opened 31 Jul '25

31 Jul '25

alsa-project/alsa-lib pull request #469 was opened from kv2019i: snd_pcm_avail_delay() is expected to report avail and delay values in atomic fashion. However the function does two separate syscalls and it cannot guarantee the avail value is the same as was used to calculate the delay. This is a problem as the reported delay is always relative to avail frames value. If application (like e.g. alsa_conformance_test) uses snd_pcm_avail_delay() to estimate the effective play position, it can observe bogus delay values (and effective play position going backwards) if snd_pcm_avail_delay() is called during a DMA burst where hw_ptr moves quickly. This commit adds a loop similar to that used in snd_pcm_hw_htimestamp() to wait until we get a stable avail reading, and only then extract the delay. This will avoid bogus values if function is called during DMA bursts. Closes: https://github.com/alsa-project/alsa-lib/issues/468 Request URL : https://github.com/alsa-project/alsa-lib/pull/469 Patch URL : https://github.com/alsa-project/alsa-lib/pull/469.patch Repository URL: https://github.com/alsa-project/alsa-lib

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snd_pcm_avail_delay() reporting inaccurate delay values
by GitHub issues - opened 31 Jul '25

31 Jul '25

alsa-project/alsa-lib issue #468 was opened from kv2019i: The alsa_conformance_test tool (https://chromium.googlesource.com/chromiumos/platform/audiotest/+/HEAD/alsa…) has recently added support to use snd_pcm_avail_delay() to estimate the actual playback position. This has proven a very useful tool to test ALSA drivers on hardware (like that support by https://github.com/thesofproject/ ) where there is significant buffering in hardware (and hw_ptr can move in bursts). When testing with high sampling rates, an issue was found where the estimated playback position seemed to go backwards in time, as observed by alsa_conformance_test. This was observed on multiple Intel platform using SOF firmware for DSP. Debugging the kernel driver delay() implementation showed, that driver never reported incorrect delay values, but somehow alsa_conformance_test still observed these. Example log: ``` alsa_conformance_test -P hw:sofnocodec,2 -d 1 -r 96000 -B 480 -c 2 -f S32_LE -p 480 --avail-delay --debug ... TIME_DIFF(s) HW_LEVEL PLAYED DELAY DIFF RATE [.. snip ] 0.000999950 960 3324 1476 96 96004.800240 0.000995762 864 3420 1380 96 96408.579560 0.001024820 672 3516 1284 96 93674.986827 0.000987638 621 3613 1187 97 98214.122988 [Notice] frame_diff -1 >= 4096. 0.000049322 576 3612 1188 -1 -20274.928024 # KV: the delay value of 1187 is never reported by driver and it causes the playback position to go backwards in time despite driver reporting the accurate delay correctly 0.000992228 480 3708 1092 96 96751.956204 0.000972403 864 3804 1476 96 98724.500027 0.000998287 768 3900 1380 96 96164.730183 0.000979150 768 3996 1284 96 98044.222029 0.000999349 672 4092 1188 96 96062.536711 0.001018172 480 4188 1092 96 94286.623478 0.000977025 960 4284 1476 96 98257.465264 0.001022131 768 4380 1380 96 93921.424945 0.000977795 768 4476 1284 96 98180.088873 0.001019073 601 4573 1187 97 95184.545170 [Notice] frame_diff -1 >= 4096. 0.000049433 576 4572 1188 -1 -20229.401412 0.000953494 480 4668 1092 96 100682.332558 0.000981225 960 4764 1476 96 97836.887564 0.001025294 768 4860 1380 96 93631.680279 ``` When instrumenting alsa-lib snd_pcm_avail_delay() it became apparent the function is calling in middle of a DMA burst (when hw_ptr is moving fast ahead and the "avail" number is changing fast). alsa-lib does two syscalls to read avail and delay, the two values are not in sync and we end up reporting a bogus value. The error is not a big one, but this is enough to fail alsa_conformance_test test plans and push the calculcated standard rate error above acceptance criteria. This could also confuse media players who use this for video lip-sync, although in theory applications should handle this (a 0...3 frame error in delay reporting is not going to make a difference unless negative value breaks the application logic like happens here with alsa_conformance_test). We can observe the error especially with higher rates (96, 176.4 and 192kHz). This is harder to hit with the more typical 48khz use-cases, and thus maybe has not been hit before. Versions tested: 1.12.24, 1.12.11 Issue URL : https://github.com/alsa-project/alsa-lib/issues/468 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ucm: use close_range on _GNU_SOURCE
by Jochen Sprickerhof 30 Jul '25

30 Jul '25

--- src/ucm/ucm_exec.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/src/ucm/ucm_exec.c b/src/ucm/ucm_exec.c index 276cf592..b79a84ae 100644 --- a/src/ucm/ucm_exec.c +++ b/src/ucm/ucm_exec.c @@ -254,8 +254,12 @@ int uc_mgr_exec(const char *prog) close(f); +#if defined(_GNU_SOURCE) + close_range(3, maxfd, 0); +#else for (f = 3; f < maxfd; f++) close(f); +#endif /* install default handlers for the forked process */ signal(SIGINT, SIG_DFL); -- 2.50.0

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[PATCH v1] ASoC: tas2781: Normalize the volume kcontrol name
by Baojun Xu 30 Jul '25

30 Jul '25

As more devices become supported, add "tas2781" as a suffix before the TLV name, correct the DVC step, and normalize the volume kcontrol name. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781-tlv.h | 6 ++++-- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 4 ++-- sound/hda/codecs/side-codecs/tas2781_hda_spi.c | 5 +++-- sound/soc/codecs/tas2781-i2c.c | 8 ++++---- 4 files changed, 13 insertions(+), 10 deletions(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index d87263e43fdb..38b8f828a353 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -15,7 +15,9 @@ #ifndef __TAS2781_TLV_H__ #define __TAS2781_TLV_H__ -static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); -static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_dvc_tlv, + -10000, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_amp_tlv, + 1100, 50, 0); #endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index a0b132681804..6ec93b27aa7f 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -265,9 +265,9 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { }; static const struct snd_kcontrol_new tas2781_snd_controls[] = { - ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_putvol, tas2781_amp_tlv), ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c index 09a5d0f131b2..b4664a696bad 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_spi.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_spi.c @@ -494,9 +494,10 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new tas2781_snd_ctls[] = { ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_AMP_LEVEL, 1, 0, 20, 0, - tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv), + tas2781_amp_getvol, tas2781_amp_putvol, tas2781_amp_tlv), ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_DVC_LVL, 0, 0, 200, 1, - tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), + tas2781_digital_getvol, tas2781_digital_putvol, + tas2781_dvc_tlv), ACARD_SINGLE_BOOL_EXT(NULL, 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9f4d965a1335..a51f20cd869a 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -908,12 +908,12 @@ static const struct snd_kcontrol_new tasdevice_cali_controls[] = { }; static const struct snd_kcontrol_new tas2781_snd_controls[] = { - SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, - tas2781_amp_putvol, amp_vol_tlv), - SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, + tas2781_amp_putvol, tas2781_amp_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, - tas2781_digital_putvol, dvc_tlv), + tas2781_digital_putvol, tas2781_dvc_tlv), }; static const struct snd_kcontrol_new tas2781_cali_controls[] = { -- 2.43.0

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[PATCH v2] ALSA: hda: Fix the wrong register was used for DVC of TAS2770
by Baojun Xu 29 Jul '25

29 Jul '25

The wrong register was used for digital volume control of TAS2770, The definition was changed, and usage was also updated. Fixes: ab29b3460c5c ("ALSA: hda: Add TAS2770 support") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v2 - Correct number from -1000 to -10000 for DVC. --- include/sound/tas2770-tlv.h | 4 ++-- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/tas2770-tlv.h b/include/sound/tas2770-tlv.h index c0bd495b4a07..c7380925417a 100644 --- a/include/sound/tas2770-tlv.h +++ b/include/sound/tas2770-tlv.h @@ -14,10 +14,10 @@ #ifndef __TAS2770_TLV_H__ #define __TAS2770_TLV_H__ -#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x17) +#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x05) #define TAS2770_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) -static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, -10000, 50, 0); static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_amp_tlv, 1100, 50, 0); #endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index a0b132681804..45ac5e41bd4f 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -260,7 +260,7 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { 0, 0, 20, 0, tas2781_amp_getvol, tas2781_amp_putvol, tas2770_amp_tlv), ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2770_DVC_LEVEL, - 0, 0, 31, 0, tas2781_amp_getvol, + 0, 0, 200, 1, tas2781_amp_getvol, tas2781_amp_putvol, tas2770_dvc_tlv), }; -- 2.43.0

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