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

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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[alsa-devel] Applied "ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF" to the asoc tree
by Mark Brown 17 Jul '25

17 Jul '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

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[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

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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

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[alsa-devel] ASoC TDM channel re-mapping
by Edward Cragg 17 Jun '25

17 Jun '25

Hi, We have an ASoC driver in which we have added support for TDM over an I2S interface. We would like to have the capability of re-mapping the channel ordering, for which i know there is a callback in place to start this process, set_channel_map. The only backend implementation of this is using the ALSA API and the .copy handler in an ALSA PCM driver to mutate data copied out of ALSA before it's DMA'd to hardware. This is in sound/soc/blackfin/bf5xx-i2s-pcm.c The problem we have is that the driver we're using currently uses a dmaengine based PCM driver, and i'm not sure whether this is compatible with using the .copy callback, without reimplementing the driver entirely as an ALSA PCM driver. Since there aren't very many examples in mainline, I was hoping to ask for advice for the best way of doing this. I wondered: - Would it be possible to add a functional copy callback using a dmaengine based PCM driver? - Otherwise, would there be a good way to implement channel remapping in a TDM frame using the dmaengine based PCM driver? or - Would the best way indeed be to reimplement the PCM driver as a normal ALSA PCM driver? Thanks, Ed Cragg

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[PATCH] ASoC: qcom: soundwire: Add support for controlling audio CGCR from HLOS
by Srinivasa Rao Mandadapu 17 Jun '25

17 Jun '25

Add support for controlling soundwire audio CGCR interface using clock framework to make hclk ungating with software. As per new hardware changes, software has to always ungate hclk if soundwire is operational and keep it running. This requirement is for latest LPASS chipsets for RX, TX and WSA path to work. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> --- This patch set depends on: -- Clock driver patches with reset control support, yet to be posted. drivers/soundwire/qcom.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index fb183bd..440aad8 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -12,6 +12,7 @@ #include <linux/of_irq.h> #include <linux/of_device.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <linux/slab.h> #include <linux/slimbus.h> #include <linux/soundwire/sdw.h> @@ -129,6 +130,7 @@ struct qcom_swrm_ctrl { struct device *dev; struct regmap *regmap; void __iomem *mmio; + struct reset_control *audio_cgcr; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs; #endif @@ -618,6 +620,8 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) val = FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_ROW_CTRL_BMSK, ctrl->rows_index); val |= FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_COL_CTRL_BMSK, ctrl->cols_index); + reset_control_reset(ctrl->audio_cgcr); + ctrl->reg_write(ctrl, SWRM_MCP_FRAME_CTRL_BANK_ADDR(0), val); /* Enable Auto enumeration */ @@ -1269,6 +1273,9 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.ops = &qcom_swrm_ops; ctrl->bus.port_ops = &qcom_swrm_port_ops; ctrl->bus.compute_params = &qcom_swrm_compute_params; + ctrl->audio_cgcr = devm_reset_control_get_exclusive(dev, "swr_audio_cgcr"); + if (IS_ERR(ctrl->audio_cgcr)) + dev_err(dev, "Failed to get audio_cgcr reset required for soundwire-v1.6.0\n"); ret = qcom_swrm_get_port_config(ctrl); if (ret) -- 2.7.4

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[PATCH] ASoC: amd: acp: Fix support for a Huawei Matebook laptop
by Marian Postevca 17 Jun '25

17 Jun '25

Previous commit that added support for Huawei MateBook D16 2021 with Ryzen 4600H (HVY-WXX9 M1010) was incomplete. To activate support for this laptop, the DMI table in acp3x-es83xx machine driver must also be updated. Fixes: b5338b1b901e ("ASoC: amd: acp: Add support for a new Huawei Matebook laptop") Signed-off-by: Marian Postevca <posteuca(a)mutex.one> --- sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c index f85b85ea4be9..2b0aa270a3e9 100644 --- a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c +++ b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c @@ -354,6 +354,14 @@ static const struct dmi_system_id acp3x_es83xx_dmi_table[] = { }, .driver_data = (void *)(ES83XX_ENABLE_DMIC|ES83XX_48_MHZ_MCLK), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "HVY-WXX9"), + DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "M1010"), + }, + .driver_data = (void *)(ES83XX_ENABLE_DMIC), + }, { .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"), -- 2.43.0

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Request for help: How to upgrade a sound driver from kernel 4.14 to 6.6
by Steve Broshar 10 Jun '25

10 Jun '25

I have an audio driver/module that builds for kernel 4.14, but not for 6.6 due to changes in the kernel API.I find that API types (structs) and functions have been removed or changed, but IDK how to modify my code to use the updated API. Further, some of the changes seem less than clean. For example, in a couple places struct fields have been removed, but comments that used to precede the fields are still there such the comments are clearly no longer relevant and are therefore confusing. I have to wonder how many comments are out-of-date. My ask is for help fixing my code. I don't expect you to fix directly, but I need to know how to go about figuring out how to fix it. What changed in the API? Is there an upgrade guide/info? I am going to include more than a little info. I realize that it might be overwhelming. But, I fear that I try to reduce the info too much it will not convey the relevant context. I'll start with my analysis of my compiler errors that I get. Seems that the struct snd_soc_codec and struct snd_soc_codec_driver were removed. Seems that snd_soc_register_codec() and snd_soc_unregister_codec() were removed. What replaces these things? What do I need to change to accommodate the removal of these? FYI, here are my compiler errors: <build-errors> $ make -C ../../.. M=$PWD make: Entering directory '/home/steve/code/compton-kernel' CC [M] /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.o /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:110:38: warning: 'struct snd_soc_codec' declared inside parameter list will not be visible outside of this definition or declaration 110 | static int codec_driver_probe(struct snd_soc_codec *codec) | ^~~~~~~~~~~~~ In file included from ./include/linux/device.h:15, from ./include/linux/platform_device.h:13, from ./include/linux/of_device.h:5, from /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:7: /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c: In function 'codec_driver_probe': /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:112:19: error: invalid use of undefined type 'struct snd_soc_codec' 112 | dev_info(codec->dev, "Entered %s", __func__); | ^~ ./include/linux/dev_printk.h:110:25: note: in definition of macro 'dev_printk_index_wrap' 110 | _p_func(dev, fmt, ##__VA_ARGS__); \ | ^~~ /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:112:5: note: in expansion of macro 'dev_info' 112 | dev_info(codec->dev, "Entered %s", __func__); | ^~~~~~~~ /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c: At top level: /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:116:21: error: variable 'codec_driver' has initializer but incomplete type 116 | static const struct snd_soc_codec_driver codec_driver = { | ^~~~~~~~~~~~~~~~~~~~ /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:117:6: error: 'const struct snd_soc_codec_driver' has no member named 'probe' 117 | .probe = codec_driver_probe, | ^~~~~ /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:117:14: warning: excess elements in struct initializer 117 | .probe = codec_driver_probe, | ^~~~~~~~~~~~~~~~~~ /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:117:14: note: (near initialization for 'codec_driver') /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:118:6: error: 'const struct snd_soc_codec_driver' has no member named 'component_driver' 118 | .component_driver = { | ^~~~~~~~~~~~~~~~ /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:118:25: error: extra brace group at end of initializer 118 | .component_driver = { | ^ /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:118:25: note: (near initialization for 'codec_driver') /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:118:25: warning: excess elements in struct initializer /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:118:25: note: (near initialization for 'codec_driver') /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c: In function 'platform_driver_probe_': /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:153:12: error: implicit declaration of function 'snd_soc_register_codec'; did you mean 'snd_soc_register_dai'? [-Werror=implicit-function-declaration] 153 | stat = snd_soc_register_codec(&pdev->dev, &codec_driver, &dai_driver, 1); | ^~~~~~~~~~~~~~~~~~~~~~ | snd_soc_register_dai /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c: In function 'platform_driver_remove': /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:166:5: error: implicit declaration of function 'snd_soc_unregister_codec'; did you mean 'snd_soc_unregister_dai'? [-Werror=implicit-function-declaration] 166 | snd_soc_unregister_codec(&pdev->dev); | ^~~~~~~~~~~~~~~~~~~~~~~~ | snd_soc_unregister_dai /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c: At top level: /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.c:116:42: error: storage size of 'codec_driver' isn't known 116 | static const struct snd_soc_codec_driver codec_driver = { | ^~~~~~~~~~~~ cc1: some warnings being treated as errors make[2]: *** [scripts/Makefile.build:243: /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s/dep_i2s.o] Error 1 make[1]: *** [/home/steve/code/compton-kernel/Makefile:1921: /home/steve/code/compton-kernel/drivers/compton/dep-imx8-i2s] Error 2 make: *** [Makefile:234: __sub-make] Error 2 make: Leaving directory '/home/steve/code/compton-kernel' </build-errors> My code (for 4.14 kernel) --------------------------------- Here's my driver code. The macro INCLUDE_PROBLEMATIC_CODE is used around each chuck that fails to compile. If not defined, the code does compile, but of course is missing important parts. <my-code> #define INCLUDE_PROBLEMATIC_CODE /* * Audindate Dante Embedded Platform (DEP) I2S Codec (ALSA SoC) driver; Linux module * Supports sample rates, bit depths and TDM modes for Dante devices. */ #include <linux/module.h> #include <linux/of_device.h> #include <linux/clk.h> #include <sound/soc.h> #include <sound/pcm.h> #include <linux/printk.h> #define DEP_I2S_PCM_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) #define DEP_I2S_PCM_FORMATS ( SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE ) // for future use struct instance_state { unsigned int dummy_holder; }; static const struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_OUTPUT("LINEVOUT"), SND_SOC_DAPM_INPUT("LINEVIN"), }; static const struct snd_soc_dapm_route routes[] = { { "LINEVOUT", NULL, "DAC" }, { "ADC", NULL, "LINEVIN" }, }; static int set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: // supported break; case SND_SOC_DAIFMT_LEFT_J: // supported break; default: // all other formats unsupported dev_err(dai->dev, "Invalid DAI format: DEP I2S only supports left justified and I2S, got 0x%04X", fmt); return -EINVAL; } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { // codec as bit master and frame master (or external clock generator) // note that jumpers on DEP-EVK-IMX8 must be set correctly case SND_SOC_DAIFMT_CBM_CFM: // supported break; // codec as bit slave and frame slave (ie clocks driven from SOC) case SND_SOC_DAIFMT_CBS_CFS: // supported break; default: // all others unsupported dev_err(dai->dev, "Invalid DAI format: DEP I2S only supports CODEC as bit and frame clock master, got 0x%04X", fmt); return -EINVAL; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { default: // support bit and frame inversion break; } dev_info(dai->dev, "Set fmt to %u", fmt); return 0; } static struct snd_soc_dai_ops dep_i2s_dai_ops = { .set_fmt = set_fmt, // add mute here }; struct snd_soc_dai_driver dai_driver = { .name = "dep-i2s", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 16, .rate_min = 44100, .rate_max = 192000, .rates = DEP_I2S_PCM_RATES, .formats = DEP_I2S_PCM_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 16, .rate_min = 44100, .rate_max = 192000, .rates = DEP_I2S_PCM_RATES, .formats = DEP_I2S_PCM_FORMATS, }, .ops = &dep_i2s_dai_ops, .symmetric_rate = 1, }; #ifdef INCLUDE_PROBLEMATIC_CODE // Dummy probe static int codec_driver_probe(struct snd_soc_codec *codec) { dev_info(codec->dev, "Entered %s", __func__); return 0; } static const struct snd_soc_codec_driver codec_driver = { .probe = codec_driver_probe, .component_driver = { .dapm_widgets = widgets, .num_dapm_widgets = ARRAY_SIZE(widgets), .dapm_routes = routes, .num_dapm_routes = ARRAY_SIZE(routes), }, }; #endif #if defined(CONFIG_OF) static const struct of_device_id match_table[] = { {.compatible = "audinate,dep_i2s"}, {} }; MODULE_DEVICE_TABLE(of, match_table); #endif static int platform_driver_probe_(struct platform_device *pdev) { dev_info(&pdev->dev, "Entered %s", __func__); int stat = 0; struct instance_state *instance_state = devm_kzalloc(&pdev->dev, sizeof(struct instance_state), GFP_KERNEL); if (instance_state == NULL) { return -ENOMEM; } platform_set_drvdata(pdev, instance_state); dev_info(&pdev->dev, "Codec probe pt 2"); #ifdef INCLUDE_PROBLEMATIC_CODE // 1 == number of DAIs, might need to be 2 for dual direction, or for 2* sai? stat = snd_soc_register_codec(&pdev->dev, &codec_driver, &dai_driver, 1); if (stat < 0) { dev_err(&pdev->dev, "Failed to register codec: %d", stat); } #endif return stat; } static int platform_driver_remove(struct platform_device *pdev) { dev_info(&pdev->dev, "Entered %s", __func__); #ifdef INCLUDE_PROBLEMATIC_CODE snd_soc_unregister_codec(&pdev->dev); #endif return 0; } static struct platform_driver platform_driver = { .probe = platform_driver_probe_, .remove = platform_driver_remove, .driver = { .name = "dep-i2s-codec", .of_match_table = of_match_ptr(match_table), }, }; module_platform_driver(platform_driver); MODULE_DESCRIPTION("Audinate DEP generic I2S ASoC codec"); MODULE_AUTHOR("James Stuart <opensource(a)audinate.com>"); MODULE_ALIAS("platform:dep-i2s-codec"); MODULE_LICENSE("GPL"); </my-code> Steve Broshar Palmer Wireless Medtech

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[PATCH v4 0/5] Add DisplayPort sound support for Fairphone 5 smartphone
by Luca Weiss 09 Jun '25

09 Jun '25

Add the necessary sound card bits and some dts additions to enable sound over DisplayPort-over-USB-C, e.g. to a connected TV or monitor. The UCM files can be found here: https://gitlab.postmarketos.org/postmarketOS/pmaports/-/tree/master/device/… This series - in spirit - depends on the series enabling DisplayPort in the first place, but can land pretty independently, especially the ASoC bits: https://lore.kernel.org/linux-arm-msm/20250312-fp5-pmic-glink-dp-v2-0-a5592… The ASoC bits depend on the patches for USB audio offloading merged through Greg's tree so this cannot easily be applied for v6.16 through the sound tree. Either it waits for v6.17 or also goes through Greg's tree. Signed-off-by: Luca Weiss <luca.weiss(a)fairphone.com> --- Changes in v4: - Use "sm4250" for qcom,qrb4210-rb2-sndcard (Srinivas) - Expand cover letter to mention dependencies for ASoC patches - Pick up tags - Link to v3: https://lore.kernel.org/r/20250425-fp5-dp-sound-v3-0-7cb45180091b@fairphone… Changes in v3: - Simplify return qcom_snd_wcd_jack_setup in machine driver - Pick up tags - Link to v2: https://lore.kernel.org/r/20250418-fp5-dp-sound-v2-0-05d65f084b05@fairphone… Changes in v2: - Revamp series based on comments on v1, doesn't have much too much in common anymore - Use sm8250 instead of sc8280xp sndcard file, so port other required changes from sc8280xp.c to sm8250.c - This also changes the sound card compatible to from qcom,qcm6490-sndcard to fairphone,fp5-sndcard - Link to v1: https://lore.kernel.org/r/20240809-fp5-dp-sound-v1-0-d7ba2c24f6b9@fairphone… --- Luca Weiss (5): ASoC: dt-bindings: qcom,sm8250: Add Fairphone 5 sound card ASoC: qcom: sm8250: set card driver name from match data ASoC: qcom: sm8250: add DisplayPort Jack support ASoC: qcom: sm8250: Add Fairphone 5 soundcard compatible arm64: dts: qcom: qcm6490-fairphone-fp5: Add DisplayPort sound support .../devicetree/bindings/sound/qcom,sm8250.yaml | 1 + arch/arm64/boot/dts/qcom/qcm6490-fairphone-fp5.dts | 31 ++++++++++++++++++++++ sound/soc/qcom/sm8250.c | 27 ++++++++++--------- 3 files changed, 47 insertions(+), 12 deletions(-) --- base-commit: f819a667fa466d3b96b542b147053b6873f18ea1 change-id: 20240809-fp5-dp-sound-b3768f3019bd Best regards, -- Luca Weiss <luca.weiss(a)fairphone.com>

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[PATCH v3 0/5] Add DisplayPort sound support for Fairphone 5 smartphone
by Luca Weiss 09 Jun '25

09 Jun '25

Add the necessary sound card bits and some dts additions to enable sound over DisplayPort-over-USB-C, e.g. to a connected TV or monitor. The UCM files can be found here: https://gitlab.postmarketos.org/postmarketOS/pmaports/-/tree/master/device/… This series - in spirit - depends on the series enabling DisplayPort in the first place, but can land pretty independently, especially the ASoC bits: https://lore.kernel.org/linux-arm-msm/20250312-fp5-pmic-glink-dp-v2-0-a5592… Signed-off-by: Luca Weiss <luca.weiss(a)fairphone.com> --- Changes in v3: - Simplify return qcom_snd_wcd_jack_setup in machine driver - Pick up tags - Link to v2: https://lore.kernel.org/r/20250418-fp5-dp-sound-v2-0-05d65f084b05@fairphone… Changes in v2: - Revamp series based on comments on v1, doesn't have much too much in common anymore - Use sm8250 instead of sc8280xp sndcard file, so port other required changes from sc8280xp.c to sm8250.c - This also changes the sound card compatible to from qcom,qcm6490-sndcard to fairphone,fp5-sndcard - Link to v1: https://lore.kernel.org/r/20240809-fp5-dp-sound-v1-0-d7ba2c24f6b9@fairphone… --- Luca Weiss (5): ASoC: dt-bindings: qcom,sm8250: Add Fairphone 5 sound card ASoC: qcom: sm8250: set card driver name from match data ASoC: qcom: sm8250: add DisplayPort Jack support ASoC: qcom: sm8250: Add Fairphone 5 soundcard compatible arm64: dts: qcom: qcm6490-fairphone-fp5: Add DisplayPort sound support .../devicetree/bindings/sound/qcom,sm8250.yaml | 1 + arch/arm64/boot/dts/qcom/qcm6490-fairphone-fp5.dts | 31 ++++++++++++++++++++++ sound/soc/qcom/sm8250.c | 27 ++++++++++--------- 3 files changed, 47 insertions(+), 12 deletions(-) --- base-commit: 7f1dd4e6634d085f3c9652d4e0f1903659fb96f9 change-id: 20240809-fp5-dp-sound-b3768f3019bd Best regards, -- Luca Weiss <luca.weiss(a)fairphone.com>

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