February 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

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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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[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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[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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[PATCH 0/1] Swapped l/r channels on ThinkBook 13x Gen4
by MadEnvel 28 May '25

28 May '25

Earlier commits introduced audio support for Lenovo ThinBook 13x Gen 4 that needed a patch due to missing _DSD config. Unfortunately, it looks like the left and right channels in the config are swapped. This patch fixes channel assignment. Tested on Lenovo ThinkBook 13x Gen4. MadEnvel (1): Fix swapped left and right channels for Lenovo ThinBook 13x Gen4 sound sound/pci/hda/cs35l41_hda_property.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) -- 2.45.2

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[PATCH 0/9] ALSA: usb-audio: More fixes / improvements on PCM
by Takashi Iwai 22 Apr '25

22 Apr '25

Hi, this is a patch set for USB-audio to fix and improve the PCM stream handling. The primary target is the enhancement of the low-latency playback that was introduced recently for dealing with the small period size (that is used typically by JACK). There are a few other fixes that are relevant with it. The current patches are found in topic/usb-audio branch of sound.git tree, too. Takashi === Takashi Iwai (9): ALSA: usb-audio: Restrict rates for the shared clocks ALSA: usb-audio: Fix possible race at sync of urb completions ALSA: usb-audio: Rename early_playback_start flag with lowlatency_playback ALSA: usb-audio: Disable low-latency playback for free-wheel mode ALSA: usb-audio: Disable low-latency mode for implicit feedback sync ALSA: usb-audio: Check available frames for the next packet size ALSA: usb-audio: Add spinlock to stop_urbs() ALSA: usb-audio: Improved lowlatency playback support ALSA: usb-audio: Avoid killing in-flight URBs during draining sound/usb/card.h | 11 ++- sound/usb/endpoint.c | 230 +++++++++++++++++++++++++++++++------------ sound/usb/endpoint.h | 13 ++- sound/usb/pcm.c | 157 ++++++++++++++++++++++------- 4 files changed, 305 insertions(+), 106 deletions(-) -- 2.26.2

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ALSA 1.2.8 release
by Jaroslav Kysela 15 Apr '25

15 Apr '25

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-lib alsa-utils alsa-ucm-conf tinycompress Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.7.2_v1.2.8 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

2 8

[PATCH] ALSA: hda: intel: Add Dell ALC3271 to power_save denylist
by Hoku Ishibe 05 Mar '25

05 Mar '25

Dell XPS 13 7390 with the Realtek ALC3271 codec experiences persistent humming noise when the power_save mode is enabled. This issue occurs when the codec enters power saving mode, leading to unwanted noise from the speakers. This patch adds the affected model (PCI ID 0x1028:0x0962) to the power_save denylist to ensure power_save is disabled by default, preventing power-off related noise issues. Steps to Reproduce 1. Boot the system with `snd_hda_intel` loaded. 2. Verify that `power_save` mode is enabled: ```sh cat /sys/module/snd_hda_intel/parameters/power_save ```` output: 10 (default power save timeout) 3. Wait for the power save timeout 4. Observe a persistent humming noise from the speakers 5. Disable `power_save` manually: ```sh echo 0 | sudo tee /sys/module/snd_hda_intel/parameters/power_save ```` 6. Confirm that the noise disappears immediately. This issue has been observed on my system, and this patch successfully eliminates the unwanted noise. If other users experience similar issues, additional reports would be helpful. Signed-off-by: Hoku Ishibe <me(a)hokuishi.be> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7d7f9aac50a9..ff536a0e480c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2232,6 +2232,8 @@ static const struct snd_pci_quirk power_save_denylist[] = { SND_PCI_QUIRK(0x1631, 0xe017, "Packard Bell NEC IMEDIA 5204", 0), /* KONTRON SinglePC may cause a stall at runtime resume */ SND_PCI_QUIRK(0x1734, 0x1232, "KONTRON SinglePC", 0), + /* Dell ALC3271 */ + SND_PCI_QUIRK(0x1028, 0x0962, "Dell ALC3271", 0), {} }; -- 2.48.1

2 1

[PATCH v1 0/2] ASoC: codecs: add ES8389 codec driver
by Zhang Yi 03 Mar '25

03 Mar '25

The driver is for codec ES8389 of everest-semi. Zhang Yi (2): ASoC: codecs: add support for ES8389 ASoC: dt-bindings: Add Everest ES8389 audio CODEC .../bindings/sound/everest,es8389.yaml | 98 ++ sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8389.c | 985 ++++++++++++++++++ sound/soc/codecs/es8389.h | 139 +++ 5 files changed, 1229 insertions(+), 1 deletion(-) create mode 100755 Documentation/devicetree/bindings/sound/everest,es8389.yaml create mode 100644 sound/soc/codecs/es8389.c create mode 100644 sound/soc/codecs/es8389.h -- 2.17.1

3 4

Build with gettext-0.24 intermittently fails
by GitHub issues - edited 28 Feb '25

28 Feb '25

alsa-project/alsa-utils issue #294 was edited from heitbaum: After Autoconf the errors are as follows. The Makefile is not honouring / passing the builddir through. This is a build being done in a subdirectory. ``` Making all in po make[2]: Entering directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu/po' make[2]: *** No rule to make target '/config.status', needed by 'Makefile'. Stop. make[2]: Leaving directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu/po' make[1]: *** [Makefile:471: all-recursive] Error 1 make[1]: Leaving directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu' ``` - This is occurring just like https://github.com/zapping-vbi/zvbi/issues/57 - now testing to see if the change to `AM_GNU_GETTEXT_VERSION([0.24])` fixes the issue Issue URL : https://github.com/alsa-project/alsa-utils/issues/294 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Build with gettext-0.24 intermittently fails
by GitHub issues - opened 28 Feb '25

28 Feb '25

alsa-project/alsa-utils issue #294 was opened from heitbaum: After Autoconf the errors are as follows. The Makefile is not honouring / passing the builddir through. This is a build being done in a subdirectory. ``` Making all in po make[2]: Entering directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu/po' make[2]: *** No rule to make target '/config.status', needed by 'Makefile'. Stop. make[2]: Leaving directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu/po' make[1]: *** [Makefile:471: all-recursive] Error 1 make[1]: Leaving directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu' ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/294 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

How to clone from https://git.alsa-project.org/?
by GitHub issues - opened 28 Feb '25

28 Feb '25

alsa-project/alsa-utils issue #293 was opened from K0-RR: `git clone https://git.alsa-project.org/?p=alsa.git` gives me fatal: repository 'https://git.alsa-project.org/?p=alsa.git/' not found Issue URL : https://github.com/alsa-project/alsa-utils/issues/293 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v3] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 27 Feb '25

27 Feb '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convery the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - Merge the 4 patches into one patch v2: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: | https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/2025 | 02160812.P7szOygN-lkp(a)intel.com/__;!!G3vK!WriwhtbQGlg6p00bcWutlhb-y9vS | pDrSu5_g9oOzzGOoABdirEARTntFzyX6Ss3UPRDxi_E-IA$ - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: | https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/2025 | 02160632.u0zxG7yD-lkp(a)intel.com/__;!!G3vK!VtEdfur3fZklvPVfHFFfFP60Ub2E | JYvmjSUZ65lJ6BsQPJDl1fjYRDqh3c6Q5y6sA70ssXDG5g$ v1: - update the year from 2024 to 2025 in the files. - Add dspbin_type to distinguish different dspbin types, such as TASDEV_BASIC, TASDEV_ALPHA, and TASDEV_BETA. - Add struct bulk_reg_val alp_cali_bckp to store the value of pilot tone threshold only for TASDEV_ALPHA & TASDEV_BETA firmware. - Add the declaration of (*fw_parse_fct_param_address)() to parse the addresses of calibration params. - Add the minimum version of TASDEV_ALPHA and TASDEV_BETA firmware. - Add the definition of struct fct_param_address. - update the year from 2024 to 2025. - release the memory to save calibration addresses in tasdevice_dsp_remove(). - Add TASDEV_ALPHA and TASDEV_BETA firmware checking code. - Add fct_param_address_parser for Alpha and Beta firmware. - Add an extra branch to handle calibration address for TASDEV_ALPHA and TASDEV_BETA firmware. --- include/sound/tas2781-dsp.h | 26 +++- include/sound/tas2781.h | 13 +- sound/pci/hda/tas2781_spi_fwlib.c | 6 +- sound/soc/codecs/tas2781-comlib.c | 3 +- sound/soc/codecs/tas2781-fmwlib.c | 149 +++++++++++++++++++++- sound/soc/codecs/tas2781-i2c.c | 203 +++++++++++++++++++++++------- 6 files changed, 340 insertions(+), 60 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..77a240daaae0 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -30,8 +30,10 @@ #define PRE_DEVICE_C 0x12 #define PRE_DEVICE_D 0x16 -#define PPC3_VERSION 0x4100 -#define PPC3_VERSION_TAS2781 0x14600 +#define PPC3_VERSION_BASE 0x4100 +#define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 +#define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 +#define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -106,6 +108,23 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + unsigned char thr[3]; + unsigned char thr2[3]; + unsigned char plt_flg[3]; + unsigned char sin_gn[3]; + unsigned char sin_gn2[3]; + /* real-time spk impedance */ + unsigned char r0_reg[3]; + /* check spk connection */ + unsigned char tf_reg[3]; + /* check spk resonant frequency */ + unsigned char a1_reg[3]; + unsigned char a2_reg[3]; + unsigned short size; + char *data; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +133,7 @@ struct tasdevice_fw { struct tasdevice_config *configs; unsigned short nr_calibrations; struct tasdevice_calibration *calibrations; + struct fct_param_address fct_par_addr; struct device *dev; }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 72d2060904f6..5cfbb9e0a626 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -112,6 +112,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -126,6 +132,7 @@ struct bulk_reg_val { struct tasdevice { struct bulk_reg_val *cali_data_backup; + struct bulk_reg_val alp_cali_bckp; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -171,6 +178,7 @@ struct tasdevice_priv { unsigned char dev_name[32]; const char *name_prefix; unsigned char ndev; + unsigned int dspbin_typ; unsigned int magic_num; unsigned int chip_id; unsigned int sysclk; @@ -196,6 +204,9 @@ struct tasdevice_priv { int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset); + int (*fw_parse_fct_param_address)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); diff --git a/sound/pci/hda/tas2781_spi_fwlib.c b/sound/pci/hda/tas2781_spi_fwlib.c index 131d9a77d140..c90641119509 100644 --- a/sound/pci/hda/tas2781_spi_fwlib.c +++ b/sound/pci/hda/tas2781_spi_fwlib.c @@ -370,10 +370,10 @@ static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, int i, n = ARRAY_SIZE(non_ppc3_mapping_table); unsigned char dev_idx = 0; - if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) { p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; n = ARRAY_SIZE(ppc3_tas2781_mapping_table); - } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION_BASE) { p = (struct blktyp_devidx_map *)ppc3_mapping_table; n = ARRAY_SIZE(ppc3_mapping_table); } @@ -1605,7 +1605,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, int rc = 0; if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION) { + if (ppcver >= PPC3_VERSION_BASE) { tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 1e0b3aa95749..bba5abe4736f 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -2,7 +2,7 @@ // // TAS2563/TAS2781 Common functions for HDA and ASoC Audio drivers // -// Copyright 2023 - 2024 Texas Instruments, Inc. +// Copyright 2023 - 2025 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -429,6 +429,7 @@ void tasdevice_dsp_remove(void *context) if (tas_fmw->configs) tasdev_dsp_cfg_remove(tas_fmw->configs, tas_fmw->nr_configurations); + kfree(tas_fmw->fct_par_addr.data); kfree(tas_fmw); tas_dev->fmw = NULL; } diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 61d9c220b6a4..2fae7be99454 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2,7 +2,7 @@ // // tas2781-fmwlib.c -- TASDEVICE firmware support // -// Copyright 2023 - 2024 Texas Instruments, Inc. +// Copyright 2023 - 2025 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -389,10 +389,10 @@ static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, int i, n = ARRAY_SIZE(non_ppc3_mapping_table); unsigned char dev_idx = 0; - if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) { p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; n = ARRAY_SIZE(ppc3_tas2781_mapping_table); - } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION_BASE) { p = (struct blktyp_devidx_map *)ppc3_mapping_table; n = ARRAY_SIZE(ppc3_mapping_table); } @@ -559,6 +559,118 @@ static int fw_parse_configuration_data_kernel( return offset; } +static void hex_parse_u8(unsigned char *item, const unsigned char *val, + int len) +{ + for (int i = 0; i < len; i++) + item[i] = val[i]; +} + +static void hex_parse_u24(unsigned int *item, const unsigned char *val) +{ + *item = TASDEVICE_REG(val[0], val[1], val[2]); +} + +static void fct_param_address_parser(struct cali_reg *r, + struct tasdevice_fw *tas_fmw) +{ + struct fct_param_address *p = &(tas_fmw->fct_par_addr); + int i; + + for (i = 0; i < 20; i++) { + const unsigned char *dat = &p->data[24 * i]; + + if (dat[23] != 1) + break; + + if (strstr(dat, "umg_SsmKEGCye") != NULL) { + hex_parse_u24(&r->pow_reg, &dat[20]); + continue; + } + if (strstr(dat, "iks_E0") != NULL) { + hex_parse_u24(&r->r0_reg, &dat[20]); + continue; + } + if (strstr(dat, "yep_LsqM0") != NULL) { + hex_parse_u24(&r->invr0_reg, &dat[20]); + continue; + } + if (strstr(dat, "oyz_U0_ujx") != NULL) { + hex_parse_u24(&r->r0_low_reg, &dat[20]); + continue; + } + if (strstr(dat, "iks_GC_GMgq") != NULL) { + hex_parse_u24(&r->tlimit_reg, &dat[20]); + continue; + } + if (strstr(dat, "gou_Yao") != NULL) { + hex_parse_u8(p->thr, &dat[20], 3); + continue; + } + if (strstr(dat, "kgd_Wsc_Qsbp") != NULL) { + hex_parse_u8(p->plt_flg, &dat[20], 3); + continue; + } + if (strstr(dat, "yec_CqseSsqs") != NULL) { + hex_parse_u8(p->sin_gn, &dat[20], 3); + continue; + } + if (strstr(dat, "iks_SogkGgog2") != NULL) { + hex_parse_u8(p->sin_gn2, &dat[20], 3); + continue; + } + if (strstr(dat, "yec_Sae_Y") != NULL) { + hex_parse_u8(p->thr2, &dat[20], 3); + continue; + } + if (strstr(dat, "Re_Int") != NULL) { + hex_parse_u8(p->r0_reg, &dat[20], 3); + continue; + } + /* Check whether the spk connection is open */ + if (strstr(dat, "SigFlag") != NULL) { + hex_parse_u8(p->tf_reg, &dat[20], 3); + continue; + } + if (strstr(dat, "a1_Int") != NULL) { + hex_parse_u8(p->a1_reg, &dat[20], 3); + continue; + } + if (strstr(dat, "a2_Int") != NULL) { + hex_parse_u8(p->a2_reg, &dat[20], 3); + continue; + } + } +} + +static int fw_parse_fct_param_address(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct fct_param_address *p = &(tas_fmw->fct_par_addr); + struct calidata *cali_data = &tas_priv->cali_data; + struct cali_reg *r = &cali_data->cali_reg_array; + const unsigned char *data = fmw->data; + + if (offset + 520 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + + /* skip reserved part */ + offset += 40; + p->size = 480; + p->data = kmemdup(&data[offset], 480, GFP_KERNEL); + if (!p->data) { + offset = -1; + goto out; + } + offset += 480; + fct_param_address_parser(r, tas_fmw); +out: + return offset; +} + static int fw_parse_variable_header_kernel( struct tasdevice_priv *tas_priv, const struct firmware *fmw, int offset) @@ -1686,13 +1798,29 @@ static int tasdevice_load_block(struct tasdevice_priv *tas_priv, return rc; } +static void dspbin_type_check(struct tasdevice_priv *tas_priv, + unsigned int ppcver) +{ + if (ppcver >= PPC3_VERSION_TAS2781_ALPHA_MIN) { + if (ppcver >= PPC3_VERSION_TAS2781_BETA_MIN) + tas_priv->dspbin_typ = TASDEV_BETA; + else if (ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) + tas_priv->dspbin_typ = TASDEV_BASIC; + else + tas_priv->dspbin_typ = TASDEV_ALPHA; + } + if (tas_priv->dspbin_typ != TASDEV_BASIC) + tas_priv->fw_parse_fct_param_address = + fw_parse_fct_param_address; +} + static int dspfw_default_callback(struct tasdevice_priv *tas_priv, unsigned int drv_ver, unsigned int ppcver) { int rc = 0; if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION) { + if (ppcver >= PPC3_VERSION_BASE) { tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = @@ -1701,6 +1829,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, fw_parse_configuration_data_kernel; tas_priv->tasdevice_load_block = tasdevice_load_block_kernel; + dspbin_type_check(tas_priv, ppcver); } else { switch (ppcver) { case 0x00: @@ -1716,7 +1845,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, default: dev_err(tas_priv->dev, "%s: PPCVer must be 0x0 or 0x%02x", - __func__, PPC3_VERSION); + __func__, PPC3_VERSION_BASE); dev_err(tas_priv->dev, " Current:0x%02x\n", ppcver); rc = -EINVAL; @@ -2023,9 +2152,17 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, } offset = tas_priv->fw_parse_configuration_data(tas_priv, tas_fmw, fmw, offset); - if (offset < 0) + if (offset < 0) { ret = offset; + goto out; + } + if (tas_priv->fw_parse_fct_param_address) { + offset = tas_priv->fw_parse_fct_param_address(tas_priv, + tas_fmw, fmw, offset); + if (offset < 0) + ret = offset; + } out: return ret; } diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 90c5b2e74d12..8eb9fee27a42 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -349,13 +349,31 @@ static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, return rc; } +static int partial_cali_data_update(int *reg, int j) +{ + switch (tas2781_cali_start_reg[j].reg) { + case 0: + return reg[0]; + case TAS2781_PRM_PLT_FLAG_REG: + return reg[1]; + case TAS2781_PRM_SINEGAIN_REG: + return reg[2]; + case TAS2781_PRM_SINEGAIN2_REG: + return reg[3]; + default: + return 0; + } +} + static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, int *reg, unsigned char *dat) { struct tasdevice *tasdev = tas_priv->tasdevice; struct bulk_reg_val *p = tasdev[i].cali_data_backup; + struct bulk_reg_val *t = &tasdev[i].alp_cali_bckp; const int sum = ARRAY_SIZE(tas2781_cali_start_reg); - int j; + unsigned char val[4]; + int j, r; if (p == NULL) return; @@ -370,30 +388,23 @@ static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, tasdevice_dev_read(tas_priv, i, p[j].reg, (int *)&p[j].val[0]); } else { - switch (tas2781_cali_start_reg[j].reg) { - case 0: { - if (!reg[0]) - continue; - p[j].reg = reg[0]; + if (!tas_priv->dspbin_typ) { + r = partial_cali_data_update(reg, j); + if (r) + p[j].reg = r; } - break; - case TAS2781_PRM_PLT_FLAG_REG: - p[j].reg = reg[1]; - break; - case TAS2781_PRM_SINEGAIN_REG: - p[j].reg = reg[2]; - break; - case TAS2781_PRM_SINEGAIN2_REG: - p[j].reg = reg[3]; - break; - } - tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, - p[j].val, 4); + + if (p[j].reg) + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); } } + if (tas_priv->dspbin_typ == TASDEV_ALPHA) + tasdevice_dev_bulk_read(tas_priv, i, t->reg, t->val, 4); + /* Update the setting for calibration */ - for (j = 0; j < sum - 2; j++) { + for (j = 0; j < sum - 4; j++) { if (p[j].val_len == 1) { if (p[j].is_locked) tasdevice_dev_write(tas_priv, i, @@ -401,17 +412,33 @@ static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, TAS2781_TEST_PAGE_UNLOCK); tasdevice_dev_write(tas_priv, i, p[j].reg, tas2781_cali_start_reg[j].val[0]); - } else { - if (!p[j].reg) - continue; - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, - (unsigned char *) - tas2781_cali_start_reg[j].val, 4); } } - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, &dat[1], 4); - tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, &dat[5], 4); + if (tas_priv->dspbin_typ == TASDEV_ALPHA) { + val[0] = 0x00; + val[1] = 0x00; + val[2] = 0x21; + val[3] = 0x8e; + } else { + val[0] = tas2781_cali_start_reg[j].val[0]; + val[1] = tas2781_cali_start_reg[j].val[1]; + val[2] = tas2781_cali_start_reg[j].val[2]; + val[3] = tas2781_cali_start_reg[j].val[3]; + } + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, val, 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, + (unsigned char *)tas2781_cali_start_reg[j + 1].val, 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 2].reg, &dat[1], 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 3].reg, &dat[5], 4); + if (tas_priv->dspbin_typ == TASDEV_ALPHA) { + val[0] = 0x00; + val[1] = 0x00; + val[2] = 0x2a; + val[3] = 0x0b; + + tasdevice_dev_bulk_read(tas_priv, i, t->reg, val, 4); + } } static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, @@ -460,6 +487,7 @@ static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) for (i = 0; i < tas_priv->ndev; i++) { struct tasdevice *tasdev = tas_priv->tasdevice; struct bulk_reg_val *p = tasdev[i].cali_data_backup; + struct bulk_reg_val *t = &tasdev[i].alp_cali_bckp; if (p == NULL) continue; @@ -479,6 +507,10 @@ static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) p[j].val, 4); } } + + if (tas_priv->dspbin_typ == TASDEV_ALPHA) + tasdevice_dev_bulk_write(tas_priv, i, t->reg, + t->val, 4); } } @@ -590,16 +622,20 @@ static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, i += 2; priv->is_user_space_calidata = true; - p->r0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->r0_low_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->invr0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->pow_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->tlimit_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; + if (priv->dspbin_typ == TASDEV_BASIC) { + p->r0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->r0_low_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->invr0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->pow_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->tlimit_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + } else { + i += 15; + } memcpy(dst, &src[i], cali_data->total_sz); return 1; @@ -646,12 +682,19 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; - unsigned int reg; + unsigned int reg = TAS2781_RUNTIME_RE_REG_TF; + + if (tas_priv->chip_id == TAS2781) { + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); - if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG_TF; - else + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->tf_reg[0], p->tf_reg[1], + p->tf_reg[2]); + } else { reg = TAS2563_RUNTIME_RE_REG_TF; + } guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; @@ -666,12 +709,19 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; - unsigned int reg; + unsigned int reg = TAS2781_RUNTIME_RE_REG; - if (tas_priv->chip_id == TAS2781) - reg = TAS2781_RUNTIME_RE_REG; - else + if (tas_priv->chip_id == TAS2781) { + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); + + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->r0_reg[0], p->r0_reg[1], + p->r0_reg[2]); + } else { reg = TAS2563_RUNTIME_RE_REG; + } + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -705,11 +755,16 @@ static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, { struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A1_REG; + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->a1_reg[0], p->a1_reg[1], p->a1_reg[2]); + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -720,11 +775,16 @@ static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, { struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A2_REG; + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->a2_reg[0], p->a2_reg[1], p->a2_reg[2]); + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -1172,10 +1232,50 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) nr_controls < mix_index ? nr_controls : mix_index); } +static void cali_reg_update(struct bulk_reg_val *p, + struct fct_param_address *t) +{ + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int reg = 0; + int j; + + for (j = 0; j < sum; j++) { + switch (tas2781_cali_start_reg[j].reg) { + case 0: + reg = TASDEVICE_REG(t->thr[0], t->thr[1], t->thr[2]); + break; + case TAS2781_PRM_PLT_FLAG_REG: + reg = TASDEVICE_REG(t->plt_flg[0], t->plt_flg[1], + t->plt_flg[2]); + break; + case TAS2781_PRM_SINEGAIN_REG: + reg = TASDEVICE_REG(t->sin_gn[0], t->sin_gn[1], t->sin_gn[2]); + break; + case TAS2781_PRM_SINEGAIN2_REG: + reg = TASDEVICE_REG(t->sin_gn[0], t->sin_gn[1], t->sin_gn[2]); + break; + default: + reg = 0; + break; + } + if (reg) + p[j].reg = reg; + } +} + +static void alpa_cali_update(struct bulk_reg_val *p, + struct fct_param_address *t) +{ + p->is_locked = false; + p->reg = TASDEVICE_REG(t->thr2[0], t->thr2[1], t->thr2[2]); + p->val_len = 4; +} + static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) { struct calidata *cali_data = &priv->cali_data; struct tasdevice *tasdev = priv->tasdevice; + struct tasdevice_fw *fmw = priv->fmw; struct soc_bytes_ext *ext_cali_data; struct snd_kcontrol_new *cali_ctrls; unsigned int nctrls; @@ -1191,14 +1291,25 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) } if (priv->chip_id == TAS2781) { + struct fct_param_address *t = &(fmw->fct_par_addr); + cali_ctrls = (struct snd_kcontrol_new *)tas2781_cali_controls; nctrls = ARRAY_SIZE(tas2781_cali_controls); for (i = 0; i < priv->ndev; i++) { - tasdev[i].cali_data_backup = + struct bulk_reg_val *p; + + p = tasdev[i].cali_data_backup = kmemdup(tas2781_cali_start_reg, sizeof(tas2781_cali_start_reg), GFP_KERNEL); if (!tasdev[i].cali_data_backup) return -ENOMEM; + if (priv->dspbin_typ) { + cali_reg_update(p, t); + if (priv->dspbin_typ == TASDEV_ALPHA) { + p = &tasdev[i].alp_cali_bckp; + alpa_cali_update(p, t); + } + } } } else { cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; -- 2.34.1

3 4

[PATCH] soundwire: amd: change the log level for command response log
by Vijendar Mukunda 27 Feb '25

27 Feb '25

Change log level for command response log to dev_dbg_ratelimited when command is ignored. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index dcf85f94950a..a12c68b93b1c 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -346,7 +346,7 @@ static enum sdw_command_response amd_sdw_fill_msg_resp(struct amd_sdw_manager *a msg->dev_num); return SDW_CMD_FAIL; } - dev_err_ratelimited(amd_manager->dev, "command is ignored for Slave %d\n", + dev_dbg_ratelimited(amd_manager->dev, "command is ignored for Slave %d\n", msg->dev_num); return SDW_CMD_IGNORED; } -- 2.34.1

3 2

ALSA USB quirk mixer driver question
by Lena 27 Feb '25

27 Feb '25

Hi all! I'm currently trying to develop an usb quirk mixer driver for a desktop USB audio interface. The interface exposes the following: - interface 0-2 are USB UAC compliant, bound to by snd-usb-audio, and working flawlessly - interface 3 is vendor specific, and we can ignore it (used for firmware updates) - interface 4 exposes a non-standard mixer with a HID interface exposing 2 interrupt endpoints, and is used by the vendor windows app. I have studied the protocol with usbpcap and can successfully talk to the interface on the userspace, but i had a question regarding how to proceed in the kernel. This is because, after creating a new entry in `sound/usb/mixer_quirks.c` with all the needed controls, userspace can interact with them, etc. I can't figure out how to tell ALSA to bind to the correct interface for the mixer (and kick out usbhid). Looking at the other quirk mixer drivers it doesn't look (?) like any other quirk mixer driver is binding to a *different* interface, not even `sound/usb/mixer_scarlett.c`. Generally, in a standalone USB driver you'd have tracking/teardown handled easily with a callback, but i'm having trouble with the dependencies here. (simply using `usb_sndintpipe()` or `usb_interrupt_msg()` triggers kernel oopses, so that's probably not it) What i think i have to do is to use `usb_get_intf()`, then i have to somehow claim it (does `usb_driver_claim_interface()` work here?). But `usb_get_intf()` requires me to then release the interface with `usb_put_intf()`, and it doesn't look like there's anywhere for me to do that. (`snd_usb_mixer_elem_free()`? doesn't look like the order is correct) `iface_ref_find()` would sorta do what i need, but i cannot use it here. Could someone please point me in the right direction? I'm very stuck at the moment. Thanks, Lena

2 3

[PATCH v1] ASoC: tas2781: Clean up for some define
by Baojun Xu 26 Feb '25

26 Feb '25

Do some clean up for some define in header file. Like change lower case in value, up case in define, space add in recommends, change date of files and add author. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781.h | 21 +++++++++++---------- sound/soc/codecs/tas2781-fmwlib.c | 7 ++++--- 2 files changed, 15 insertions(+), 13 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 72d2060904f6..a2c8eb111e5f 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -11,6 +11,7 @@ // // Author: Shenghao Ding <shenghao-ding(a)ti.com> // Author: Kevin Lu <kevin-lu(a)ti.com> +// Author: Baojun Xu <baojun.xu(a)ti.com> // #ifndef __TAS2781_H__ @@ -31,7 +32,7 @@ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) -/*PAGE Control Register (available in page0 of each book) */ +/* PAGE Control Register (available in page0 of each book) */ #define TASDEVICE_PAGE_SELECT 0x00 #define TASDEVICE_BOOKCTL_PAGE 0x00 #define TASDEVICE_BOOKCTL_REG 127 @@ -42,12 +43,12 @@ #define TASDEVICE_REG(book, page, reg) (((book * 256 * 128) + \ (page * 128)) + reg) -/*Software Reset */ -#define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) +/* Software Reset */ +#define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) #define TASDEVICE_REG_SWRESET_RESET BIT(0) -/*I2C Checksum */ -#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) +/* I2C Checksum */ +#define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) /* XM_340 */ #define TASDEVICE_XM_A1_REG TASDEVICE_REG(0x64, 0x63, 0x3c) @@ -55,8 +56,8 @@ #define TASDEVICE_XM_A2_REG TASDEVICE_REG(0x64, 0x63, 0x38) /* Volume control */ -#define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) -#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) +#define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0c) +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1a) #define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) @@ -95,8 +96,8 @@ #define TAS2781_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x14, 0x40) #define TAS2781_PRM_SINEGAIN2_REG TASDEVICE_REG(0x00, 0x14, 0x44) -#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xFD, 0x0D) -#define TAS2781_TEST_PAGE_UNLOCK 0x0D +#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xfd, 0x0d) +#define TAS2781_TEST_PAGE_UNLOCK 0x0d #define TAS2781_RUNTIME_LATCH_RE_REG TASDEVICE_REG(0x00, 0x00, 0x49) #define TAS2781_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x62, 0x48) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 61d9c220b6a4..bc2f68bd9846 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2,9 +2,10 @@ // // tas2781-fmwlib.c -- TASDEVICE firmware support // -// Copyright 2023 - 2024 Texas Instruments, Inc. +// Copyright 2023 - 2025 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Baojun Xu <baojun.xu(a)ti.com> #include <linux/crc8.h> #include <linux/firmware.h> @@ -1496,7 +1497,7 @@ static int tasdev_block_chksum(struct tasdevice_priv *tas_priv, unsigned int nr_value; int ret; - ret = tasdevice_dev_read(tas_priv, chn, TASDEVICE_I2CChecksum, + ret = tasdevice_dev_read(tas_priv, chn, TASDEVICE_CHECKSUM_REG, &nr_value); if (ret < 0) { dev_err(tas_priv->dev, "%s: Chn %d\n", __func__, chn); @@ -1540,7 +1541,7 @@ static int tasdev_load_blk(struct tasdevice_priv *tas_priv, while (block->nr_retry > 0) { if (block->is_pchksum_present) { ret = tasdevice_dev_write(tas_priv, chn, - TASDEVICE_I2CChecksum, 0); + TASDEVICE_CHECKSUM_REG, 0); if (ret < 0) break; } -- 2.34.1

2 1

ucm2: Add support for MSI PRO B650-A w/ Realtek ALC4080
by GitHub pull_request - opened 26 Feb '25

26 Feb '25

alsa-project/alsa-ucm-conf pull request #514 was opened from gbrener: Gets the front panel audio to work. [alsa-info.txt](https://github.com/user-attachments/files/18993322/alsa-info… Thanks in advance. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/514 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/514.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: MediaTek: mt8390-evk: Add support for SOF
by GitHub pull_request - opened 26 Feb '25

26 Feb '25

alsa-project/alsa-ucm-conf pull request #513 was opened from nfraprado: Add support for MT8390 EVK with the SOF enabled. The only differences compared to the UCM with SOF disabled is the device number for playback and capture, which uses the PCMs for the SOF instead, and the number of channels for capture, which is stereo rather than mono. Define those parameters as variables for each case, with and without SOF, and move the common intialization to an init.conf, so the UCM can be shared by both. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/513 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/513.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Potential incorrect argument to snd_seq_port_info_set_ump_group in seqmid.c
by GitHub issues - opened 26 Feb '25

26 Feb '25

alsa-project/alsa-lib issue #440 was opened from reuk: https://github.com/alsa-project/alsa-lib/blob/df8f1cc1ec9d9ee15be5e2c23ad25… I think this call in seqmid.c is incorrect. According to the docs, the second argument to `snd_seq_port_info_set_ump_group` should be a 1-based group number (probably 0 for the UMP endpoint). I'm testing on Fedora 41 with alsa-lib-devel 1.2.13-3.fc41, and I'm seeing the following printed to the console when I call `snd_seq_create_ump_endpoint`: ``` ALSA lib seqmid.c:592:(snd_seq_create_ump_endpoint) Failed to create MIDI 2.0 port ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/440 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC fixes for v6.14-rc4
by Mark Brown 26 Feb '25

26 Feb '25

The following changes since commit 571b69f2f9b1ec7cf7d0e9b79e52115a87a869c4: ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device (2025-02-13 11:34:32 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc4 for you to fetch changes up to 9da0ed4a85027063441fa1c73967cafc38f0677c: ASoC: Intel: don't check number of sdw links when set (2025-02-25 17:47:04 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 More driver specific fixes, the firmware change is part of fixing the race conditions in the Cirrus driver. ---------------------------------------------------------------- Bard Liao (2): ASoC: SOF: Intel: don't check number of sdw links when set dmic_fixup ASoC: Intel: sof_sdw: warn both sdw and pch dmic are used Chancel Liu (1): ASoC: fsl: Rename stream name of SAI DAI driver Hector Martin (3): ASoC: tas2770: Fix volume scale ASoC: tas2764: Fix power control mask ASoC: tas2764: Set the SDOUT polarity correctly Mark Brown (1): ASoC: Intel: don't check number of sdw links when set Nicolas Frattaroli (2): ASoC: es8328: fix route from DAC to output ASoC: dapm-graph: set fill colour of turned on nodes Richard Fitzgerald (2): firmware: cs_dsp: Remove async regmap writes ASoC: cs35l56: Prevent races when soft-resetting using SPI control drivers/firmware/cirrus/cs_dsp.c | 24 +++--------- include/sound/cs35l56.h | 31 +++++++++++++++ sound/pci/hda/cs35l56_hda_spi.c | 3 ++ sound/soc/codecs/cs35l56-shared.c | 80 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l56-spi.c | 3 ++ sound/soc/codecs/es8328.c | 15 ++------ sound/soc/codecs/tas2764.c | 10 ++++- sound/soc/codecs/tas2764.h | 8 +++- sound/soc/codecs/tas2770.c | 2 +- sound/soc/fsl/fsl_sai.c | 6 +-- sound/soc/fsl/imx-audmix.c | 4 +- sound/soc/intel/boards/sof_sdw.c | 7 ++++ sound/soc/sof/intel/hda.c | 18 +-------- tools/sound/dapm-graph | 2 +- 14 files changed, 159 insertions(+), 54 deletions(-)

2 1

[GIT PULL] ASoC fixes for v6.14-rc3
by Mark Brown 26 Feb '25

26 Feb '25

The following changes since commit 571b69f2f9b1ec7cf7d0e9b79e52115a87a869c4: ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device (2025-02-13 11:34:32 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc3 for you to fetch changes up to f5468beeab1b1adfc63c2717b1f29ef3f49a5fab: ASoC: tas2764: Set the SDOUT polarity correctly (2025-02-18 20:08:43 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 A few fixes I and James Calligero picked out of the Asahi tree. ---------------------------------------------------------------- Hector Martin (3): ASoC: tas2770: Fix volume scale ASoC: tas2764: Fix power control mask ASoC: tas2764: Set the SDOUT polarity correctly sound/soc/codecs/tas2764.c | 10 +++++++++- sound/soc/codecs/tas2764.h | 8 +++++++- sound/soc/codecs/tas2770.c | 2 +- 3 files changed, 17 insertions(+), 3 deletions(-)

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[PATCH 0/2] ASoC: codecs: add ES8389 codec driver
by Zhang Yi 26 Feb '25

26 Feb '25

The driver is for codec ES8389 of everest-semi. Zhang Yi (2): ASoC: codecs: add support for ES8389 ASoC: dt-bindings: Add Everest ES8389 audio CODEC .../bindings/sound/everest,es8389.txt | 68 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8389.c | 977 ++++++++++++++++++ sound/soc/codecs/es8389.h | 139 +++ 5 files changed, 1191 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/everest,es8389.txt create mode 100755 sound/soc/codecs/es8389.c create mode 100755 sound/soc/codecs/es8389.h -- 2.17.1

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add rt713 & rt1318 LED control
by GitHub pull_request - opened 26 Feb '25

26 Feb '25

alsa-project/alsa-ucm-conf pull request #512 was opened from shumingfan: add rt713 & rt1318 LED control Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/512 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/512.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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