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

[PATCH] MAINTAINERS: update Tzung-Bi's email address
by Tzung-Bi Shih 11 Nov '25

11 Nov '25

Use kernel.org account instead. Signed-off-by: Tzung-Bi Shih <tzungbi(a)kernel.org> --- .../devicetree/bindings/sound/google,cros-ec-codec.yaml | 2 +- Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml | 2 +- MAINTAINERS | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml index c3e9f3485449..dea293f403d9 100644 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -8,7 +8,7 @@ title: Audio codec controlled by ChromeOS EC maintainers: - Cheng-Yi Chiang <cychiang(a)chromium.org> - - Tzung-Bi Shih <tzungbi(a)google.com> + - Tzung-Bi Shih <tzungbi(a)kernel.org> description: | Google's ChromeOS EC codec is a digital mic codec provided by the diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml index 1d73204451b1..ea7d4900ee4a 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Realtek rt1015p codec devicetree bindings maintainers: - - Tzung-Bi Shih <tzungbi(a)google.com> + - Tzung-Bi Shih <tzungbi(a)kernel.org> description: | Rt1015p is a rt1015 variant which does not support I2C and diff --git a/MAINTAINERS b/MAINTAINERS index cf0f18502372..f9749afc0b9d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4906,7 +4906,7 @@ F: drivers/platform/chrome/ CHROMEOS EC CODEC DRIVER M: Cheng-Yi Chiang <cychiang(a)chromium.org> -M: Tzung-Bi Shih <tzungbi(a)google.com> +M: Tzung-Bi Shih <tzungbi(a)kernel.org> R: Guenter Roeck <groeck(a)chromium.org> L: chrome-platform(a)lists.linux.dev S: Maintained -- 2.38.0.135.g90850a2211-goog

3 3

[alsa-devel] MOTU Midi Express XT
by holger＠dehnhardt.org 06 Nov '25

06 Nov '25

I own a MOTU Midi Express XT 8x8 interface which is connected through the parallel port. Recently I upgraded my system and the new motherboard does not have an ISA parallel port anymore... So I bought a PCI parallel port extension card - hoping that it will work with the ALSA driver (mtpav). So I changed the interrupt an port address in modules.conf but the driver is no longer recognized as an ALSA driver even though it loads without errors. As the device still perfectly fits my needs I would like to modify the driver to get it working again. Can anyone give me advice what should be changed in the drivers code to make it work on a pci port? Thanks Holger

3 12

[alsa-devel] Applied "ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF" to the asoc tree
by Mark Brown 29 Oct '25

29 Oct '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 v1] slimbus: qcom-ngd-ctrl: Add stream disable support
by Viken Dadhaniya 24 Sep '25

24 Sep '25

Currently slimbus driver doesn't support stream disable callback, it only supports stream enable callback. In slimbus usecase, client is switching to new frequency with same channel and calling enable stream callback for new frequency but DSP subsystem is crashing as we are switching to new frequency with same channel without disabling stream for older frequency. Ideally, before switching to another frequency, client should call disable stream callback and then enable stream for newer frequency. Hence add support to disable stream via qcom_slim_ngd_disable_stream(). Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 70 +++++++++++++++++++++++++++++++++ drivers/slimbus/slimbus.h | 13 ++++++ 2 files changed, 83 insertions(+) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index e0b21f0f79c1..d952827d2e12 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1,6 +1,7 @@ // SPDX-License-Identifier: GPL-2.0 // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved. // Copyright (c) 2018, Linaro Limited +// Copyright (c) 2024 Qualcomm Innovation Center, Inc. All rights reserved. #include <linux/irq.h> #include <linux/kernel.h> @@ -1084,6 +1085,74 @@ static int qcom_slim_ngd_enable_stream(struct slim_stream_runtime *rt) return ret; } +static int qcom_slim_ngd_disable_stream(struct slim_stream_runtime *rt) +{ + struct slim_device *sdev = rt->dev; + struct slim_controller *ctrl = sdev->ctrl; + struct slim_val_inf msg = {0}; + u8 wbuf[SLIM_MSGQ_BUF_LEN]; + u8 rbuf[SLIM_MSGQ_BUF_LEN]; + struct slim_msg_txn txn = {0,}; + int i, ret; + + txn.mt = SLIM_MSG_MT_DEST_REFERRED_USER; + txn.dt = SLIM_MSG_DEST_LOGICALADDR; + txn.la = SLIM_LA_MGR; + txn.ec = 0; + txn.msg = &msg; + txn.msg->num_bytes = 0; + txn.msg->wbuf = wbuf; + txn.msg->rbuf = rbuf; + + for (i = 0; i < rt->num_ports; i++) { + struct slim_port *port = &rt->ports[i]; + + if (txn.msg->num_bytes == 0) { + wbuf[txn.msg->num_bytes++] = (u8)(SLIM_CH_REMOVE << 6) + | (sdev->laddr & 0x1f); + + ret = slim_alloc_txn_tid(ctrl, &txn); + if (ret) { + dev_err(&sdev->dev, "Fail to allocate TID\n"); + return -ENXIO; + } + wbuf[txn.msg->num_bytes++] = txn.tid; + } + wbuf[txn.msg->num_bytes++] = port->ch.id; + } + + txn.mc = SLIM_USR_MC_CHAN_CTRL; + txn.rl = txn.msg->num_bytes + 4; + ret = qcom_slim_ngd_xfer_msg_sync(ctrl, &txn); + if (ret) { + slim_free_txn_tid(ctrl, &txn); + dev_err(&sdev->dev, "TX timed out:MC:0x%x,mt:0x%x ret:%d\n", + txn.mc, txn.mt, ret); + return ret; + } + + txn.mc = SLIM_USR_MC_RECONFIG_NOW; + txn.msg->num_bytes = 2; + wbuf[1] = sdev->laddr; + txn.rl = txn.msg->num_bytes + 4; + + ret = slim_alloc_txn_tid(ctrl, &txn); + if (ret) { + dev_err(ctrl->dev, "Fail to allocate TID ret:%d\n", ret); + return ret; + } + + wbuf[0] = txn.tid; + ret = qcom_slim_ngd_xfer_msg_sync(ctrl, &txn); + if (ret) { + slim_free_txn_tid(ctrl, &txn); + dev_err(&sdev->dev, "TX timed out:MC:0x%x,mt:0x%x ret:%d\n", + txn.mc, txn.mt, ret); + } + + return ret; +} + static int qcom_slim_ngd_get_laddr(struct slim_controller *ctrl, struct slim_eaddr *ea, u8 *laddr) { @@ -1642,6 +1711,7 @@ static int qcom_slim_ngd_ctrl_probe(struct platform_device *pdev) ctrl->ctrl.clkgear = SLIM_MAX_CLK_GEAR; ctrl->ctrl.get_laddr = qcom_slim_ngd_get_laddr; ctrl->ctrl.enable_stream = qcom_slim_ngd_enable_stream; + ctrl->ctrl.disable_stream = qcom_slim_ngd_disable_stream; ctrl->ctrl.xfer_msg = qcom_slim_ngd_xfer_msg; ctrl->ctrl.wakeup = NULL; ctrl->state = QCOM_SLIM_NGD_CTRL_DOWN; diff --git a/drivers/slimbus/slimbus.h b/drivers/slimbus/slimbus.h index 00a7f112574b..21543d125614 100644 --- a/drivers/slimbus/slimbus.h +++ b/drivers/slimbus/slimbus.h @@ -1,6 +1,7 @@ /* SPDX-License-Identifier: GPL-2.0 */ /* * Copyright (c) 2011-2017, The Linux Foundation + * Copyright (c) 2024 Qualcomm Innovation Center, Inc. All rights reserved. */ #ifndef _DRIVERS_SLIMBUS_H @@ -316,6 +317,18 @@ enum slim_transport_protocol { SLIM_PROTO_EXT_HALF_DUP, }; +/* + * enum slim_ch_control: Channel control. + * Activate will schedule channel and/or group of channels in the TDM frame. + * Suspend will keep the schedule but data-transfer won't happen. + * Remove will remove the channel/group from the TDM frame. + */ +enum slim_ch_control { + SLIM_CH_ACTIVATE, + SLIM_CH_SUSPEND, + SLIM_CH_REMOVE, +}; + /** * struct slim_stream_runtime - SLIMbus stream runtime instance * -- 2.17.1

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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 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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