June 2025 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF" to the asoc tree
by Mark Brown 11 Aug '25

11 Aug '25

The patch ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fd2df3aeafa4b4cc468d58e147e0822967034b71 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Thu, 21 Mar 2019 17:58:46 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF Model and manage codec clock input as a component in the Core Clock Framework. This should allow us to do some more complex clock management and power control. Also, some of the on-board chip clocks can be exposed to the outside, and this change will make those clocks easier to consume by other parts of the kernel. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4-clk.c | 34 ++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 18 +++++++++++---- 2 files changed, 47 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 5e495fc8d931..cded85009c8c 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -265,6 +265,30 @@ static const struct clk_ops aic32x4_pll_ops = { .get_parent = clk_aic32x4_pll_get_parent, }; +static int clk_aic32x4_codec_clkin_set_parent(struct clk_hw *hw, u8 index) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + + return regmap_update_bits(mux->regmap, + AIC32X4_CLKMUX, + AIC32X4_CODEC_CLKIN_MASK, index << AIC32X4_CODEC_CLKIN_SHIFT); +} + +static u8 clk_aic32x4_codec_clkin_get_parent(struct clk_hw *hw) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + unsigned int val; + + regmap_read(mux->regmap, AIC32X4_CLKMUX, &val); + + return (val & AIC32X4_CODEC_CLKIN_MASK) >> AIC32X4_CODEC_CLKIN_SHIFT; +} + +static const struct clk_ops aic32x4_codec_clkin_ops = { + .set_parent = clk_aic32x4_codec_clkin_set_parent, + .get_parent = clk_aic32x4_codec_clkin_get_parent, +}; + static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { { .name = "pll", @@ -274,6 +298,14 @@ static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { .ops = &aic32x4_pll_ops, .reg = 0, }, + { + .name = "codec_clkin", + .parent_names = + (const char *[]) { "mclk", "bclk", "gpio", "pll" }, + .num_parents = 4, + .ops = &aic32x4_codec_clkin_ops, + .reg = 0, + }, }; static struct clk *aic32x4_register_clk(struct device *dev, @@ -314,6 +346,8 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name) */ aic32x4_clkdesc_array[0].parent_names = (const char* []) { mclk_name, "bclk", "gpio", "din" }; + aic32x4_clkdesc_array[1].parent_names = + (const char *[]) { mclk_name, "bclk", "gpio", "pll" }; for (i = 0; i < ARRAY_SIZE(aic32x4_clkdesc_array); ++i) aic32x4_register_clk(dev, &aic32x4_clkdesc_array[i]); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 7cf8c7cedfe1..5496e4e080f4 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -735,12 +735,9 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); - /* PLL as CODEC_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_CLKMUX, - AIC32X4_CODEC_CLKIN_MASK, - AIC32X4_CODEC_CLKIN_PLL << AIC32X4_CODEC_CLKIN_SHIFT); /* DAC_MOD_CLK as BDIV_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_IFACE3, AIC32X4_BDIVCLK_MASK, + snd_soc_component_update_bits(component, AIC32X4_IFACE3, + AIC32X4_BDIVCLK_MASK, AIC32X4_DACMOD2BCLK << AIC32X4_BDIVCLK_SHIFT); /* NDAC divider value */ @@ -987,6 +984,15 @@ static int aic32x4_component_probe(struct snd_soc_component *component) { struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); u32 tmp_reg; + int ret; + + struct clk_bulk_data clocks[] = { + { .id = "codec_clkin" }, + }; + + ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); + if (ret) + return ret; if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); @@ -999,6 +1005,8 @@ static int aic32x4_component_probe(struct snd_soc_component *component) if (aic32x4->setup) aic32x4_setup_gpios(component); + clk_set_parent(clocks[0].clk, clocks[1].clk); + /* Power platform configuration */ if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { snd_soc_component_write(component, AIC32X4_MICBIAS, -- 2.20.1

4 5

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)

2 8

[PATCH] ASoC: qcom: x1e80100: Support boards with two speakers
by Krzysztof Kozlowski 17 Jul '25

17 Jul '25

Some Qualcomm X1E laptops have only two speakers. Regardless whether this sound card driver is suitable for them (we could re-use one for some older SoC), we should set reasonable channel map depending on the number of channels, not always 4-speaker setup. This change is necessary for bringing audio support on Lenovo Thinkpad T14s with Qualcomm X1E78100 and only two speakers. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/qcom/x1e80100.c | 40 ++++++++++++++++++++++++++++++++++----- 1 file changed, 35 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/x1e80100.c b/sound/soc/qcom/x1e80100.c index 898b5c26bf1e..8eb57fc12f0d 100644 --- a/sound/soc/qcom/x1e80100.c +++ b/sound/soc/qcom/x1e80100.c @@ -95,23 +95,53 @@ static int x1e80100_snd_hw_params(struct snd_pcm_substream *substream, return qcom_snd_sdw_hw_params(substream, params, &data->sruntime[cpu_dai->id]); } +static int x1e80100_snd_hw_map_channels(unsigned int *ch_map, int num) +{ + switch (num) { + case 1: + ch_map[0] = PCM_CHANNEL_FC; + break; + case 2: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + break; + case 3: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + ch_map[2] = PCM_CHANNEL_FC; + break; + case 4: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_LB; + ch_map[2] = PCM_CHANNEL_FR; + ch_map[3] = PCM_CHANNEL_RB; + break; + default: + return -EINVAL; + } + + return 0; +} + static int x1e80100_snd_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; - const unsigned int rx_slot[4] = { PCM_CHANNEL_FL, - PCM_CHANNEL_LB, - PCM_CHANNEL_FR, - PCM_CHANNEL_RB }; + unsigned int channels = substream->runtime->channels; + unsigned int rx_slot[4]; int ret; switch (cpu_dai->id) { case WSA_CODEC_DMA_RX_0: case WSA_CODEC_DMA_RX_1: + ret = x1e80100_snd_hw_map_channels(rx_slot, channels); + if (ret) + return ret; + ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, - ARRAY_SIZE(rx_slot), rx_slot); + channels, rx_slot); if (ret) return ret; break; -- 2.43.0

3 2

[BUG] Mackie DLZ Creator XS – USB Audio: write error -77 after accepted hw_params
by Brian Knutsson 04 Jul '25

04 Jul '25

Hi ALSA devs, I'm reporting an issue with the Mackie DLZ Creator XS USB audio device on Linux (tested on Ubuntu 24.04 and 25.05 with kernel 6.8+). The device is detected correctly via ALSA (card 2: XS), but as soon as playback is initiated – whether via `aplay` or `speaker-test` – it disconnects with `write error: -77, File descriptor in bad state`. Relevant kernel logs show: usb 3-7: Product: DLZ Creator XS usb 3-7: Manufacturer: LOUD Audio, LLC usb 3-7: USB disconnect, device number X This occurs with and without PipeWire/udev running. I’ve created a full ALSA log with `alsa-info` and opened a Kernel Bugzilla entry: <https://bugzilla.kernel.org/show_bug.cgi?id=220143> Is there any known quirk or patch needed for this hardware? Or a way to assist debugging further (usbmon trace etc.)? Thanks in advance, Brian Knutsson

2 1

Request for help: Outputting audio as an I2S stream
by Steve Broshar 01 Jul '25

01 Jul '25

Summary: My goal is to output audio to an ADAU1467 via I2S on Linux 6.6. I tried and then abandoned using a custom, device-specific driver. Now trying to use built-in simple-sound-card. But, driver does not initialize (probe). Don't know how to debug/fix the issue. I've been working with chapgpt extensively since I have found no other source of information on how to setup linux to output an I2S audio stream. In fact, I posted a question to a linux maillist about an issue a couple weeks ago and the reply from Lars Clausen showed output from chatgpt. In the old days folks would joke about googling something for someone. Today, I guess we can do the same WRT chatgpt. But, that question was about updating a driver. Chatgpt tells me that I shouldn't need a device-specific driver since the device is configured and controlled elsewhere. I just need to pump an audio stream into it. So, I started over with my solution; no longer working on a custom driver. Apparently, I can use the simple-sound-card and a dummy codec. But, it does not work. Maybe chatgpt is wrong and I need a device-specific driver. Or maybe my setup is wrong. I have worked with chatgpt to resolve my issues, but have hit a wall. Hoping to get help from humans now. 😊 The boot-time log (attached) shows nothing about my audio setup until the last line: platform sound-i2s: deferred probe pending I think that ‘deferred probe’ means that an initial probe failed. How can I tell what failed? What prevented it from succeeding? Would be nice if the driver told me what failed. But, there’s no error message, just an info msg that probe is pending. Here's the device tree setup that I added (based on input from chatgpt): / { // Compton audio output can be accomplished via a built-in driver -- simple-audio-card sound_i2s: sound-i2s { // specify the built-in simple sound card compatible = "simple-audio-card"; // seems to be a descriptive name that shows in userland; i.e. via 'aplay -l' simple-audio-card,name = "imx8mn-adauslave"; // might specify I2S (as opposed to descriptive text) simple-audio-card,format = "i2s"; // defines CPU as master for bit-clock simple-audio-card,bitclock-master = <&sound_i2s_cpu>; // defines CPU as master for frame-clock simple-audio-card,frame-master = <&sound_i2s_cpu>; sound_i2s_cpu: simple-audio-card,cpu { sound-dai = <&sai3>; dai-tdm-slot-num = <2>; /* stereo */ dai-tdm-slot-width = <32>; /* bits per slot */ }; simple-audio-card,codec { sound-dai = <&dummy_codec>; }; }; // Compton audio output does not require a device-specific codec; the built-in, dummy driver is sufficient dummy_codec: dummy-codec { compatible = "linux,snd-soc-dummy"; #sound-dai-cells = <0>; }; Here's the device tree config that comes with the NXP IMX8M build for nano: pinctrl_sai3: sai3grp { fsl,pins = < MX8MN_IOMUXC_SAI3_TXFS_SAI3_TX_SYNC 0xd6 MX8MN_IOMUXC_SAI3_TXC_SAI3_TX_BCLK 0xd6 MX8MN_IOMUXC_SAI3_MCLK_SAI3_MCLK 0xd6 MX8MN_IOMUXC_SAI3_TXD_SAI3_TX_DATA0 0xd6 >; }; sai3: sai@30030000 { compatible = "fsl,imx8mn-sai", "fsl,imx8mq-sai"; reg = <0x30030000 0x10000>; #sound-dai-cells = <0>; interrupts = <GIC_SPI 50 IRQ_TYPE_LEVEL_HIGH>; clocks = <&clk IMX8MN_CLK_SAI3_IPG>, <&clk IMX8MN_CLK_DUMMY>, <&clk IMX8MN_CLK_SAI3_ROOT>, <&clk IMX8MN_CLK_DUMMY>, <&clk IMX8MN_CLK_DUMMY>; clock-names = "bus", "mclk0", "mclk1", "mclk2", "mclk3"; dmas = <&sdma2 4 2 0>, <&sdma2 5 2 0>; dma-names = "rx", "tx"; status = "disabled"; }; &sai3 { #sound-dai-cells = <0>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sai3>; assigned-clocks = <&clk IMX8MN_CLK_SAI3>; assigned-clock-parents = <&clk IMX8MN_AUDIO_PLL1_OUT>; assigned-clock-rates = <24576000>; fsl,sai-mclk-direction-output; status = "okay"; }; In summary, how do I further debug this? In general, and if you know, how does one output an audio stream as I2S? Steve Broshar Palmer Wireless Medtech

2 1

[PATCH v1] ASoC: Intel: soc-acpi-intel-ptl-match: Add rt721_l0 support
by Ajye Huang 30 Jun '25

30 Jun '25

Patch adds driver data & match table for rt721 multi-function codec on PTL at sdw link0. Signed-off-by: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> --- .../intel/common/soc-acpi-intel-ptl-match.c | 24 +++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c index eae75f3f0fa4..0257e12e8b6e 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c @@ -330,6 +330,15 @@ static const struct snd_soc_acpi_adr_device rt1320_3_group1_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt721_0_single_adr[] = { + { + .adr = 0x000030025d072101ull, + .num_endpoints = ARRAY_SIZE(rt_mf_endpoints), + .endpoints = rt_mf_endpoints, + .name_prefix = "rt721" + } +}; + static const struct snd_soc_acpi_adr_device rt721_3_single_adr[] = { { .adr = 0x000330025d072101ull, @@ -339,6 +348,15 @@ static const struct snd_soc_acpi_adr_device rt721_3_single_adr[] = { } }; +static const struct snd_soc_acpi_link_adr ptl_rt721_l0[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt721_0_single_adr), + .adr_d = rt721_0_single_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr ptl_rt721_l3[] = { { .mask = BIT(3), @@ -664,6 +682,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_sdw_machines[] = { .sof_tplg_filename = "sof-ptl-rt712-l3-rt1320-l3.tplg", .get_function_tplg_files = sof_sdw_get_tplg_files, }, + { + .link_mask = BIT(0), + .links = ptl_rt721_l0, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-ptl-rt721.tplg", + }, { .link_mask = BIT(3), .links = ptl_rt721_l3, -- 2.25.1

1 0

ucm2: Qualcomm: Add Dell Latitude 7455 / Inspiron 14 Plus 7441 support
by GitHub pull_request - edited 29 Jun '25

29 Jun '25

alsa-project/alsa-ucm-conf pull request #589 was edited from valpackett: (Both SKUs are identical, the codename is Thena) This configuration is: - 4 speakers - 2 DMICs - headset with mic - DisplayPort (not yet) --- audioreach: https://github.com/linux-msm/audioreach-topology/pull/25 cc @0xB0D @alexVinarskis Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/589 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/589.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ucm2: Qualcomm: Add Dell Latitude 7455 / Inspiron 14 Plus 7441 support
by GitHub pull_request - opened 29 Jun '25

29 Jun '25

alsa-project/alsa-ucm-conf pull request #589 was opened from valpackett: (Both SKUs are identical, the codename is Thena) This configuration is: - 4 speakers - 2 DMICs - headset with mic - DisplayPort (not yet) --- cc @0xB0D @alexVinarskis Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/589 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/589.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Focusrite Scarlett 18i20 Gen 1: UCM configuration defines incorrect capture channel count
by GitHub issues - opened 29 Jun '25

29 Jun '25

alsa-project/alsa-ucm-conf issue #588 was opened from mkuhl: ## Summary The ALSA UCM configuration for Focusrite Scarlett 18i20 incorrectly defines 20 capture channels (`Define.DirectCaptureChannels 20`), but the 1st generation hardware only supports 18 channels. This causes the device to fail initialization with UCM profile errors. ## Device Information - **Device**: Focusrite Scarlett 18i20 (1st Generation) - **USB ID**: 1235:800c - **USB Name**: Focusrite Scarlett 18i20 USB - **Driver**: snd_usb_audio ## Channel Configuration (Gen 1) - 8 analog inputs - 8 ADAT inputs - 2 S/PDIF inputs - **Total**: 18 input channels (not 20) ## Error Message ``` spa.alsa: Error in ALSA UCM profile for _ucm0007.hw:USB,0 (HiFi: Direct 2: source): CaptureChannels=20 > avail 18 spa.alsa: Errors in ALSA UCM profile for card Scarlett 18i20 ``` ## Affected Files The channel definition is in the main configuration file: `/usr/share/alsa/ucm2/USB-Audio/Focusrite/Scarlett-18i20.conf` Current (incorrect): ``` Define.DirectCaptureChannels 20 ``` Should be: ``` Define.DirectCaptureChannels 18 ``` This file then includes `Scarlett-18i20-HiFi.conf` which uses the defined value. ## System Information - **Distribution**: Fedora 42 / Aurora-dx - **Kernel**: 6.15.3-200.fc42.x86_64 - **PipeWire**: 1.4.5-1.fc42 - **WirePlumber**: 0.5.8-1.fc42 - **ALSA UCM**: 1.2.14-2.fc42 - **ALSA lib**: 1.2.14-2.fc42 ## Steps to Reproduce 1. Connect Focusrite Scarlett 18i20 Gen 1 (USB ID 1235:800c) 2. Start PipeWire/WirePlumber 3. Check device status: `wpctl status | grep Scarlett` 4. Device shows "[ALSA UCM error]" 5. Journal shows channel count mismatch error ## Workaround Users can create a local override: ```bash mkdir -p ~/.local/share/alsa/ucm2/USB-Audio/Focusrite/ cp /usr/share/alsa/ucm2/USB-Audio/Focusrite/Scarlett-18i20.conf \ ~/.local/share/alsa/ucm2/USB-Audio/Focusrite/ cp /usr/share/alsa/ucm2/USB-Audio/Focusrite/Scarlett-18i20-HiFi.conf \ ~/.local/share/alsa/ucm2/USB-Audio/Focusrite/ sed -i 's/Define.DirectCaptureChannels 20/Define.DirectCaptureChannels 18/g' \ ~/.local/share/alsa/ucm2/USB-Audio/Focusrite/Scarlett-18i20.conf ``` ## Root Cause Analysis The UCM configuration appears to be written for newer Scarlett 18i20 generations (2nd/3rd/4th Gen) which may have 20 channels due to additional loopback channels. The configuration includes generation detection logic but doesn't properly identify USB ID 1235:800c as a 1st Gen device, causing it to use the default 20-channel configuration. ## Proposed Solution The UCM configuration should detect Gen 1 devices and set the appropriate channel count: 1. Add detection for USB ID 1235:800c as Gen 1 2. Set `DirectCaptureChannels` to 18 for Gen 1 devices 3. Keep 20 channels for newer generations if applicable ## Impact - Device fails to initialize properly - Users cannot use the audio interface without manual workarounds - Affects all Linux distributions using this UCM configuration ## Additional Information - This is a regression - older ALSA versions worked correctly - The issue affects PipeWire/WirePlumber but not JACK when using Pro Audio profile (which bypasses UCM) - Other Gen 1 Scarlett devices (6i6, 8i8) may have similar issues ## References - Device specifications: https://support.focusrite.com/hc/en-gb/articles/206849839-Scarlett-1st-Gen-… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/588 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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tascam us 322 / 366
by GitHub issues - edited 29 Jun '25

29 Jun '25

alsa-project/alsa-firmware issue #12 was edited from riramau: Can someone add support for tascam 322 / 366 ? Issue URL : https://github.com/alsa-project/alsa-firmware/issues/12 Repository URL: https://github.com/alsa-project/alsa-firmware

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