November 2023 - Alsa-devel - mailman.alsa-project.org

ALSA 1.2.8 release
by Jaroslav Kysela 08 Dec '25

08 Dec '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 9

[alsa-devel] View alsa-lib plugins setup for any app?
by Eliot Blennerhassett 05 Dec '25

05 Dec '25

Greetings, running aplay -v will show the plugin chain setup. /proc/asound/card/pcm/* shows hw and sw params for the card driver when in use by any app Is there any way to show the alsa-lib plugin settings a-la aplay for an arbitrary application? thanks -- Eliot

4 8

[PATCH] ASoC: SOF: Intel: Retry codec probing if it fails
by Kai Vehmanen 04 Dec '25

04 Dec '25

From: Hui Wang <hui.wang(a)canonical.com> On the latest Lenovo Thinkstation laptops, we often experience the speaker failure after rebooting, check the dmesg, we could see: sof-audio-pci-intel-tgl 0000:00:1f.3: codec #0 probe error, ret: -5 The analogue codec on the machine is ALC287, then we designed a testcase to reboot and check the codec probing result repeatedly, we found the analogue codec probing always failed at least once within several minutes to several hours (roughly 1 reboot per min). This issue happens on all laptops of this Thinkstation model, but with legacy HDA driver, we couldn't reproduce this issue on those laptops. And so far, this issue is not reproduced on machines which don't belong to this model. We tried to make the hda_dsp_ctrl_init_chip() same as hda_intel_init_chip() which is the controller init routine in the legacy HDA driver, but it didn't help. We found when issue happens, the resp is -1, and if we let driver re-run send_cmd() and get_response(), it will get the correct response 10ec0287, then driver continues the rest work, finally boot to the desktop and all audio function work well. Here adding codec probing retries to 3 times, it could fix the issue on this Thinkstation model, and it doesn't bring impact to other machines. Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/intel/hda-codec.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) Notes: - patch for-5.16 on top of for-5.16 8a724d5f60 diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 6744318de612..13cd96e6724a 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -22,6 +22,7 @@ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) #define IDISP_VID_INTEL 0x80860000 +#define CODEC_PROBE_RETRIES 3 /* load the legacy HDA codec driver */ static int request_codec_module(struct hda_codec *codec) @@ -121,12 +122,15 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; u32 resp = -1; - int ret; + int ret, retry = 0; + + do { + mutex_lock(&hbus->core.cmd_mutex); + snd_hdac_bus_send_cmd(&hbus->core, hda_cmd); + snd_hdac_bus_get_response(&hbus->core, address, &resp); + mutex_unlock(&hbus->core.cmd_mutex); + } while (resp == -1 && retry++ < CODEC_PROBE_RETRIES); - mutex_lock(&hbus->core.cmd_mutex); - snd_hdac_bus_send_cmd(&hbus->core, hda_cmd); - snd_hdac_bus_get_response(&hbus->core, address, &resp); - mutex_unlock(&hbus->core.cmd_mutex); if (resp == -1) return -EIO; dev_dbg(sdev->dev, "HDA codec #%d probed OK: response: %x\n", base-commit: 8a724d5f60904084e09de2685ff9e77bc8a40815 -- 2.34.1

3 2

[alsa-devel] Applied "ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF" to the asoc tree
by Mark Brown 02 Dec '25

02 Dec '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

7 8

[PATCH] ASoC: wm8994: Prevent access to invalid VU register bits on WM1811
by Sylwester Nawrocki 20 Nov '25

20 Nov '25

The ADC2 and DAC2 are not available on WM1811 device. This patch moves the ADC2, DAC2 VU bitfields to a separate array so we can skip accessing them and avoid unreadable register access on WM1811. This allows to get rid of warnings during boot like: wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5 Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/codecs/wm8994.c | 60 ++++++++++++++++++++++++++++----------- 1 file changed, 44 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a84ae879d37e..038be667c1a6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -43,10 +43,12 @@ #define WM8994_NUM_DRC 3 #define WM8994_NUM_EQ 3 -static struct { +struct wm8994_reg_mask { unsigned int reg; unsigned int mask; -} wm8994_vu_bits[] = { +}; + +static struct wm8994_reg_mask wm8994_vu_bits[] = { { WM8994_LEFT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, { WM8994_RIGHT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, { WM8994_LEFT_LINE_INPUT_3_4_VOLUME, WM8994_IN2_VU }, @@ -60,14 +62,10 @@ static struct { { WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1DAC1_VU }, { WM8994_AIF1_DAC1_RIGHT_VOLUME, WM8994_AIF1DAC1_VU }, - { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU }, - { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU }, { WM8994_AIF2_DAC_LEFT_VOLUME, WM8994_AIF2DAC_VU }, { WM8994_AIF2_DAC_RIGHT_VOLUME, WM8994_AIF2DAC_VU }, { WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1ADC1_VU }, { WM8994_AIF1_ADC1_RIGHT_VOLUME, WM8994_AIF1ADC1_VU }, - { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU }, - { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, { WM8994_AIF2_ADC_LEFT_VOLUME, WM8994_AIF2ADC_VU }, { WM8994_AIF2_ADC_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, { WM8994_DAC1_LEFT_VOLUME, WM8994_DAC1_VU }, @@ -76,6 +74,14 @@ static struct { { WM8994_DAC2_RIGHT_VOLUME, WM8994_DAC2_VU }, }; +/* VU bitfields for ADC2, DAC2 not available on WM1811 */ +static struct wm8994_reg_mask wm8994_adc2_dac2_vu_bits[] = { + { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU }, + { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, +}; + static int wm8994_drc_base[] = { WM8994_AIF1_DRC1_1, WM8994_AIF1_DRC2_1, @@ -1030,6 +1036,26 @@ static bool wm8994_check_class_w_digital(struct snd_soc_component *component) return true; } +static void wm8994_update_vu_bits(struct snd_soc_component *component) +{ + struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); + struct wm8994 *control = wm8994->wm8994; + int i; + + for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) + snd_soc_component_write(component, wm8994_vu_bits[i].reg, + snd_soc_component_read(component, + wm8994_vu_bits[i].reg)); + if (control->type == WM1811) + return; + + for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++) + snd_soc_component_write(component, + wm8994_adc2_dac2_vu_bits[i].reg, + snd_soc_component_read(component, + wm8994_adc2_dac2_vu_bits[i].reg)); +} + static int aif_mclk_set(struct snd_soc_component *component, int aif, bool enable) { struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); @@ -1076,7 +1102,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); struct wm8994 *control = wm8994->wm8994; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; - int ret, i; + int ret; int dac; int adc; int val; @@ -1144,10 +1170,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) - snd_soc_component_write(component, wm8994_vu_bits[i].reg, - snd_soc_component_read(component, - wm8994_vu_bits[i].reg)); + wm8994_update_vu_bits(component); break; case SND_SOC_DAPM_PRE_PMD: @@ -1181,7 +1204,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - int ret, i; + int ret; int dac; int adc; int val; @@ -1237,10 +1260,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) - snd_soc_component_write(component, wm8994_vu_bits[i].reg, - snd_soc_component_read(component, - wm8994_vu_bits[i].reg)); + wm8994_update_vu_bits(component); break; case SND_SOC_DAPM_PRE_PMD: @@ -4346,6 +4366,14 @@ static int wm8994_component_probe(struct snd_soc_component *component) wm8994_vu_bits[i].mask, wm8994_vu_bits[i].mask); + if (control->type != WM1811) { + for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++) + snd_soc_component_update_bits(component, + wm8994_adc2_dac2_vu_bits[i].reg, + wm8994_adc2_dac2_vu_bits[i].mask, + wm8994_adc2_dac2_vu_bits[i].mask); + } + /* Set the low bit of the 3D stereo depth so TLV matches */ snd_soc_component_update_bits(component, WM8994_AIF1_DAC1_FILTERS_2, 1 << WM8994_AIF1DAC1_3D_GAIN_SHIFT, -- 2.17.1

4 3

[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

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

01 Aug '25

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

2 8

[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

2 2

[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

2 3

[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

2 23

[RESEND PATCH v2] drm/bridge: dw-hdmi-i2s: set insert_pcuv bit if hardware supports it
by Geraldo Nascimento 30 Dec '24

30 Dec '24

Hi Mark, resending this as it failed to apply in my last submission. Added Neil Armstrong to Cc: as hopefully he will be able to better review this. Thanks, Geraldo Nascimento --- Starting with version 2.10a of Synopsys DesignWare HDMI controller the insert_pcuv bit was introduced. On RK3399pro SoM (Radxa Rock Pi N10), for example, if we neglect to set this bit and proceed to enable hdmi_sound and i2s2 on the device tree there will be extreme clipping of sound output, to the point that music sounds like white noise. Problem could also manifest as just mild cracking depending of HDMI audio implementation of sink. Setting insert_pcuv bit (bit 2 of aud_conf2 Audio Sample register) fixes this. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- v1->v2: SoC->SoM on description, better commenting, minor style changes, conditional application of fix for L-PCM only --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c @@ -42,6 +42,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, struct dw_hdmi *hdmi = audio->hdmi; u8 conf0 = 0; u8 conf1 = 0; + u8 conf2 = 0; u8 inputclkfs = 0; /* it cares I2S only */ @@ -101,6 +102,28 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, return -EINVAL; } + /* + * dw-hdmi introduced insert_pcuv bit in + * version 2.10a. + * + * This single bit (bit 2 of HDMI_AUD_CONF2) + * when set to 1 will enable the insertion of the PCUV + * (Parity, Channel Status, User bit and Validity) + * bits on the incoming audio stream. + * + * Support is limited to Linear PCM audio. If + * neglected, the lack of valid PCUV bits + * on L-PCM streams will cause anything from + * mild cracking to full blown extreme + * clipping depending on the HDMI audio + * implementation of the sink. + * + */ + + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21 && + !(hparms->iec.status[0] & IEC958_AES0_NONAUDIO)) + conf2 = HDMI_AUD_CONF2_INSERT_PCUV; + dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); @@ -109,6 +120,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, inputclkfs, HDMI_AUD_INPUTCLKFS); hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + hdmi_write(audio, conf2, HDMI_AUD_CONF2); return 0; } --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h @@ -931,6 +931,11 @@ enum { HDMI_AUD_CONF1_WIDTH_16 = 0x10, HDMI_AUD_CONF1_WIDTH_24 = 0x18, +/* AUD_CONF2 field values */ + HDMI_AUD_CONF2_HBR = 0x01, + HDMI_AUD_CONF2_NLPCM = 0x02, + HDMI_AUD_CONF2_INSERT_PCUV = 0x04, + /* AUD_CTS3 field values */ HDMI_AUD_CTS3_N_SHIFT_OFFSET = 5, HDMI_AUD_CTS3_N_SHIFT_MASK = 0xe0,

3 6

hda: how to implement component master_unbind?
by Kai Vehmanen 20 Dec '24

20 Dec '24

Hi Takashi et al, I've been having multiple discussions with our i915 team w.r.t. audio driver behaviour when the aggregate component is unbound, triggered by i915 unbind. This came up again in the recent regression with devres allocations and I now dug into the topic again. In short, I'm puzzled how to really implement this. ALSA (or ASoC) don't really have support for individual components of a card disappearing in a hotplug fashion. At card level, we do have such support (USB, firewire and recent work for PCI hotplug). But not individual components of a card getting unplugged. In current code we have this: static void hdac_component_master_unbind(struct device *dev) { » struct drm_audio_component *acomp = hdac_get_acomp(dev); » if (acomp->audio_ops && acomp->audio_ops->master_unbind) » » acomp->audio_ops->master_unbind(dev, acomp); » module_put(acomp->ops->owner); » component_unbind_all(dev, acomp); » WARN_ON(acomp->ops || acomp->dev); } ... when e.g. i915 driver is unbound (but could be any driver using the component framework, not jus Intel hw), i915 calls component_del() and HDA gets call to the above. After this, acomp ops are null are audio no longer has ability to talk to i915 driver (which makes sense given it's unbound). If audio was runtime suspended, this kind of works (but relies on some good luck). Upon HDA controller resume, we note acomp ops are NULL and HDMI/DP operations (like snd_hdac_display_power()) silently become no-ops. PCM streaming will go to /dev/null, but this is ok'ish since no monitor can be connected anyways. If audio was active, we start to get warnings or worse. Audio expects hw display codec to be powered and programmed, but suddenly it mey lose state. If the audio controller is actually part of the display hardware (e.g. discrete GPUs), then controller registers can stop functioning (they should be still available, but given the main diplay driver is unbound, odds of suprising behaviour are high). So this is less than ideal. I've been looking at options: 1) prevent/block the unbind if audio device is busy The component framework does not allow individual components to return -EBUSY, so there's no way to let others know that unbind is not possible now. This would force anyone testing DRM driver unbind to first unbind the audio driver and stop any active audio use-cases if needed. 2) unbind the ALSA card I've experimented doing a device_unregister() from the above callback, but this has not really worked (onion peeling exercise of new probelms). The code is shared by multiple controllers, so getting a handle to an snd_card object is not straighforward (drvdata is differnet for SOF and snd-hda-intel for instance). But with some new work, maybe we could call snd_card_disconnect() to detach the card and inform user-space. 3) continue as-is and try to fix bugs If audio is active, maybe we could force a acomp->put_power() to ensure a clean unregister of the display part. But this leaves a big chunk of issues especially with HDA controllers that require the display to be powered on, to function. .. Any ideas, and/or has there been prior work? It seems Takashi it's been mostly you who has been active working on the acomp integration recently. I also included Chris, Daniel and Jani who've had patches to hdac_component.c. Br, Kai

3 5

[PATCH v2 0/3] ASoC: mediatek:mt8186: fix both the speaker and hdmi
by Jiaxin Yu 06 Dec '24

06 Dec '24

The speaker and hdmi of mt8186 platform are shared the same port of I2S, when connect the external display, use build-in speakers to play audio, they both playback at the same time. So we want to manage the playback device through DAPM events. Change since v1: Support audio function for it6505 bridge to hdmi-codec Jiaxin Yu (3): ASoC: hdmi-codec: Add event handler for hdmi TX ASoC: mediatek: mt8186: correct the HDMI widgets drm/bridge: it6505: Add audio support drivers/gpu/drm/bridge/ite-it6505.c | 81 +++++++++++++++++-- include/sound/hdmi-codec.h | 6 ++ sound/soc/codecs/hdmi-codec.c | 37 ++++++++- .../mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 5 files changed, 116 insertions(+), 12 deletions(-) -- 2.18.0

3 16

[alsa-devel] Error patching file pcm_native.c
by Delusion's Master 20 Nov '24

20 Nov '24

Hi, first of all I apologize in case you receive multiple copies of this message, but I forgot to specify a subject in the original mail. I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

3 10

[alsa-devel] [RFC PATCH 00/10] Add support for DMA and audio codec of F1C100s
by Mesih Kilinc 23 Oct '24

23 Oct '24

This is RFC patchset for Allwinner suniv F1C100s to support DMA and audio codec. Allwinner F1C100s has a audio codec that has necessary digital and analog parts. It has r-l headphone output and microphone, line, r-l FM inputs. ADC can capture any inputs and also output channels via mux. Any input channels or DAC samples can feed output channels. Add support for this audio codec. F1C100s utilizes DMA channels to send and receive ADC-DAC samples. So DMA support needed. Patch 1~5 adds support for DMA. Suniv F1C100s has very similar DMA to sun4i. But there is some dissimilarities also. Suniv features a DMA reset bit in clock control unit. It has smaller number of DMA channels. Several registers has different addresses. It's max burst size is 4 instead of 8. Also DMA endpoint numbers are different. Patch 6 adds DMA max burst option to sun4i-codec. Patch 7~8 Add support for suniv F1C100s audio codec. Patch 9 adds audio codec to suniv-f1c100s.dtsi Patch 10 adds audio codec support to Lichee Pi Nano board. Thanks! Mesih Kilinc (10): dma-engine: sun4i: Add a quirk to support different chips dma-engine: sun4i: Add has_reset option to quirk dt-bindings: dmaengine: Add Allwinner suniv F1C100s DMA dma-engine: sun4i: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for DMA ASoC: sun4i-codec: Add DMA Max Burst field dt-bindigs: sound: Add Allwinner suniv F1C100s Audio Codec ASoC: sun4i-codec: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for Audio Codec ARM: dts: suniv: f1c100s: Activate Audio Codec for Lichee Pi Nano .../devicetree/bindings/dma/sun4i-dma.txt | 4 +- .../devicetree/bindings/sound/sun4i-codec.txt | 5 + arch/arm/boot/dts/suniv-f1c100s-licheepi-nano.dts | 8 + arch/arm/boot/dts/suniv-f1c100s.dtsi | 25 ++ drivers/dma/Kconfig | 4 +- drivers/dma/sun4i-dma.c | 221 ++++++++++-- sound/soc/sunxi/sun4i-codec.c | 371 ++++++++++++++++++++- 7 files changed, 601 insertions(+), 37 deletions(-) -- 2.7.4

5 18

[PATCH v11 00/10] spi: Add support for stacked/parallel memories
by Amit Kumar Mahapatra 19 Aug '24

19 Aug '24

This patch series updated the spi-nor, spi core and the AMD-Xilinx GQSPI driver to add stacked and parallel memories support. The first nine patches https://lore.kernel.org/all/20230119185342.2093323-1-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-2-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-3-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-4-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-5-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-6-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-7-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-8-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-9-amit.kumar-mahapatra@a… of the previous series got applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/spi.git for-next But the rest of the patches in the series did not get applied as failure was reported for spi driver with GPIO CS, so send the remaining patches in the series after rebasing it on top of for-next branch and fixing the issue. CS GPIO is not tested on our hardware, but it has been tested by @Stefan https://lore.kernel.org/all/005001da1efc$619ad5a0$24d080e0$@opensource.cirr… --- BRANCH: for-next Changes in v11: - Rebased patch series on top of latest for-next branch. - Added a new patch(1/10) to replace spi->chip_select with spi_get_chipselect() call in tps6594-spi.c. - Added a new patch(2/10) to replace spi->chip_select with spi_get_chipseletc() call in cs35l56_hda_spi.c. - In spi.c initialized unused CS[] to 0xff and spi->cs_index_mask to 0x01 in all flows. - Updated spi_dev_check() to compare the CS of old spi device with the new spi device CS. - Updated cover letter description to add information regarding GPIO CS testing and added Stefen's Tested-by tag in 3/10 patch. Changes in v10: - Rebased patch series on top of latest for-next branch and fixed merge conflicts. Changes in v9: - Updated 1/8 patch description to add an high-level overview of parallel(multi-cs) & stacked design. - Initialized all unused CS to 0xFF. - Moved CS check from spi_add_device() to __spi_add_device(). - Updated __spi_add_device() to check to make sure that multiple logical CS don't map to the same physical CS and same physical CS doesn't map to different logical CS. - Updated 1/8, 5/8 & 7/8 to support arbitrary number of flash devices connected in parallel or stacked mode. - Updated documentation for chip_select. - Added a new spi-nor structure member nor->num_flash to keep track of the number of flashes connected. - Added a new patch in the series 4/8 to move write_enable call just before spi_mem ops call in SPI-NOR. - Added comments in SPI core & SPI-NOR. - Rebased the patch series on top of the latest for-next branch. Changes in v8: - Updated __spi_add_device() and spi_set_cs() to fix spi driver failure with GPIO CS. - Rebased the patch series on top of latest for-next branch and fixed merge conflicts. - Updated cover letter description to add information regarding GPIO CS testing and request Stefan to provide his Tested-by tag for 1/7 patch. - Updated 1/7 patch description. Changes in v7: - Updated spi_dev_check() to avoid failures for spi driver GPIO CS and moved the error message from __spi_add_device() to spi_dev_check(). - Resolved code indentation issue in spi_set_cs(). - In spi_set_cs() call spi_delay_exec( ) once if the controller supports multi cs with both the CS backed by GPIO. - Updated __spi_validate()to add checks for both the GPIO CS. - Replaced cs_index_mask bit mask with SPI_CS_CNT_MAX. - Updated struct spi_controller to represent multi CS capability of the spi controller through a flag bit SPI_CONTROLLER_MULTI_CS instead of a boolen structure member "multi_cs_cap". - Updated 1/7 patch description . Changes in v6: - Rebased on top of latest v6.3-rc1 and fixed merge conflicts in spi-mpc512x-psc.c, sfdp.c, spansion.c files and removed spi-omap-100k.c. - Updated spi_dev_check( ) to reject new devices if any one of the chipselect is used by another device. Changes in v5: - Rebased the patches on top of v6.3-rc1 and fixed the merge conflicts. - Fixed compilation warnings in spi-sh-msiof.c with shmobile_defconfig Changes in v4: - Fixed build error in spi-pl022.c file - reported by Mark. - Fixed build error in spi-sn-f-ospi.c file. - Added Reviewed-by: Serge Semin <fancer.lancer(a)gmail.com> tag. - Added two more patches to replace spi->chip_select with API calls in mpc832x_rdb.c & cs35l41_hda_spi.c files. Changes in v3: - Rebased the patches on top of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/spi.git for-next - Added a patch to convert spi_nor_otp_region_len(nor) & spi_nor_otp_n_regions(nor) macros into inline functions - Added Reviewed-by & Acked-by tags Changes in v2: - Rebased the patches on top of v6.2-rc1 - Created separate patch to add get & set APIs for spi->chip_select & spi->cs_gpiod, and replaced all spi->chip_select and spi->cs_gpiod references with the API calls. - Created separate patch to add get & set APIs for nor->params. --- Amit Kumar Mahapatra (10): mfd: tps6594: Use set/get APIs to access spi->chip_select ALSA: hda/cs35l56: Use set/get APIs to access spi->chip_select spi: Add multi-cs memories support in SPI core mtd: spi-nor: Convert macros with inline functions mtd: spi-nor: Add APIs to set/get nor->params mtd: spi-nor: Move write enable inside specific write & erase APIs mtd: spi-nor: Add stacked memories support in spi-nor spi: spi-zynqmp-gqspi: Add stacked memories support in GQSPI driver mtd: spi-nor: Add parallel memories support in spi-nor spi: spi-zynqmp-gqspi: Add parallel memories support in GQSPI driver drivers/mfd/tps6594-spi.c | 2 +- drivers/mtd/spi-nor/atmel.c | 17 +- drivers/mtd/spi-nor/core.c | 683 +++++++++++++++++++++++++------ drivers/mtd/spi-nor/core.h | 8 + drivers/mtd/spi-nor/debugfs.c | 4 +- drivers/mtd/spi-nor/gigadevice.c | 4 +- drivers/mtd/spi-nor/issi.c | 11 +- drivers/mtd/spi-nor/macronix.c | 10 +- drivers/mtd/spi-nor/micron-st.c | 35 +- drivers/mtd/spi-nor/otp.c | 48 ++- drivers/mtd/spi-nor/sfdp.c | 33 +- drivers/mtd/spi-nor/spansion.c | 57 +-- drivers/mtd/spi-nor/sst.c | 7 +- drivers/mtd/spi-nor/swp.c | 25 +- drivers/mtd/spi-nor/winbond.c | 2 +- drivers/mtd/spi-nor/xilinx.c | 18 +- drivers/spi/spi-zynqmp-gqspi.c | 58 ++- drivers/spi/spi.c | 259 ++++++++++-- include/linux/mtd/spi-nor.h | 21 +- include/linux/spi/spi.h | 51 ++- sound/pci/hda/cs35l56_hda_spi.c | 2 +- 21 files changed, 1085 insertions(+), 270 deletions(-) -- 2.17.1

10 61

[alsa-devel] unsupported variant of snd_hda_intel
by Alexandre 04 Jul '24

04 Jul '24

Hello, I have an Acer Aspire 4736G laptop. The soundcard that is shipped in this laptop is intel-hda based as shown (actually it's a realtek): cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x9b300000 irq 46 Basic sound output is correct but few feature are missing: - internal mic seems to be found but do not works. - sound output stop working when trying to send it using HDMI output - heads do not work when plugged neither when forcing pulseaudio to use it. I have an ubuntu 10.10 which is shipped with alsa 1.0.23. I have googled around and tried loading the snd-hda-intel with several parameters () as found in HD-Audio-Models.txt to no avail. cat /proc/asound/pcm 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 00-01: ALC888 Digital : ALC888 Digital : playback 1 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 Where can I begin to help implement the support for this sound card? I have very skills in developement in general but am ready to try if it is not to hard (I supose I can use previous code for similar models).. I just need a shepherd. Thanks

4 4

[PATCH v4 0/7] Patches to update for rockchip pdm
by Sugar Zhang 23 Apr '24

23 Apr '24

These patches fixup or update for rockchip pdm. Changes in v4: - Acked by Rob Herring Changes in v3: - Fix property 'path-map' suggested by Rob Herring. Changes in v2: - Fix yamllint errors. Sugar Zhang (7): ASoC: rockchip: Add support for rv1126 pdm ASoC: dt-bindings: rockchip: Add binding for rv1126 pdm ASoC: rockchip: pdm: Add support for rk3568 pdm ASoC: dt-bindings: rockchip: Add binding for rk3568 pdm ASoC: rockchip: pdm: Add support for path map ASoC: dt-bindings: rockchip: pdm: Document property 'rockchip,path-map' ASoC: dt-bindings: rockchip: Convert pdm bindings to yaml .../devicetree/bindings/sound/rockchip,pdm.txt | 46 -------- .../devicetree/bindings/sound/rockchip,pdm.yaml | 120 +++++++++++++++++++++ sound/soc/rockchip/rockchip_pdm.c | 112 +++++++++++++++++-- sound/soc/rockchip/rockchip_pdm.h | 6 ++ 4 files changed, 232 insertions(+), 52 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.yaml -- 2.7.4

3 9

[PATCH 0/2] ALSA/ASoC/SOF: SSP dev_type support
by Peter Ujfalusi 26 Mar '24

26 Mar '24

Hi, SOF always passed 0 as dev_type to intel_nhlt_get_endpoint_blob() when looking up the blob from the NHLT table. This causes issues since alsa-utils commit 3a47ef2487ed ("topology: nhlt: intel: support more device types and directions") The dev_type is no longer always 0 in the topology embedded NHLT table resulting lookup failures for analog codecs since they will have dev_type=4. With SOF the dev_type is not used, we always use the SSP port index for looking up the configuration blob. The solution for the issue is to fetch the dev_type for the SSP port and feed it back to the intel_nhlt_get_endpoint_blob() to be able to find the configuration regardless of the assigned dev_type. Regards, Peter --- Brent Lu (2): ALSA: hda: intel-nhlt: add intel_nhlt_ssp_device_type() function ASoC: SOF: ipc4-topology: support NHLT device type include/sound/intel-nhlt.h | 10 ++++++++++ sound/hda/intel-nhlt.c | 26 ++++++++++++++++++++++++++ sound/soc/sof/ipc4-topology.c | 19 ++++++++++++++++--- 3 files changed, 52 insertions(+), 3 deletions(-) -- 2.43.0

3 6

[PATCH v9 00/27] Add support for QMC HDLC, framer infrastructure and PEF2256 framer
by Herve Codina 14 Mar '24

14 Mar '24

Hi, I have a system where I need to handle an HDLC interface and some audio data. The HDLC data are transferred using a TDM bus on which a PEF2256 (E1/T1 framer) is present. The PEF2256 transfers data from/to the TDM bus to/from the E1 line. This PEF2256 is connected to a PowerQUICC SoC for the control path and the TDM is connected to the SoC (QMC component) for the data path. From the QMC HDLC driver, I need to handle HDLC data using the QMC, carrier detection using the PEF2256 (E1 line carrier) and set/get some PEF2256 configuration. The QMC HDLC driver considers the PEF2256 as a generic framer. It performs operations that involve the PEF2256 through the generic framer API. The audio data are exchanged with the PEF2256 using a CPU DAI connected to the TDM bus through the QMC and the PEF2256 needs to be seen as a codec in order to be linked to the CPU DAI. The codec handles the carrier detection using the PEF2256 and reports the carrier state to the ALSA subsystem using the ASoC jack detection. The codec, even if instantiated by the PEF2256 driver, considers the PEF2256 as a generic framer. The generic framer has: - 2 consumers (QMC HDLC drv and codec) - 1 provider (PEF2256) So, the design is the following: +------------------+ +---------+ | QMC | <- TDM -> | PEF2256 | <-> E1 +---------+ | +-------------+ | | | | CPU DAI | <-data--> | QMC channel | | | | +---------+ | +-------------+ | | | +--------------+ | +-------------+ | | | | QMC HDLC drv | <-data--> | QMC channel | | | | +--------------+ | +-------------+ | | | ^ +------------------+ | | | +--------+ +-------------+ | | +-> | framer | <-> | PEF2256 drv | <- local bus ->| | | | | | +---------+ +-> | | | | | +--------+ | +-------+ | +-------------------> | codec | | | +-------+ | +-------------+ Further more, the TDM timeslots used by the QMC HDLC driver need to be configured at runtime (QMC dynamic timeslots). Several weeks ago, I sent two series related to this topic: - Add the Lantiq PEF2256 audio support [1] - RFC Add support for QMC HDLC and PHY [2] This current series is a rework of these two series taking into account feedbacks previously received. In order to implement all of this, I do the following: 1) Perform some fixes (patches 1, 2, 3, 4, 5 ) 2) Introduce the QMC HDLC driver (patches 6, 7, 8) 3) Add QMC dynamic timeslot support (patches 9 - 19) 4) Add timeslots change support in QMC HDLC (patch 20) 5) Introduce framer infrastructure (patch 21) 6) Add PEF2256 framer provider (patches 22, 23, 24, 25) 7) Add framer codec as a framer consumer (patch 26) 8) Add framer support as a framer consumer in QMC HDLC (patch 27) The series contains the full story and detailed modifications. If needed, the series can be split and/or commits can be squashed. Let me know. Compare to the previous iteration https://lore.kernel.org/linux-kernel/20231011061437.64213-1-herve.codina@bo… This v9 series: - Removes some patches already applied - Adds some 'Acked-by' tags Best regards, Hervé [1]: https://lore.kernel.org/all/20230417171601.74656-1-herve.codina@bootlin.com/ [2]: https://lore.kernel.org/all/20230323103154.264546-1-herve.codina@bootlin.co… Changes v8 -> v9 - Removed Patches 6, 7 and 8 (patches applied) - Patches 7, 20, 21, 23 (patches 10, 23, 24, 26 in v8) Add 'Acked-by: Jakub Kicinski <kuba(a)kernel.org>' Changes v7 -> v8 - Patch 10 Fix a race condition when stopping the queue in qmc_hdlc_close() - Patch 24 Move to menuconfig and hide CONFIG_GENERIC_FRAMER Remove unneeded check (defensive programming) Remove unneeded variable assignment - Patch 25 Add 'Reviewed-by: Rob Herring <robh(a)kernel.org>' - Patch 26 Use array notation for 'audio_devs' Changes v6 -> v7 - Patch 8 Move the 'compatible' property to the first property. Add device/SoC specific compatible strings. Add 'Reviewed-by: Rob Herring <robh(a)kernel.org>' - Patch 25 Remove '$ref' from the lantiq,data-rate-bps property. '-bps' is now a standard suffix. Changes v5 -> v6 - All patches Remove 'Signed-off-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' These Sob were added by Christophe when he sent the v4 while I was on vacation. https://lore.kernel.org/linux-kernel/992a2b31-e21f-eee3-8bfc-a65b69fe5bd7@c… - Patch 6, 7 Add 'Acked-by: Conor Dooley <conor.dooley(a)microchip.com>' Add 'Acked-by: Rob Herring <robh(a)kernel.org>' - Patch 8 Add a constraint on fsl,framer. - Patch 25 Fix clocks description (inconsistent capitalisation) Use '8-bit' instead of '8bit' Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Removed patch 26 (patch applied) - Patch 26 (patch 27 in v5) Add 'Reviewed-by: Linus Walleij <linus.walleij(a)linaro.org>' - Patch 27 (patch 28 in v5) Move registers definition to .c file and remove .h file Add 'select PINCONF' in Kconfig Use dev_err_probe() Add 'Reviewed-by: Linus Walleij <linus.walleij(a)linaro.org>' - Patch 28 (patch 29 in v5) Remove .h file Changes v4 -> v5 - Patches 1 to 5 No changes - Patch 6 (new in v5) Fix QMC binding example - Patch 7 (new in v5) Add missing 'additionalProperties: false' - Patch 8 (new in v5, replace v4 patch 6) Add QMC HDLC properties in the QMC channel node Renamed the 'framer' property to 'fsl,framer' - Patch 9 (new in v5) Add support for QMC child devices - Patch 10 (patch 7 in v4) No changes - Patch 11 (patch 8 in v4) Remove fsl,qmc-hdlc.yaml (no more existing file) - Patches 12 to 22 (patches 9 to 19 in v4) No changes - Patch 23 (patch 20 in v4) Remove unused variable initializations Remove extra space - Patch 24 (patch 21 in v4) Improve Kconfig help text Fix variable declaration (reverse xmas tree) Fix typos and extra spaces Fix documentation issues raised by 'kernel-doc -none' Move of_node_put() and kfree() out of the mutex Replace ida_simple_{get,remove}() by ida_{alloc,free}() Support framer device-tree nodes without '#framer-cells' property - Patch 25 (patch 22 in v4) Fix $ref in the pinctrl subnode Remove '#framer-cells' property Add needed '|' - Patch 26 (patch 23 in v4) Fix a typo in the commit subject - Patches 27, 28, 29 (patch 24, 25, 26 in v4) No changes - Patch 30 (patch 27 in v4) Fix a typo in the commit log - Patch 31 (patch 28 in v4) Used 'fsl,framer' property name instead of 'framer' Changes v3 -> v4 - Patch 21 Fixes build failure with CONFIG_MODULES Changes v2 -> v3 - Patches 1, 2, 3, 4 Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - New patch Remove inline keyword from the existing registers accessors helpers - Patch 6 (patches 5, 27 in v2) Update the binding title Squash patch 27 - Patch 7 (patch 6 in v2) Remove the cast in netdev_to_qmc_hdlc() Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 8 (patch 7 in v2): No change - Patches 9, 10 (patches 8, 9 in v2) Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 11 (patch 10 in v2) Remove inline keyword from the introduced qmc_clrsetbits16() helper Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patches 12, 13, 14, 15, 16, 17, 18, 19, 20 Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 21 (patch 20 in v2) Remove unneeded framer NULL pointer check Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 22 (patch 21 in v2) Change sclkr and sclkx clocks description Remove the framer phandle property from the framer subnodes (ie. from framer-codec nodes) - Patch 23 (patch 22 in v2) Initialize 'disabled' variable at declaration Fix commit log Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 24 (patch 23 in v2) Remove inline keyword from the existing registers accessors helpers Use dev_warn_ratelimited() in default interrupt handler Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 25 (patch 24 in v2) Replace #include "linux/bitfield.h" by #include <linux/bitfield.h> Fold the pinctrl anonymous struct into the struct pef2256_pinctrl Update commit log Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 26 (patch 25 in v2) Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 27 (patch 26 in v2) Fix error message Changed the ch.max computation in framer_dai_hw_rule_channels_by_format() Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 28 Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' Changes v1 -> v2 - Patches 1, 2 (New in v2) Fix __iomem addresses declaration - Patch 19 (17 in v1) Fix a compilation warning - Patch 26 (24 in v1) Fix a typo in Kconfig file Fix issues raised by sparse (make C=1) Herve Codina (27): soc: fsl: cpm1: tsa: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix rx channel reset soc: fsl: cpm1: qmc: Extend the API to provide Rx status soc: fsl: cpm1: qmc: Remove inline function specifiers soc: fsl: cpm1: qmc: Add support for child devices net: wan: Add support for QMC HDLC MAINTAINERS: Add the Freescale QMC HDLC driver entry soc: fsl: cpm1: qmc: Introduce available timeslots masks soc: fsl: cpm1: qmc: Rename qmc_setup_tsa* to qmc_init_tsa* soc: fsl: cpm1: qmc: Introduce qmc_chan_setup_tsa* soc: fsl: cpm1: qmc: Remove no more needed checks from qmc_check_chans() soc: fsl: cpm1: qmc: Check available timeslots in qmc_check_chans() soc: fsl: cpm1: qmc: Add support for disabling channel TSA entries soc: fsl: cpm1: qmc: Split Tx and Rx TSA entries setup soc: fsl: cpm1: qmc: Introduce is_tsa_64rxtx flag soc: fsl: cpm1: qmc: Handle timeslot entries at channel start() and stop() soc: fsl: cpm1: qmc: Remove timeslots handling from setup_chan() soc: fsl: cpm1: qmc: Introduce functions to change timeslots at runtime wan: qmc_hdlc: Add runtime timeslots changes support net: wan: Add framer framework support dt-bindings: net: Add the Lantiq PEF2256 E1/T1/J1 framer net: wan: framer: Add support for the Lantiq PEF2256 framer pinctrl: Add support for the Lantic PEF2256 pinmux MAINTAINERS: Add the Lantiq PEF2256 driver entry ASoC: codecs: Add support for the framer codec net: wan: fsl_qmc_hdlc: Add framer support .../bindings/net/lantiq,pef2256.yaml | 213 +++++ MAINTAINERS | 15 + drivers/net/wan/Kconfig | 14 + drivers/net/wan/Makefile | 3 + drivers/net/wan/framer/Kconfig | 41 + drivers/net/wan/framer/Makefile | 7 + drivers/net/wan/framer/framer-core.c | 882 ++++++++++++++++++ drivers/net/wan/framer/pef2256/Makefile | 8 + drivers/net/wan/framer/pef2256/pef2256-regs.h | 250 +++++ drivers/net/wan/framer/pef2256/pef2256.c | 880 +++++++++++++++++ drivers/net/wan/fsl_qmc_hdlc.c | 820 ++++++++++++++++ drivers/pinctrl/Kconfig | 15 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/pinctrl-pef2256.c | 358 +++++++ drivers/soc/fsl/qe/qmc.c | 592 +++++++++--- drivers/soc/fsl/qe/tsa.c | 22 +- include/linux/framer/framer-provider.h | 194 ++++ include/linux/framer/framer.h | 205 ++++ include/linux/framer/pef2256.h | 31 + include/soc/fsl/qe/qmc.h | 27 +- sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/framer-codec.c | 413 ++++++++ sound/soc/fsl/fsl_qmc_audio.c | 2 +- 24 files changed, 4873 insertions(+), 137 deletions(-) create mode 100644 Documentation/devicetree/bindings/net/lantiq,pef2256.yaml create mode 100644 drivers/net/wan/framer/Kconfig create mode 100644 drivers/net/wan/framer/Makefile create mode 100644 drivers/net/wan/framer/framer-core.c create mode 100644 drivers/net/wan/framer/pef2256/Makefile create mode 100644 drivers/net/wan/framer/pef2256/pef2256-regs.h create mode 100644 drivers/net/wan/framer/pef2256/pef2256.c create mode 100644 drivers/net/wan/fsl_qmc_hdlc.c create mode 100644 drivers/pinctrl/pinctrl-pef2256.c create mode 100644 include/linux/framer/framer-provider.h create mode 100644 include/linux/framer/framer.h create mode 100644 include/linux/framer/pef2256.h create mode 100644 sound/soc/codecs/framer-codec.c -- 2.41.0

6 36

[PATCH v7 0/5] ASoC: makes CPU/Codec channel connection map more generic
by Kuninori Morimoto 14 Mar '24

14 Mar '24

Hi Mark Cc Bard, Pierre-Louis, Jerome, DT-ML This is v7 patch-set. Current ASoC is supporting CPU/Codec = N:M (N < M) connection by using ch_map idea. This patch-set expands it that all connection uses this idea, and no longer N < M limit [1][2]. Link: https://lore.kernel.org/r/87fs6wuszr.wl-kuninori.morimoto.gx@renesas.com [1] Link: https://lore.kernel.org/r/878r7yqeo4.wl-kuninori.morimoto.gx@renesas.com [2] ASoC core code ([PATCH 1/5]) is same as v6 and it was tested by Pierre-Louis, and Jerome. Big change on v7 is basically for Audio-Graph-Card2. v6 -> v7 - use "endpoint" for N:M connection instead of ch-map-idx on Audio-Graph-Card2 v5 -> v6 - tidyup some warnings - rename "ch-map-idx" -> "channel-map-index" - Update "channel-map-index" description - add Tested-by from Pierre-Louis / Jerome v4 -> v5 - use cpu/codec index on ch_maps - separate card2 sample DT patch into prepare and new feature - ch-maps -> ch-map-idx on DT v3 -> v4 - add Jerome on To - add "description" on "ch-maps" v2 -> v3 - tidyup comment - use more clear connection image on DT - "ch_maps" -> "ch-maps" on DT - Add DT maintainer on "To:" for all patches v1 -> v2 - makes CPU/Codec connection relation clear on comment/sample - fixup type "connction" -> "connection" - makes error message clear Kuninori Morimoto (5): ASoC: makes CPU/Codec channel connection map more generic ASoC: audio-graph-card2: use better image for Multi connection ASoC: audio-graph-card2: add CPU:Codec = N:M support ASoC: audio-graph-card2-custom-sample: Add connection image ASoC: audio-graph-card2-custom-sample: add CPU/Codec = N:M sample include/sound/soc.h | 56 ++- .../audio-graph-card2-custom-sample.dtsi | 380 ++++++++++++++++-- sound/soc/generic/audio-graph-card2.c | 277 ++++++++++--- sound/soc/intel/boards/sof_sdw.c | 28 +- sound/soc/soc-core.c | 95 ++++- sound/soc/soc-dapm.c | 45 +-- sound/soc/soc-pcm.c | 44 +- 7 files changed, 754 insertions(+), 171 deletions(-) -- 2.25.1

3 15

[PATCH v3] powerpc: macio: Make remove callback of macio driver void returned
by Dawei Li 13 Mar '24

13 Mar '24

Commit fc7a6209d571 ("bus: Make remove callback return void") forces bus_type::remove be void-returned, it doesn't make much sense for any bus based driver implementing remove callbalk to return non-void to its caller. This change is for macio bus based drivers. Signed-off-by: Dawei Li <set_pte_at(a)outlook.com> --- v2 -> v3 - Rebased on latest powerpc/next. - cc' to relevant subsysem lists. v1 -> v2 - Revert unneeded changes. - Rebased on latest powerpc/next. v1 - https://lore.kernel.org/all/TYCP286MB2323FCDC7ECD87F8D97CB74BCA189@TYCP286M… --- arch/powerpc/include/asm/macio.h | 2 +- drivers/ata/pata_macio.c | 4 +--- drivers/macintosh/rack-meter.c | 4 +--- drivers/net/ethernet/apple/bmac.c | 4 +--- drivers/net/ethernet/apple/mace.c | 4 +--- drivers/net/wireless/intersil/orinoco/airport.c | 4 +--- drivers/scsi/mac53c94.c | 5 +---- drivers/scsi/mesh.c | 5 +---- drivers/tty/serial/pmac_zilog.c | 7 ++----- sound/aoa/soundbus/i2sbus/core.c | 4 +--- 10 files changed, 11 insertions(+), 32 deletions(-) diff --git a/arch/powerpc/include/asm/macio.h b/arch/powerpc/include/asm/macio.h index ff5fd82d9ff0..cb9c386dacf8 100644 --- a/arch/powerpc/include/asm/macio.h +++ b/arch/powerpc/include/asm/macio.h @@ -125,7 +125,7 @@ static inline struct pci_dev *macio_get_pci_dev(struct macio_dev *mdev) struct macio_driver { int (*probe)(struct macio_dev* dev, const struct of_device_id *match); - int (*remove)(struct macio_dev* dev); + void (*remove)(struct macio_dev *dev); int (*suspend)(struct macio_dev* dev, pm_message_t state); int (*resume)(struct macio_dev* dev); diff --git a/drivers/ata/pata_macio.c b/drivers/ata/pata_macio.c index 9ccaac9e2bc3..653106716a4b 100644 --- a/drivers/ata/pata_macio.c +++ b/drivers/ata/pata_macio.c @@ -1187,7 +1187,7 @@ static int pata_macio_attach(struct macio_dev *mdev, return rc; } -static int pata_macio_detach(struct macio_dev *mdev) +static void pata_macio_detach(struct macio_dev *mdev) { struct ata_host *host = macio_get_drvdata(mdev); struct pata_macio_priv *priv = host->private_data; @@ -1202,8 +1202,6 @@ static int pata_macio_detach(struct macio_dev *mdev) ata_host_detach(host); unlock_media_bay(priv->mdev->media_bay); - - return 0; } #ifdef CONFIG_PM_SLEEP diff --git a/drivers/macintosh/rack-meter.c b/drivers/macintosh/rack-meter.c index c28893e41a8b..f2f83c4f3af5 100644 --- a/drivers/macintosh/rack-meter.c +++ b/drivers/macintosh/rack-meter.c @@ -523,7 +523,7 @@ static int rackmeter_probe(struct macio_dev* mdev, return rc; } -static int rackmeter_remove(struct macio_dev* mdev) +static void rackmeter_remove(struct macio_dev *mdev) { struct rackmeter *rm = dev_get_drvdata(&mdev->ofdev.dev); @@ -558,8 +558,6 @@ static int rackmeter_remove(struct macio_dev* mdev) /* Get rid of me */ kfree(rm); - - return 0; } static int rackmeter_shutdown(struct macio_dev* mdev) diff --git a/drivers/net/ethernet/apple/bmac.c b/drivers/net/ethernet/apple/bmac.c index 9e653e2925f7..292b1f9cd9e7 100644 --- a/drivers/net/ethernet/apple/bmac.c +++ b/drivers/net/ethernet/apple/bmac.c @@ -1591,7 +1591,7 @@ bmac_proc_info(char *buffer, char **start, off_t offset, int length) } #endif -static int bmac_remove(struct macio_dev *mdev) +static void bmac_remove(struct macio_dev *mdev) { struct net_device *dev = macio_get_drvdata(mdev); struct bmac_data *bp = netdev_priv(dev); @@ -1609,8 +1609,6 @@ static int bmac_remove(struct macio_dev *mdev) macio_release_resources(mdev); free_netdev(dev); - - return 0; } static const struct of_device_id bmac_match[] = diff --git a/drivers/net/ethernet/apple/mace.c b/drivers/net/ethernet/apple/mace.c index fd1b008b7208..e6350971c707 100644 --- a/drivers/net/ethernet/apple/mace.c +++ b/drivers/net/ethernet/apple/mace.c @@ -272,7 +272,7 @@ static int mace_probe(struct macio_dev *mdev, const struct of_device_id *match) return rc; } -static int mace_remove(struct macio_dev *mdev) +static void mace_remove(struct macio_dev *mdev) { struct net_device *dev = macio_get_drvdata(mdev); struct mace_data *mp; @@ -296,8 +296,6 @@ static int mace_remove(struct macio_dev *mdev) free_netdev(dev); macio_release_resources(mdev); - - return 0; } static void dbdma_reset(volatile struct dbdma_regs __iomem *dma) diff --git a/drivers/net/wireless/intersil/orinoco/airport.c b/drivers/net/wireless/intersil/orinoco/airport.c index a890bfa0d5cc..276a06cdd1f5 100644 --- a/drivers/net/wireless/intersil/orinoco/airport.c +++ b/drivers/net/wireless/intersil/orinoco/airport.c @@ -85,7 +85,7 @@ airport_resume(struct macio_dev *mdev) return err; } -static int +static void airport_detach(struct macio_dev *mdev) { struct orinoco_private *priv = dev_get_drvdata(&mdev->ofdev.dev); @@ -111,8 +111,6 @@ airport_detach(struct macio_dev *mdev) macio_set_drvdata(mdev, NULL); free_orinocodev(priv); - - return 0; } static int airport_hard_reset(struct orinoco_private *priv) diff --git a/drivers/scsi/mac53c94.c b/drivers/scsi/mac53c94.c index f75928f7773e..42648ca9b8ed 100644 --- a/drivers/scsi/mac53c94.c +++ b/drivers/scsi/mac53c94.c @@ -508,7 +508,7 @@ static int mac53c94_probe(struct macio_dev *mdev, const struct of_device_id *mat return rc; } -static int mac53c94_remove(struct macio_dev *mdev) +static void mac53c94_remove(struct macio_dev *mdev) { struct fsc_state *fp = (struct fsc_state *)macio_get_drvdata(mdev); struct Scsi_Host *host = fp->host; @@ -526,11 +526,8 @@ static int mac53c94_remove(struct macio_dev *mdev) scsi_host_put(host); macio_release_resources(mdev); - - return 0; } - static struct of_device_id mac53c94_match[] = { { diff --git a/drivers/scsi/mesh.c b/drivers/scsi/mesh.c index 84b541a57b7b..cd2575b88c85 100644 --- a/drivers/scsi/mesh.c +++ b/drivers/scsi/mesh.c @@ -1986,7 +1986,7 @@ static int mesh_probe(struct macio_dev *mdev, const struct of_device_id *match) return -ENODEV; } -static int mesh_remove(struct macio_dev *mdev) +static void mesh_remove(struct macio_dev *mdev) { struct mesh_state *ms = (struct mesh_state *)macio_get_drvdata(mdev); struct Scsi_Host *mesh_host = ms->host; @@ -2013,11 +2013,8 @@ static int mesh_remove(struct macio_dev *mdev) macio_release_resources(mdev); scsi_host_put(mesh_host); - - return 0; } - static struct of_device_id mesh_match[] = { { diff --git a/drivers/tty/serial/pmac_zilog.c b/drivers/tty/serial/pmac_zilog.c index 13668ffdb1e7..d4640479c338 100644 --- a/drivers/tty/serial/pmac_zilog.c +++ b/drivers/tty/serial/pmac_zilog.c @@ -1507,12 +1507,12 @@ static int pmz_attach(struct macio_dev *mdev, const struct of_device_id *match) * That one should not be called, macio isn't really a hotswap device, * we don't expect one of those serial ports to go away... */ -static int pmz_detach(struct macio_dev *mdev) +static void pmz_detach(struct macio_dev *mdev) { struct uart_pmac_port *uap = dev_get_drvdata(&mdev->ofdev.dev); if (!uap) - return -ENODEV; + return; uart_remove_one_port(&pmz_uart_reg, &uap->port); @@ -1523,11 +1523,8 @@ static int pmz_detach(struct macio_dev *mdev) dev_set_drvdata(&mdev->ofdev.dev, NULL); uap->dev = NULL; uap->port.dev = NULL; - - return 0; } - static int pmz_suspend(struct macio_dev *mdev, pm_message_t pm_state) { struct uart_pmac_port *uap = dev_get_drvdata(&mdev->ofdev.dev); diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 51ed2f34b276..35f39727994d 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -364,15 +364,13 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) return 0; } -static int i2sbus_remove(struct macio_dev* dev) +static void i2sbus_remove(struct macio_dev *dev) { struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev); struct i2sbus_dev *i2sdev, *tmp; list_for_each_entry_safe(i2sdev, tmp, &control->list, item) soundbus_remove_one(&i2sdev->sound); - - return 0; } #ifdef CONFIG_PM -- 2.25.1

6 6

[PATCH] soundwire: Use snd_soc_substream_to_rtd() to obtain rtd
by Amadeusz Sławiński 03 Mar '24

03 Mar '24

From: Cezary Rojewski <cezary.rojewski(a)intel.com> Utilize the helper function instead of casting from ->private_data directly. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- drivers/soundwire/stream.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 69719b335bcb..652aae4e221d 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1717,7 +1717,7 @@ EXPORT_SYMBOL(sdw_deprepare_stream); static int set_stream(struct snd_pcm_substream *substream, struct sdw_stream_runtime *sdw_stream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *dai; int ret = 0; int i; @@ -1770,7 +1770,7 @@ EXPORT_SYMBOL(sdw_alloc_stream); int sdw_startup_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; char *name; int ret; @@ -1814,7 +1814,7 @@ EXPORT_SYMBOL(sdw_startup_stream); void sdw_shutdown_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; struct snd_soc_dai *dai; -- 2.34.1

2 2

[PATCH] ALSA: dice: add stream format parameters for Weiss devices
by Michele Perrone 29 Feb '24

29 Feb '24

Hard-coded stream format parameters are added for Weiss Engineering FireWire devices. When the device vendor and model match, the parameters are copied into the stream format cache. This allows for setting all supported sampling rates up to 192kHz, and consequently adjusting the number of available I/O channels. Signed-off-by: Rolf Anderegg <rolf.anderegg(a)weiss.ch> Signed-off-by: Michele Perrone <michele.perrone(a)weiss.ch> --- sound/firewire/dice/Makefile | 2 +- sound/firewire/dice/dice-weiss.c | 120 +++++++++++++++++++++++++++++++ sound/firewire/dice/dice.c | 72 +++++++++++++++++++ 3 files changed, 193 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/dice/dice-weiss.c diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index a5f3fbf28b8c..bac8712f9014 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -2,5 +2,5 @@ snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \ dice-alesis.o dice-extension.o dice-mytek.o dice-presonus.o \ - dice-harman.o dice-focusrite.o + dice-harman.o dice-focusrite.o dice-weiss.o obj-$(CONFIG_SND_DICE) += snd-dice.o diff --git a/sound/firewire/dice/dice-weiss.c b/sound/firewire/dice/dice-weiss.c new file mode 100644 index 000000000000..ad22bfafd324 --- /dev/null +++ b/sound/firewire/dice/dice-weiss.c @@ -0,0 +1,120 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * dice-weiss.c - a part of driver for DICE based devices + * + * Copyright (c) 2023 Rolf Anderegg and Michele Perrone + */ + +#include "dice.h" + +struct dice_weiss_spec { + unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + bool has_midi; +}; + +/* Weiss DAC202: 192kHz 2-channel DAC */ +static const struct dice_weiss_spec dac202 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss MAN301: 192kHz 2-channel music archive network player */ +static const struct dice_weiss_spec man301 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss INT202: 192kHz unidirectional 2-channel digital Firewire interface */ +static const struct dice_weiss_spec int202 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{0, 0, 0}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss INT203: 192kHz bidirectional 2-channel digital Firewire interface */ +static const struct dice_weiss_spec int203 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss ADC2: 192kHz A/D converter with microphone preamps and line inputs */ +static const struct dice_weiss_spec adc2 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss DAC2/Minerva: 192kHz 2-channel DAC */ +static const struct dice_weiss_spec dac2_minerva = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss Vesta: 192kHz 2-channel Firewire to AES/EBU interface */ +static const struct dice_weiss_spec vesta = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss AFI1: 192kHz 24-channel Firewire to ADAT or AES/EBU interface */ +static const struct dice_weiss_spec afi1 = { + .tx_pcm_chs = {{24, 16, 8}, {0, 0, 0} }, + .rx_pcm_chs = {{24, 16, 8}, {0, 0, 0} }, + .has_midi = false +}; + +int snd_dice_detect_weiss_formats(struct snd_dice *dice) +{ + static const struct { + u32 model_id; + const struct dice_weiss_spec *spec; + } *entry, entries[] = { + {0x000007, &dac202}, + {0x000008, &dac202}, /* Maya edition: same audio I/O as DAC202 */ + {0x000006, &int202}, + {0x00000a, &int203}, + {0x00000b, &man301}, + {0x000001, &adc2}, + {0x000003, &dac2_minerva}, + {0x000002, &vesta}, + {0x000004, &afi1}, + }; + struct fw_csr_iterator it; + int key, val, model_id; + int i; + + model_id = 0; + fw_csr_iterator_init(&it, dice->unit->directory); + while (fw_csr_iterator_next(&it, &key, &val)) { + if (key == CSR_MODEL) { + model_id = val; + break; + } + } + + for (i = 0; i < ARRAY_SIZE(entries); ++i) { + entry = entries + i; + if (entry->model_id == model_id) + break; + } + if (i == ARRAY_SIZE(entries)) + return -ENODEV; + + memcpy(dice->tx_pcm_chs, entry->spec->tx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + memcpy(dice->rx_pcm_chs, entry->spec->rx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + + if (entry->spec->has_midi) { + dice->tx_midi_ports[0] = 1; + dice->rx_midi_ports[0] = 1; + } + + return 0; +} diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 6c93e6e4982c..4bcd026a0b3f 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -396,6 +396,78 @@ static const struct ieee1394_device_id dice_id_table[] = { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, }, + /* Weiss DAC202: 192kHz 2-channel DAC */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000007, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss DAC202: 192kHz 2-channel DAC (Maya edition) */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000008, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss MAN301: 192kHz 2-channel music archive network player */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x00000b, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss INT202: 192kHz unidirectional 2-channel digital Firewire interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000006, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss INT203: 192kHz bidirectional 2-channel digital Firewire interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x00000a, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss ADC2: 192kHz A/D converter with microphone preamps and line inputs */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000001, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss DAC2/Minerva: 192kHz 2-channel DAC */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000003, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss Vesta: 192kHz 2-channel Firewire to AES/EBU interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000002, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss AFI1: 192kHz 24-channel Firewire to ADAT or AES/EBU interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000004, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, dice_id_table); base-commit: 9b4469410cf9a0fcbccc92c480fd42f7c815a745 -- 2.41.0

3 26

Codec stops working after sleep and resume
by Jesús Jiménez Sánchez 15 Feb '24

15 Feb '24

I am using the tlv320aic31xx codec in my embedded device and I've noticed that, after forcing the unit into sleep and then, resume, I get a bunch of these errors: ``` tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: aic31xx_wait_bits: Failed! 0x25 was 0xcc239ca8 expected 0x0 (-108, 0x20, 0 us) tlv320aic31xx-codec 0-0018: ASoC: POST_PMD: HPL Driver event failed: -1 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: POST_PMD: SPL ClassD event failed: -108 tlv320aic31xx-codec 0-0018: ASoC: POST_PMD: SPR ClassD event failed: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 ``` When I look at he output of `dmesg`, I found this: ``` ------------[ cut here ]------------ WARNING: CPU: 0 PID: 1506 at drivers/i2c/i2c-core.h:54 __i2c_transfer+0xac/0xb4 i2c i2c-0: Transfer while suspended Modules linked in: rfcomm stm32_adc stm32_timer_trigger stm32_lptimer_trigger sha256_generic libsha256 sha256_arm snd_soc_simple_card snd_soc_simple_card_utils cfg80211 stm32_crc32 hci_uart btqca stm32_cryp libdes crypto_engine snd_soc_stm32_i2s snd_soc_tlv320aic31xx bluetooth stm32_adc_core snd_soc_core atmel_mxt_ts snd_pcm_dmaengine snd_pcm ecdh_generic snd_timer snd ecc sgm4151x_charger soundcore libaes [last unloaded: wlan] CPU: 0 PID: 1506 Comm: sh Tainted: G O 5.15.67 #1 Hardware name: STM32 (Device Tree Support) [<c010e8cc>] (unwind_backtrace) from [<c010c120>] (show_stack+0x10/0x14) [<c010c120>] (show_stack) from [<c011dc1c>] (__warn+0xec/0x148) [<c011dc1c>] (__warn) from [<c0bed42c>] (warn_slowpath_fmt+0x78/0xac) [<c0bed42c>] (warn_slowpath_fmt) from [<c092ecdc>] (__i2c_transfer+0xac/0xb4) [<c092ecdc>] (__i2c_transfer) from [<c092ed80>] (i2c_transfer+0x9c/0x138) [<c092ed80>] (i2c_transfer) from [<c06f5a40>] (regmap_i2c_read+0x5c/0x98) [<c06f5a40>] (regmap_i2c_read) from [<c06f0e2c>] (_regmap_raw_read+0x10c/0x2a8) [<c06f0e2c>] (_regmap_raw_read) from [<c06f1008>] (_regmap_bus_read+0x40/0x6c) [<c06f1008>] (_regmap_bus_read) from [<c06ef9c4>] (_regmap_read+0x60/0x14c) [<c06ef9c4>] (_regmap_read) from [<c06effdc>] (_regmap_update_bits+0xb4/0xf4) [<c06effdc>] (_regmap_update_bits) from [<c06f00c0>] (_regmap_select_page+0xa4/0x108) [<c06f00c0>] (_regmap_select_page) from [<c06f07a8>] (_regmap_raw_write_impl+0x684/0x968) [<c06f07a8>] (_regmap_raw_write_impl) from [<c06f0b04>] (_regmap_bus_raw_write+0x78/0xa0) [<c06f0b04>] (_regmap_bus_raw_write) from [<c06efe40>] (_regmap_write+0x4c/0x134) [<c06efe40>] (_regmap_write) from [<c06f0010>] (_regmap_update_bits+0xe8/0xf4) [<c06f0010>] (_regmap_update_bits) from [<c06f1820>] (regmap_update_bits_base+0x50/0x74) [<c06f1820>] (regmap_update_bits_base) from [<bf0d0980>] (snd_soc_component_update_bits+0x78/0x124 [snd_soc_core]) [<bf0d0980>] (snd_soc_component_update_bits [snd_soc_core]) from [<bf0daa20>] (dapm_seq_run_coalesced+0x14c/0x1a4 [snd_soc_core]) [<bf0daa20>] (dapm_seq_run_coalesced [snd_soc_core]) from [<bf0dab68>] (dapm_seq_run+0xf0/0x350 [snd_soc_core]) [<bf0dab68>] (dapm_seq_run [snd_soc_core]) from [<bf0db170>] (dapm_power_widgets+0x3a8/0x530 [snd_soc_core]) [<bf0db170>] (dapm_power_widgets [snd_soc_core]) from [<bf0cc214>] (snd_soc_dapm_sync+0x48/0x54 [snd_soc_core]) [<bf0cc214>] (snd_soc_dapm_sync [snd_soc_core]) from [<bf0c652c>] (snd_soc_suspend+0xf0/0x230 [snd_soc_core]) [<bf0c652c>] (snd_soc_suspend [snd_soc_core]) from [<c06df188>] (dpm_run_callback+0x48/0x134) [<c06df188>] (dpm_run_callback) from [<c06df58c>] (__device_suspend+0x11c/0x4c4) [<c06df58c>] (__device_suspend) from [<c06e1c10>] (dpm_suspend+0x148/0x2cc) [<c06e1c10>] (dpm_suspend) from [<c06e20e8>] (dpm_suspend_start+0x64/0x6c) [<c06e20e8>] (dpm_suspend_start) from [<c0159f00>] (suspend_devices_and_enter+0x150/0x2b4) [<c0159f00>] (suspend_devices_and_enter) from [<c0bede38>] (enter_state+0x204/0x250) [<c0bede38>] (enter_state) from [<c015a0a4>] (pm_suspend+0x40/0x98) [<c015a0a4>] (pm_suspend) from [<c0158ae4>] (state_store+0x68/0xc8) [<c0158ae4>] (state_store) from [<c0330a68>] (kernfs_fop_write_iter+0x10c/0x1cc) [<c0330a68>] (kernfs_fop_write_iter) from [<c028ea40>] (vfs_write+0x298/0x45c) [<c028ea40>] (vfs_write) from [<c028ed4c>] (ksys_write+0x60/0xe4) [<c028ed4c>] (ksys_write) from [<c0100060>] (ret_fast_syscall+0x0/0x48) Exception stack(0xcffcbfa8 to 0xcffcbff0) bfa0: 00000004 005ac7d8 00000001 005ac7d8 00000004 00000000 bfc0: 00000004 005ac7d8 00000001 00000004 b6f01680 bee896d8 005a8448 005a8590 bfe0: 005a6dfc bee89690 004fcda4 b6dcb4f8 ---[ end trace 75457fd792755cae ]--- ``` I've been trying for a few days, but I can't find the source of the problem or a way to solve it. Is there anything I might be missing? I assume the problem is that it's trying to write to the registers after the suspend has happened, but I haven't been able to prove that either, and I wouldn't know how to fix that. Could that be what's happening here? Thanks.

1 1

[PATCH v7 0/6] Add cs42l43 PC focused SoundWire CODEC
by Charles Keepax 31 Jan '24

31 Jan '24

This patch chain adds support for the Cirrus Logic cs42l43 PC focused SoundWire CODEC. The chain is currently based of Lee's for-mfd-next branch. This series is mostly just a resend keeping pace with the kernel under it, except for a minor fixup in the ASoC stuff. Thanks, Charles Charles Keepax (4): dt-bindings: mfd: cirrus,cs42l43: Add initial DT binding mfd: cs42l43: Add support for cs42l43 core driver pinctrl: cs42l43: Add support for the cs42l43 ASoC: cs42l43: Add support for the cs42l43 Lucas Tanure (2): soundwire: bus: Allow SoundWire peripherals to register IRQ handlers spi: cs42l43: Add SPI controller support .../bindings/sound/cirrus,cs42l43.yaml | 313 +++ MAINTAINERS | 4 + drivers/mfd/Kconfig | 23 + drivers/mfd/Makefile | 3 + drivers/mfd/cs42l43-i2c.c | 98 + drivers/mfd/cs42l43-sdw.c | 239 ++ drivers/mfd/cs42l43.c | 1188 +++++++++ drivers/mfd/cs42l43.h | 28 + drivers/pinctrl/cirrus/Kconfig | 11 + drivers/pinctrl/cirrus/Makefile | 2 + drivers/pinctrl/cirrus/pinctrl-cs42l43.c | 609 +++++ drivers/soundwire/bus.c | 32 + drivers/soundwire/bus_type.c | 12 + drivers/spi/Kconfig | 7 + drivers/spi/Makefile | 1 + drivers/spi/spi-cs42l43.c | 284 ++ include/linux/mfd/cs42l43-regs.h | 1184 +++++++++ include/linux/mfd/cs42l43.h | 102 + include/linux/soundwire/sdw.h | 9 + include/sound/cs42l43.h | 17 + sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42l43-jack.c | 946 +++++++ sound/soc/codecs/cs42l43-sdw.c | 74 + sound/soc/codecs/cs42l43.c | 2278 +++++++++++++++++ sound/soc/codecs/cs42l43.h | 131 + 26 files changed, 7615 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml create mode 100644 drivers/mfd/cs42l43-i2c.c create mode 100644 drivers/mfd/cs42l43-sdw.c create mode 100644 drivers/mfd/cs42l43.c create mode 100644 drivers/mfd/cs42l43.h create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs42l43.c create mode 100644 drivers/spi/spi-cs42l43.c create mode 100644 include/linux/mfd/cs42l43-regs.h create mode 100644 include/linux/mfd/cs42l43.h create mode 100644 include/sound/cs42l43.h create mode 100644 sound/soc/codecs/cs42l43-jack.c create mode 100644 sound/soc/codecs/cs42l43-sdw.c create mode 100644 sound/soc/codecs/cs42l43.c create mode 100644 sound/soc/codecs/cs42l43.h -- 2.30.2

6 32

[v3 0/3] ASoC: mediatek:mt8186: fix both the speaker and hdmi
by Jiaxin Yu 31 Jan '24

31 Jan '24

The speaker and hdmi of mt8186 platform are shared the same port of I2S, when connect the external display, use build-in speakers to play audio, they both playback at the same time. So we want to manage the playback device through DAPM events. Jiaxin Yu (3): ASoC: hdmi-codec: Add event handler for hdmi TX ASoC: mediatek: mt8186: correct the HDMI widgets drm/bridge: it6505: Add audio support drivers/gpu/drm/bridge/ite-it6505.c | 81 +++++++++++++++++-- include/sound/hdmi-codec.h | 6 ++ sound/soc/codecs/hdmi-codec.c | 32 +++++++- .../mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 5 files changed, 113 insertions(+), 10 deletions(-) -- 2.25.1

5 13

[PATCH] Add a quirk for Yamaha YIT-W12TX transmitter
by Julian Sikorski 23 Jan '24

23 Jan '24

The device fails to initialize otherwise, giving the following error: [ 3676.671641] usb 2-1.1: 1:1: cannot get freq at ep 0x1 Signed-off-by: Julian Sikorski <belegdol+github(a)gmail.com> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ab2b938502eb..bf0a7cca90d0 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2031,6 +2031,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x0499, 0x1509, /* Steinberg UR22 */ QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x0499, 0x3108, /* Yamaha YIT-W12TX */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x04d8, 0xfeea, /* Benchmark DAC1 Pre */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x04e8, 0xa051, /* Samsung USBC Headset (AKG) */ -- 2.43.0

2 2

[PATCH v11 00/15] Add audio support in v4l2 framework
by Shengjiu Wang 18 Jan '24

18 Jan '24

Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. It is a specific use case, there is no reference in current kernel. v4l2 memory to memory is the closed implementation, v4l2 current support video, image, radio, tuner, touch devices, so it is not complicated to add support for this specific audio case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the first 3 patches is for refining the code to make it can be shared by the "memory->asrc->memory" driver. The main change is in the v4l2 side, A /dev/vl4-audioX will be created, user applications only use the ioctl of v4l2 framework. Other change is to add memory to memory support for two kinds of i.MX ASRC module. changes in v11 - add add-fixed-point-test-controls in vivid. - add v4l2_ctrl_fp_compose() helper function for min and max changes in v10 - remove FIXED_POINT type - change code base on media: v4l2-ctrls: add support for fraction_bits - fix issue reported by kernel test robot - remove module_alias changes in v9: - add MEDIA_ENT_F_PROC_AUDIO_RESAMPLER. - add MEDIA_INTF_T_V4L_AUDIO - add media controller support - refine the vim2m-audio to support 8k<->16k conversion. changes in v8: - refine V4L2_CAP_AUDIO_M2M to be 0x00000008 - update doc for FIXED_POINT - address comments for imx-asrc changes in v7: - add acked-by from Mark - separate commit for fixed point, m2m audio class, audio rate controls - use INTEGER_MENU for rate, FIXED_POINT for rate offset - remove used fmts - address other comments for Hans changes in v6: - use m2m_prepare/m2m_unprepare/m2m_start/m2m_stop to replace m2m_start_part_one/m2m_stop_part_one, m2m_start_part_two/m2m_stop_part_two. - change V4L2_CTRL_TYPE_ASRC_RATE to V4L2_CTRL_TYPE_FIXED_POINT - fix warning by kernel test rebot - remove some unused format V4L2_AUDIO_FMT_XX - Get SNDRV_PCM_FORMAT from V4L2_AUDIO_FMT in driver. - rename audm2m to viaudm2m. changes in v5: - remove V4L2_AUDIO_FMT_LPCM - define audio pixel format like V4L2_AUDIO_FMT_S8... - remove rate and format in struct v4l2_audio_format. - Add V4L2_CID_ASRC_SOURCE_RATE and V4L2_CID_ASRC_DEST_RATE controls - updata document accordingly. changes in v4: - update document style - separate V4L2_AUDIO_FMT_LPCM and V4L2_CAP_AUDIO_M2M in separate commit changes in v3: - Modify documents for adding audio m2m support - Add audio virtual m2m driver - Defined V4L2_AUDIO_FMT_LPCM format type for audio. - Defined V4L2_CAP_AUDIO_M2M capability type for audio m2m case. - with modification in v4l-utils, pass v4l2-compliance test. changes in v2: - decouple the implementation in v4l2 and ALSA - implement the memory to memory driver as a platfrom driver and move it to driver/media - move fsl_asrc_common.h to include/sound folder Shengjiu Wang (15): ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc: move fsl_asrc_common.h to include/sound ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device media: uapi: Add V4L2_CAP_AUDIO_M2M capability flag media: v4l2: Add audio capture and output support media: uapi: Define audio sample format fourcc type media: uapi: Add V4L2_CTRL_CLASS_M2M_AUDIO media: uapi: Add audio rate controls support media: uapi: Declare interface types for Audio media: uapi: Add an entity type for audio resampler media: vivid: add fixed point test controls media: imx-asrc: Add memory to memory driver media: vim2m-audio: add virtual driver for audio memory to memory .../media/mediactl/media-types.rst | 11 + .../userspace-api/media/v4l/buffer.rst | 6 + .../userspace-api/media/v4l/common.rst | 1 + .../media/v4l/dev-audio-mem2mem.rst | 71 + .../userspace-api/media/v4l/devices.rst | 1 + .../media/v4l/ext-ctrls-audio-m2m.rst | 41 + .../userspace-api/media/v4l/pixfmt-audio.rst | 87 ++ .../userspace-api/media/v4l/pixfmt.rst | 1 + .../media/v4l/vidioc-enum-fmt.rst | 2 + .../media/v4l/vidioc-g-ext-ctrls.rst | 4 + .../userspace-api/media/v4l/vidioc-g-fmt.rst | 4 + .../media/v4l/vidioc-querycap.rst | 3 + .../media/videodev2.h.rst.exceptions | 3 + .../media/common/videobuf2/videobuf2-v4l2.c | 4 + drivers/media/platform/nxp/Kconfig | 13 + drivers/media/platform/nxp/Makefile | 1 + drivers/media/platform/nxp/imx-asrc.c | 1264 +++++++++++++++++ drivers/media/test-drivers/Kconfig | 11 + drivers/media/test-drivers/Makefile | 1 + drivers/media/test-drivers/vim2m-audio.c | 799 +++++++++++ drivers/media/test-drivers/vivid/vivid-core.h | 2 + .../media/test-drivers/vivid/vivid-ctrls.c | 26 + drivers/media/v4l2-core/v4l2-compat-ioctl32.c | 9 + drivers/media/v4l2-core/v4l2-ctrls-defs.c | 10 + drivers/media/v4l2-core/v4l2-dev.c | 21 + drivers/media/v4l2-core/v4l2-ioctl.c | 66 + drivers/media/v4l2-core/v4l2-mem2mem.c | 13 +- include/media/v4l2-ctrls.h | 3 + include/media/v4l2-dev.h | 2 + include/media/v4l2-ioctl.h | 34 + .../fsl => include/sound}/fsl_asrc_common.h | 60 + include/uapi/linux/media.h | 2 + include/uapi/linux/v4l2-controls.h | 9 + include/uapi/linux/videodev2.h | 41 + sound/soc/fsl/fsl_asrc.c | 144 ++ sound/soc/fsl/fsl_asrc.h | 4 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 233 +++ sound/soc/fsl/fsl_easrc.h | 6 +- 39 files changed, 3008 insertions(+), 7 deletions(-) create mode 100644 Documentation/userspace-api/media/v4l/dev-audio-mem2mem.rst create mode 100644 Documentation/userspace-api/media/v4l/ext-ctrls-audio-m2m.rst create mode 100644 Documentation/userspace-api/media/v4l/pixfmt-audio.rst create mode 100644 drivers/media/platform/nxp/imx-asrc.c create mode 100644 drivers/media/test-drivers/vim2m-audio.c rename {sound/soc/fsl => include/sound}/fsl_asrc_common.h (60%) -- 2.34.1

3 25

[RESEND PATCH 0/1] ALSA: virtio: add support for audio controls
by Aiswarya Cyriac 15 Jan '24

15 Jan '24

From: Anton Yakovlev <anton.yakovlev(a)opensynergy.com> Implementation of support for audio controls in accordance with the extension of the virtio sound device specification[1] planned for virtio-v1.3-cs01. Design　of virtual audio controls is based on and derived from ALSA audio controls. It allows the driver to perform all standard operations, such as reading and writing audio control value, as well as working with metadata (represented in the TLV form). The driver part was tested on top of the Linux 5.10 kernel. As a device part was used OpenSynergy proprietary implementation. [1] https://lists.oasis-open.org/archives/virtio-comment/202104/msg00013.html Anton Yakovlev (1): ALSA: virtio: add support for audio controls include/uapi/linux/virtio_snd.h | 154 +++++++++++ sound/virtio/Makefile | 1 + sound/virtio/virtio_card.c | 21 ++ sound/virtio/virtio_card.h | 22 ++ sound/virtio/virtio_kctl.c | 464 ++++++++++++++++++++++++++++++++ 5 files changed, 662 insertions(+) create mode 100644 sound/virtio/virtio_kctl.c -- 2.25.1

2 3

[PATCH 0/7] ASoC: rt5645: Add Acer Switch V 10 quirk + report quirks in components string
by Hans de Goede 22 Dec '23

22 Dec '23

Hi all, This patch-series adds a quirk for the Acer Switch V 10, which uses a DMIC on the DMIC2 input, rather then the default analog mic on IN2. Most of the patches are actually for adding a component string with mic and speaker routing information so that the UCM profile can use this instead of duplicating all the DMI quirks in the UCM profile. I've also submitted a alsa-ucm-config patch to add support for the components string. Regards, Hans Hans de Goede (7): ASoC: rt5645: Drop double EF20 entry from dmi_platform_data[] ASoC: rt5645: Add platform-data for Acer Switch V 10 ASoC: rt5645: Refactor rt5645_parse_dt() ASoC: rt5645: Add rt5645_get_pdata() helper ASoC: rt5645: Add a rt5645_components() helper ASoC: rt5645: Add mono speaker information to the components string ASoC: Intel: cht_bsw_rt5645: Set card.components string sound/soc/codecs/rt5645.c | 113 +++++++++++++++--------- sound/soc/codecs/rt5645.h | 3 + sound/soc/intel/boards/cht_bsw_rt5645.c | 8 ++ 3 files changed, 83 insertions(+), 41 deletions(-) -- 2.41.0

2 8

[PATCH V2 1/2] ASoC: amd: acp: add Kconfig options for acp7.0 based platform driver
by Syed Saba Kareem 18 Dec '23

18 Dec '23

ACP7.0 based platform legacy drivers can be built by selecting necessary kernel config option. This patch enables build support of the same. Signed-off-by: Syed Saba Kareem <Syed.SabaKareem(a)amd.com> --- changes since v1: - added missing commit message. sound/soc/amd/acp/Kconfig | 12 ++++++++++++ sound/soc/amd/acp/Makefile | 2 ++ 2 files changed, 14 insertions(+) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 5fb322212938..c8ac0027f741 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -73,6 +73,18 @@ config SND_AMD_ASOC_ACP63 Say Y if you want to enable AUDIO on ACP6.3 If unsure select "N". +config SND_AMD_ASOC_ACP70 + tristate "AMD ACP ASOC Acp7.0 Support" + depends on X86 && PCI + depends on ACPI + select SND_SOC_AMD_ACP_PCM + select SND_SOC_AMD_ACP_I2S + select SND_SOC_AMD_ACP_PDM + help + This option enables Acp7.0 PDM support on AMD platform. + Say Y if you want to enable AUDIO on ACP7.0 + If unsure select "N". + config SND_SOC_AMD_MACH_COMMON tristate depends on X86 && PCI && I2C diff --git a/sound/soc/amd/acp/Makefile b/sound/soc/amd/acp/Makefile index dd85700f1c5f..ff5f7893b81e 100644 --- a/sound/soc/amd/acp/Makefile +++ b/sound/soc/amd/acp/Makefile @@ -15,6 +15,7 @@ snd-acp-pci-objs := acp-pci.o snd-acp-renoir-objs := acp-renoir.o snd-acp-rembrandt-objs := acp-rembrandt.o snd-acp63-objs := acp63.o +snd-acp70-objs := acp70.o #machine specific driver snd-acp-mach-objs := acp-mach-common.o @@ -30,6 +31,7 @@ obj-$(CONFIG_SND_SOC_AMD_ACP_PCI) += snd-acp-pci.o obj-$(CONFIG_SND_AMD_ASOC_RENOIR) += snd-acp-renoir.o obj-$(CONFIG_SND_AMD_ASOC_REMBRANDT) += snd-acp-rembrandt.o obj-$(CONFIG_SND_AMD_ASOC_ACP63) += snd-acp63.o +obj-$(CONFIG_SND_AMD_ASOC_ACP70) += snd-acp70.o obj-$(CONFIG_SND_SOC_AMD_MACH_COMMON) += snd-acp-mach.o obj-$(CONFIG_SND_SOC_AMD_LEGACY_MACH) += snd-acp-legacy-mach.o -- 2.25.1

7 8

[PATCH] ASoC: hdmi-codec: register hpd callback on component probe
by Jerome Brunet 18 Dec '23

18 Dec '23

The HDMI hotplug callback to the hdmi-codec is currently registered when jack is set. The hotplug not only serves to report the ASoC jack state but also to get the ELD. It should be registered when the component probes instead, so it does not depend on the card driver registering a jack for the HDMI to properly report the ELD. Fixes: 25ce4f2b3593 ("ASoC: hdmi-codec: Get ELD in before reporting plugged event") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/codecs/hdmi-codec.c | 27 +++++++++++++++++++-------- 1 file changed, 19 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 09eef6042aad..20da1eaa4f1c 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -877,18 +877,13 @@ static int hdmi_codec_set_jack(struct snd_soc_component *component, void *data) { struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); - int ret = -ENOTSUPP; if (hcp->hcd.ops->hook_plugged_cb) { hcp->jack = jack; - ret = hcp->hcd.ops->hook_plugged_cb(component->dev->parent, - hcp->hcd.data, - plugged_cb, - component->dev); - if (ret) - hcp->jack = NULL; + return 0; } - return ret; + + return -ENOTSUPP; } static int hdmi_dai_spdif_probe(struct snd_soc_dai *dai) @@ -982,6 +977,21 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, return ret; } +static int hdmi_probe(struct snd_soc_component *component) +{ + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + int ret = 0; + + if (hcp->hcd.ops->hook_plugged_cb) { + ret = hcp->hcd.ops->hook_plugged_cb(component->dev->parent, + hcp->hcd.data, + plugged_cb, + component->dev); + } + + return ret; +} + static void hdmi_remove(struct snd_soc_component *component) { struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); @@ -992,6 +1002,7 @@ static void hdmi_remove(struct snd_soc_component *component) } static const struct snd_soc_component_driver hdmi_driver = { + .probe = hdmi_probe, .remove = hdmi_remove, .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), -- 2.42.0

4 6

[PATCH v4 0/2] ASoC: audio-graph-card2: Introduce playback-only/capture-only DAI link flags
by Daniel Baluta 14 Dec '23

14 Dec '23

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries allows users to specify a link only direction with audio-graph-card2. Changes since v1 - https://lore.kernel.org/lkml/20230801082433.548206-1-daniel.baluta@oss.nxp.… - used audio-graph-card2 instead of simple-card as we are deprecting simple-card. Changes since v2: - Fix device tree binding error reported by Rob Changes since v3: - Add signed-off-by line to patch 1/2 - Enhance commit messsage for patch 1/2 to explain why the patch is needed. Daniel Baluta (2): ASoC: audio-graph-card2: Introduce playback-only/capture-only DAI link flags ASoC: dt-bindings: audio-graph-port: Document new DAI link flags playback-only/capture-only .../bindings/sound/audio-graph-port.yaml | 6 ++++++ include/sound/simple_card_utils.h | 3 +++ sound/soc/generic/audio-graph-card2.c | 6 ++++++ sound/soc/generic/simple-card-utils.c | 19 +++++++++++++++++++ 4 files changed, 34 insertions(+) -- 2.25.1

3 4

[PATCH 00/31] Fix and improve Rockchip RK3128 support
by Alex Bee 13 Dec '23

13 Dec '23

Hi list, this series fixes some issues I found when testing my "new" RK3128 board with the mainline kernel and adds some core functionality like SMP bringup, usb and networking. The propably most distinctive change is the split up of the DTs for the different SoCs of this platform: RK3126 and RK3128. Even if I'm not adding a RK3126 board in this series: I think this change should be done as early as possible in order to avoid issues in future. Actually it should have been done like that in the first place. All patches have been tested on both SoCs: Patch 01-04 are dt-bindings additions for this SoC platform and the new board I'm addding here Patch 05-07 are some clock driver fixes (there is more to do, but I'm limiting it to the things which can work now) Patch 08 and 09 adding support for the usb 2 phy found in RK312x platform to the Rockchip Innosilicon usb phy driver. Patch 10-14 are DT patches which I did in the "old" rk3128.dtsi in order to be backportable. Patch 15 does the aforementioned split-up of the current rk3128.dtsi in rk312x.dtsi, rk3126.dtsi and rk3128.dtsi Patch 16-20 adds SMP bringup and cpu frequency scaling Patch 21-23 adds power controller, GPU and I2S nodes to the RK312x DT Patch 24-26 adds the second I2S, S/PDIF and gmac nodes to the RK3128 DT Patch 27-30 adds fixes for the usb controllers and adds a missing pincontrol for the sd card detection Patch 31 finnaly adds a DT for Genatech XPI-3128 board There are quite some HW blocks which can easily be added for this platform, but I guess this series is pretty huge already, so that it is limited to some core devices for now. I hope the single patches are small enough, that reviewing won't be a burden ;) The plan is to sync the DT changes to u-boot after they are merged where I'm planning to add "full" TPL/SPL for this platform (dram controller/phy driver is currently missing) I'm planing the add a RK3126 (tablet) board here as well, but I will have to add support for RK816 pmic first. Alex Bee (29): dt-bindings: mfd: syscon: Add rockchip,rk3128-qos compatible dt-bindings: gpu: mali-utgard: Add Rockchip RK3128 compatible dt-bindings: ASoC: rockchip: Add compatible for RK3128 spdif dt-bindings: arm: rockchip: Add Geniatech XPI-3128 clk: rockchip: rk3128: Fix SCLK_SDMMC's clock name phy: rockchip-inno-usb2: Split ID interrupt phy registers phy: phy-rockchip-inno-usb2: Add RK3128 support ARM: dts: rockchip: Fix i2c0 register address for RK3128 ARM: dts: rockchip: Add missing arm timer interrupt for RK3128 ARM: dts: rockchip: Add missing quirk for RK3128's dma engine ARM: dts: rockchip: Fix timer clocks for RK3128 ARM: dts: rockchip: Disable non-required timers for RK3128 ARM: dts: rockchip: Split RK3128 devictree for RK312x SoC family ARM: dts: rockchip: Add SRAM node for RK312x ARM: dts: rockchip: Add CPU resets for RK312x ARM: dts: rockchip: Enable SMP bringup for RK312x ARM: dts: rockchip: Switch to operating-points-v2 for RK312x's CPU ARM: dts: rockchip: Add extra CPU voltages for RK3126 ARM: dts: rockchip: add power controller for RK312x ARM: dts: rockchip: Add GPU node for RK312x ARM: dts: rockchip: Add 2-channel I2S for RK312x ARM: dts: rockchip: Add 8-channel I2S for RK3128 ARM: dts: rockchip: Add spdif for RK3128 ARM: dts: rockchip: Add gmac for RK3128 ARM: dts: rockchip: Add dwc2 otg fifo siztes for RK312x ARM: dts: rockchip: Add USB host clocks for RK312x ARM: dts: rockchip: Make usbphy the parent of SCLK_USB480M for RK312x ARM: dts: rockchip: Add sdmmc_det pinctrl for RK312x ARM: dts: Add Geniatech XPI-3128 RK3128 board Finley Xiao (2): clk: rockchip: rk3128: Fix aclk_peri_src parent clk: rockchip: rk3128: Fix hclk_otg gate .../devicetree/bindings/arm/rockchip.yaml | 5 + .../bindings/gpu/arm,mali-utgard.yaml | 1 + .../devicetree/bindings/mfd/syscon.yaml | 1 + .../bindings/sound/rockchip-spdif.yaml | 1 + arch/arm/boot/dts/rockchip/Makefile | 1 + arch/arm/boot/dts/rockchip/rk3126.dtsi | 36 + .../arm/boot/dts/rockchip/rk3128-xpi-3128.dts | 431 +++++++ arch/arm/boot/dts/rockchip/rk3128.dtsi | 920 +------------- arch/arm/boot/dts/rockchip/rk312x.dtsi | 1101 +++++++++++++++++ drivers/clk/rockchip/clk-rk3128.c | 24 +- drivers/phy/rockchip/phy-rockchip-inno-usb2.c | 147 ++- 11 files changed, 1754 insertions(+), 914 deletions(-) create mode 100644 arch/arm/boot/dts/rockchip/rk3126.dtsi create mode 100644 arch/arm/boot/dts/rockchip/rk3128-xpi-3128.dts create mode 100644 arch/arm/boot/dts/rockchip/rk312x.dtsi base-commit: 2dde18cd1d8fac735875f2e4987f11817cc0bc2c -- 2.42.0

6 46

[PATCH v2 1/3] ASoC: qcom: audioreach: Commonize setting channel mappings
by Krzysztof Kozlowski 11 Dec '23

11 Dec '23

Move code assigning channel mapping values to a common helper function. This simplifies three out of four cases, with the last case using incompatible type (uint16_t array instead of uint8_t array). Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v2: 1. New patch --- sound/soc/qcom/qdsp6/audioreach.c | 35 ++++++++++++++----------------- 1 file changed, 16 insertions(+), 19 deletions(-) diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c index 5974c7929dd3..3db5ff367a29 100644 --- a/sound/soc/qcom/qdsp6/audioreach.c +++ b/sound/soc/qcom/qdsp6/audioreach.c @@ -267,6 +267,16 @@ void *audioreach_alloc_apm_cmd_pkt(int pkt_size, uint32_t opcode, uint32_t token } EXPORT_SYMBOL_GPL(audioreach_alloc_apm_cmd_pkt); +static void audioreach_set_channel_mapping(u8 *ch_map, int num_channels) +{ + if (num_channels == 1) { + ch_map[0] = PCM_CHANNEL_L; + } else if (num_channels == 2) { + ch_map[0] = PCM_CHANNEL_L; + ch_map[1] = PCM_CHANNEL_R; + } +} + static void apm_populate_container_config(struct apm_container_obj *cfg, struct audioreach_container *cont) { @@ -864,12 +874,8 @@ static int audioreach_set_compr_media_format(struct media_format *media_fmt_hdr, mp3_cfg->endianness = PCM_LITTLE_ENDIAN; mp3_cfg->num_channels = mcfg->num_channels; - if (mcfg->num_channels == 1) { - mp3_cfg->channel_mapping[0] = PCM_CHANNEL_L; - } else if (mcfg->num_channels == 2) { - mp3_cfg->channel_mapping[0] = PCM_CHANNEL_L; - mp3_cfg->channel_mapping[1] = PCM_CHANNEL_R; - } + audioreach_set_channel_mapping(mp3_cfg->channel_mapping, + mcfg->num_channels); break; case SND_AUDIOCODEC_AAC: media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED; @@ -1089,13 +1095,8 @@ static int audioreach_pcm_set_media_format(struct q6apm_graph *graph, media_cfg->q_factor = mcfg->bit_width - 1; media_cfg->bits_per_sample = mcfg->bit_width; - if (num_channels == 1) { - media_cfg->channel_mapping[0] = PCM_CHANNEL_L; - } else if (num_channels == 2) { - media_cfg->channel_mapping[0] = PCM_CHANNEL_L; - media_cfg->channel_mapping[1] = PCM_CHANNEL_R; - - } + audioreach_set_channel_mapping(media_cfg->channel_mapping, + num_channels); rc = q6apm_send_cmd_sync(graph->apm, pkt, 0); @@ -1153,12 +1154,8 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph, cfg->endianness = PCM_LITTLE_ENDIAN; cfg->num_channels = mcfg->num_channels; - if (mcfg->num_channels == 1) - cfg->channel_mapping[0] = PCM_CHANNEL_L; - else if (num_channels == 2) { - cfg->channel_mapping[0] = PCM_CHANNEL_L; - cfg->channel_mapping[1] = PCM_CHANNEL_R; - } + audioreach_set_channel_mapping(cfg->channel_mapping, + num_channels); } else { rc = audioreach_set_compr_media_format(header, p, mcfg); if (rc) { -- 2.34.1

2 3

[PATCH 1/4] ASoC: dt-bindings: qcom,lpass-rx-macro: Add SM8650 LPASS RX
by Krzysztof Kozlowski 11 Dec '23

11 Dec '23

Add bindings for Qualcomm SM8650 Low Power Audio SubSystem (LPASS) RX macro codec, which looks like compatible with earlier SM8550. Cc: Neil Armstrong <neil.armstrong(a)linaro.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../bindings/sound/qcom,lpass-rx-macro.yaml | 21 ++++++++++++------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml index ec4b0ac8ad68..cbc36646100f 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml @@ -11,12 +11,16 @@ maintainers: properties: compatible: - enum: - - qcom,sc7280-lpass-rx-macro - - qcom,sm8250-lpass-rx-macro - - qcom,sm8450-lpass-rx-macro - - qcom,sm8550-lpass-rx-macro - - qcom,sc8280xp-lpass-rx-macro + oneOf: + - enum: + - qcom,sc7280-lpass-rx-macro + - qcom,sm8250-lpass-rx-macro + - qcom,sm8450-lpass-rx-macro + - qcom,sm8550-lpass-rx-macro + - qcom,sc8280xp-lpass-rx-macro + - items: + - const: qcom,sm8650-lpass-rx-macro + - const: qcom,sm8550-lpass-rx-macro reg: maxItems: 1 @@ -96,8 +100,9 @@ allOf: - if: properties: compatible: - enum: - - qcom,sm8550-lpass-rx-macro + contains: + enum: + - qcom,sm8550-lpass-rx-macro then: properties: clocks: -- 2.34.1

3 7

Bass speaker fixup for Lenovo Yoga Pro 7 14APH8
by Yevhen Hrubiian 07 Dec '23

07 Dec '23

Hello ! I've seen a few patches fixing bass speaker problem for new Lenovo (written by Philipp Jungkamp and later by Pascal Gross), for example this line in linux/sound/pci/hda/patch_realtek.c: SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN) Could you please add this line for a similar model with the same Problem? SND_PCI_QUIRK(0x17aa, 0x3882, "Lenovo Yoga Pro 7 14APH8", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), I do not have experience in writing kernel patches so I hope I will not irritate you with this letter. Thanks! Yevhen Hrubiian

2 1

BUG hda/cs8904: Headphones not detected and no speaker sound after system suspend
by Andrés Pedraza 05 Dec '23

05 Dec '23

Hello my dudes. I've been having this bug for a few months now. Sound on my laptop works just fine until the system suspends or I close the lid. If my headphones are connected when I suspend the system, then there's no sound on headphones or speakers on wake and PulseAudio claims there's never any headphones connected. If my headphones aren't connected on suspend, then speakers work just fine, but headphones are still never detected and make no sound. Additionally, I recently found out that after waking from suspend, if I run `sudo dmesg | grep snd` I find two timeout warnings that weren't there before, and more showing up as I keep running the command. They go: > snd_hda_codec_cs8409 hdaudioC1D0: Timeout waiting for PDN_DONE for > CS42L42 I'm on a Dell Vostro 3405 laptop, which I know has a Cirrus sound card. I recently compiled and tested this bug on the Linux 6.6.2 stable kernel. No dice. I attach the outputs of `pactl list sinks`, `sudo dmesg | grep snd` and `sudo lspci -v | grep snd -B 10`. Please do tell if the issue might be caused by some other module or BIOS configuration, or if you need more data.

3 2

[PATCH 00/17] dt-bindings: samsung: add specific compatibles for existing SoC
by Krzysztof Kozlowski 05 Dec '23

05 Dec '23

Hi, Merging ======= I propose to take entire patchset through my tree (Samsung SoC), because: 1. Next cycle two new SoCs will be coming (Google GS101 and ExynosAutov920), so they will touch the same lines in some of the DT bindings (not all, though). It is reasonable for me to take the bindings for the new SoCs, to have clean `make dtbs_check` on the new DTS. 2. Having it together helps me to have clean `make dtbs_check` within my tree on the existing DTS. 3. No drivers are affected by this change. 4. I plan to do the same for Tesla FSD and Exynos ARM32 SoCs, thus expect follow up patchsets. If folks agree, please kindly Ack the patches. Description =========== Samsung Exynos SoCs reuse several devices from older designs, thus historically we kept the old (block's) compatible only. This works fine and there is no bug here, however guidelines expressed in Documentation/devicetree/bindings/writing-bindings.rst state that: 1. Compatibles should be specific. 2. We should add new compatibles in case of bugs or features. Add compatibles specific to each SoC in front of all old-SoC-like compatibles. This will also help reviews of new code using existing DTS as template. No functional impact on Linux drivers behavior. Future ====== If reasonable, I will do similar work for Tesla FSD and ARMv7/ARM32 Exynos bindings and DTS. Best regards, Krzysztof Krzysztof Kozlowski (17): dt-bindings: hwinfo: samsung,exynos-chipid: add specific compatibles for existing SoC dt-bindings: i2c: exynos5: add specific compatibles for existing SoC dt-bindings: i2c: samsung,s3c2410-i2c: add specific compatibles for existing SoC dt-bindings: mmc: samsung,exynos-dw-mshc: add specific compatibles for existing SoC dt-bindings: pinctrl: samsung: add specific compatibles for existing SoC dt-bindings: rtc: s3c-rtc: add specific compatibles for existing SoC dt-bindings: serial: samsung: add specific compatibles for existing SoC dt-bindings: samsung: exynos-pmu: add specific compatibles for existing SoC dt-bindings: gpu: arm,mali-midgard: add specific compatibles for existing Exynos SoC dt-bindings: iio: samsung,exynos-adc: add specific compatibles for existing SoC ASoC: dt-bindings: samsung-i2s: add specific compatibles for existing SoC dt-bindings: pwm: samsung: add specific compatibles for existing SoC arm64: dts: exynos5433: add specific compatibles to several blocks arm64: dts: exynos7: add specific compatibles to several blocks arm64: dts: exynos7885: add specific compatibles to several blocks arm64: dts: exynos850: add specific compatibles to several blocks arm64: dts: exynosautov9: add specific compatibles to several blocks .../bindings/gpu/arm,mali-midgard.yaml | 5 ++ .../hwinfo/samsung,exynos-chipid.yaml | 17 +++++- .../devicetree/bindings/i2c/i2c-exynos5.yaml | 10 +++- .../bindings/i2c/samsung,s3c2410-i2c.yaml | 22 ++++--- .../bindings/iio/adc/samsung,exynos-adc.yaml | 29 +++++---- .../mfd/samsung,exynos5433-lpass.yaml | 2 +- .../bindings/mmc/samsung,exynos-dw-mshc.yaml | 25 +++++--- .../samsung,pinctrl-wakeup-interrupt.yaml | 24 +++++--- .../bindings/pinctrl/samsung,pinctrl.yaml | 3 +- .../devicetree/bindings/pwm/pwm-samsung.yaml | 2 + .../devicetree/bindings/rtc/s3c-rtc.yaml | 5 ++ .../bindings/serial/samsung_uart.yaml | 14 ++++- .../bindings/soc/samsung/exynos-pmu.yaml | 6 ++ .../bindings/soc/samsung/exynos-usi.yaml | 2 +- .../bindings/sound/samsung-i2s.yaml | 19 +++--- arch/arm64/boot/dts/exynos/exynos5433.dtsi | 60 ++++++++++++------- arch/arm64/boot/dts/exynos/exynos7.dtsi | 18 +++--- arch/arm64/boot/dts/exynos/exynos7885.dtsi | 45 +++++++++----- arch/arm64/boot/dts/exynos/exynos850.dtsi | 34 ++++++----- arch/arm64/boot/dts/exynos/exynosautov9.dtsi | 6 +- 20 files changed, 233 insertions(+), 115 deletions(-) -- 2.34.1

13 65

Regression apparently caused by commit 088a40980efbc2c449b72f0f2c7ebd82f71d08e2 "ASoC: amd: acp: add pm ops support for acp pci driver"
by Marian Postevca 05 Dec '23

05 Dec '23

I'm trying to develop a sound machine driver based on the acp legacy driver. The first version of the driver was sent for review on the alsa mailing list this spring: https://lore.kernel.org/all/20230320203519.20137-1-posteuca@mutex.one I'm trying to fix some of the issues that were brought up during the review back then, but when I ported the patches to the latest commit on the for-next branch, I noticed a regression where I couldn't hear any sound at all. So I started a bisect session and found that the first bad commit is: ASoC: amd: acp: add pm ops support for acp pci driver commit 088a40980efbc2c449b72f0f2c7ebd82f71d08e2 https://lore.kernel.org/lkml/20230622152406.3709231-11-Syed.SabaKareem@amd.… If I revert this commit sound works as expected. So I started tinkering a little bit with it and I believe that what happens is that the acp pci driver enters the autosuspend state and never leaves this state at all. I noticed this because if I increase the autosuspend delay to a much larger value, then the sound works until that delay passes. I added traces and I can see that when the delay expires the suspend callback snd_acp_suspend() gets called, but the resume callback snd_acp_resume() never gets called. I'm no expert in runtime power management (though I did read a bit on it), so I don't understand all the things that happen underneath, but one thing that is not clear to me is who's supposed to mark activity on this device and keep it from entering autosuspend if the user wants to play some sound? Shouldn't there be some counterpart that calls pm_runtime_mark_last_busy() ? I looked through the code and can't find who's calling pm_runtime_mark_last_busy(). Some help here would be welcome. Is there something missing in my machine driver code, or is the runtime pm handling in acp pci driver wrong?

5 12

[PATCH] soundwire: amd: drop bus freq calculation and set 'max_clk_freq'
by Vinod Koul 02 Dec '23

02 Dec '23

max_dr_freq and curr_dr_freq is calculated and set in sdw_bus_master_add(). Setting in the driver is reduanant, so drop that. Set max_clk_freq instead. Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- drivers/soundwire/amd_manager.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index a3b1f4e6f0f9..f54bb4dd2d10 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -950,13 +950,13 @@ static int amd_sdw_manager_probe(struct platform_device *pdev) amd_manager->reg_mask = &sdw_manager_reg_mask_array[amd_manager->instance]; params = &amd_manager->bus.params; - params->max_dr_freq = AMD_SDW_DEFAULT_CLK_FREQ * 2; - params->curr_dr_freq = AMD_SDW_DEFAULT_CLK_FREQ * 2; + params->col = AMD_SDW_DEFAULT_COLUMNS; params->row = AMD_SDW_DEFAULT_ROWS; prop = &amd_manager->bus.prop; prop->clk_freq = &amd_sdw_freq_tbl[0]; prop->mclk_freq = AMD_SDW_BUS_BASE_FREQ; + prop->max_clk_freq = AMD_SDW_DEFAULT_CLK_FREQ; ret = sdw_bus_master_add(&amd_manager->bus, dev, dev->fwnode); if (ret) { -- 2.41.0

2 3

[PATCH 0/7] ASoC: Intel: Soundwire related board and match updates
by Peter Ujfalusi 01 Dec '23

01 Dec '23

Hi, A small update for SDW machine support: Small fixes for sof_sdw machine driver Support for rt722 New TGL/MTL and LNL match for new configurations Regards, Peter --- Bard Liao (1): ASoC: Intel: soc-acpi-intel-tgl-match: add cs42l43 and cs56l56 support Chao Song (4): ASoC: Intel: sof_sdw: remove unused function declaration ASoC: Intel: sof_sdw: Add rt722 support ASoC: Intel: soc-acpi: add Gen4.1 SDCA board support for LNL RVP ASoC: Intel: soc-acpi-intel-mtl-match: Add rt722 support Mac Chiang (1): ASoC: Intel: soc-acpi: rt713+rt1316, no sdw-dmic config Peter Ujfalusi (1): ASoC: Intel: sof_sdw: Make use of dev_err_probe() sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 3 +- sound/soc/intel/boards/sof_sdw.c | 32 +++++- sound/soc/intel/boards/sof_sdw_common.h | 18 ++-- sound/soc/intel/boards/sof_sdw_rt722_sdca.c | 97 +++++++++++++++++++ .../boards/sof_sdw_rt_sdca_jack_common.c | 8 ++ .../intel/common/soc-acpi-intel-lnl-match.c | 71 ++++++++++++++ .../intel/common/soc-acpi-intel-mtl-match.c | 74 ++++++++++++++ .../intel/common/soc-acpi-intel-tgl-match.c | 78 +++++++++++++++ 9 files changed, 374 insertions(+), 8 deletions(-) create mode 100644 sound/soc/intel/boards/sof_sdw_rt722_sdca.c -- 2.43.0

4 17

[PATCH 00/17] Prepare the PowerQUICC QMC and TSA for the HDLC QMC driver
by Herve Codina 01 Dec '23

01 Dec '23

Hi, This series updates PowerQUICC QMC and TSA drivers to prepare the support for the QMC HDLC driver. Patches were previously sent as part of a full feature series: "Add support for QMC HDLC, framer infrastructure and PEF2256 framer" [1] The full feature series reached the v9 iteration. The v1 was sent the 07/25/2023 followed by the other iterations (07/26/2023, 08/09/2023, 08/18/2023, 09/12/2023, 09/22/2023, 09/28/2023, 10/11/23, 11/15/2023) and was ready to be merged in its v8. https://lore.kernel.org/linux-kernel/20231025123215.5caca7d4@kernel.org/ The lack of feedback from the Freescale SoC and the Quicc Engine maintainers (i.e. drivers/soc/fsl/qe/ to which the QMC and TSA drivers belong) blocks the entire full feature series. These patches are fixes and improvements to TSA and QMC drivers. These drivers were previously acked by Li Yang but without any feedback from Li Yang nor Qiang Zhao the series cannot move forward. In order to ease the review/merge, the full feature series has been split and this series contains patches related to the PowerQUICC SoC part (QMC and TSA). - Perform some fixes (patches 1 to 5) - Add support for child devices (patch 6) - Add QMC dynamic timeslot support (patches 7 to 17) From the original full feature series, a patches extraction without any modification was done. Best regards, Hervé [1]: https://lore.kernel.org/linux-kernel/20231115144007.478111-1-herve.codina@b… Patches extracted: - Patch 1..6 : full feature series patch 1..6 - Patch 7..17 : full feature series patch 9..19 Herve Codina (17): soc: fsl: cpm1: tsa: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix rx channel reset soc: fsl: cpm1: qmc: Extend the API to provide Rx status soc: fsl: cpm1: qmc: Remove inline function specifiers soc: fsl: cpm1: qmc: Add support for child devices soc: fsl: cpm1: qmc: Introduce available timeslots masks soc: fsl: cpm1: qmc: Rename qmc_setup_tsa* to qmc_init_tsa* soc: fsl: cpm1: qmc: Introduce qmc_chan_setup_tsa* soc: fsl: cpm1: qmc: Remove no more needed checks from qmc_check_chans() soc: fsl: cpm1: qmc: Check available timeslots in qmc_check_chans() soc: fsl: cpm1: qmc: Add support for disabling channel TSA entries soc: fsl: cpm1: qmc: Split Tx and Rx TSA entries setup soc: fsl: cpm1: qmc: Introduce is_tsa_64rxtx flag soc: fsl: cpm1: qmc: Handle timeslot entries at channel start() and stop() soc: fsl: cpm1: qmc: Remove timeslots handling from setup_chan() soc: fsl: cpm1: qmc: Introduce functions to change timeslots at runtime drivers/soc/fsl/qe/qmc.c | 592 +++++++++++++++++++++++++++------- drivers/soc/fsl/qe/tsa.c | 22 +- include/soc/fsl/qe/qmc.h | 27 +- sound/soc/fsl/fsl_qmc_audio.c | 2 +- 4 files changed, 506 insertions(+), 137 deletions(-) -- 2.42.0

2 20

[PATCH 00/13] ASoC: SOF: IPC path handling and fallback support
by Peter Ujfalusi 30 Nov '23

30 Nov '23

Hi, The main aim of the series is to provide a mechanism to fallback to 'older' IPC versions in case the desired one is missing either a firmware or topology file. It is going to make the life of users and distributions if we are going to start transition existing IPC3 platforms to IPC4 (CAVS2.5) and we might have missed some topology file to convert for example. In that case the kernel will fallback to IPC3 without audio regression. To be able to support this we needed to change the probe sequence to know the topology filename earlier and check if it is present in the filesystem. No functional changes for now, the default IPC versions have not been changed. Regards, Peter --- Peter Ujfalusi (13): ASoC: SOF: Move sof_of_machine_select() to sof-of-dev.c from sof-audio.c ASoC: SOF: Move sof_machine_* functions from sof-audio.c to core.c ASoC: SOF: Add placeholder for platform IPC type and path overrides ASoC: SOF: sof-acpi-dev: Save the default IPC type and path overrides ASoC: SOF: sof-of-dev: Save the default IPC type and path overrides ASoC: SOF: sof-pci-dev: Save the default IPC type and path overrides ASoC: SOF: core: Implement firmware, topology path setup in core ASoC: SOF: sof-acpi-dev: Rely on core to create the file paths ASoC: SOF: sof-of-dev: Rely on core to create the file paths ASoC: SOF: sof-pci-dev: Rely on core to create the file paths ASoC: SOF: core: Add helper for initialization of paths, ops ASoC: SOF: Intel: Do not use resource managed allocation for ipc4_data ASoC: SOF: core: Implement IPC version fallback if firmware files are missing include/sound/sof.h | 15 ++ sound/soc/sof/Kconfig | 11 ++ sound/soc/sof/Makefile | 3 +- sound/soc/sof/core.c | 289 +++++++++++++++++++++++----- sound/soc/sof/fw-file-profile.c | 322 ++++++++++++++++++++++++++++++++ sound/soc/sof/intel/apl.c | 2 +- sound/soc/sof/intel/cnl.c | 2 +- sound/soc/sof/intel/hda-dai.c | 3 + sound/soc/sof/intel/icl.c | 2 +- sound/soc/sof/intel/lnl.c | 2 +- sound/soc/sof/intel/mtl.c | 2 +- sound/soc/sof/intel/skl.c | 2 +- sound/soc/sof/intel/tgl.c | 2 +- sound/soc/sof/sof-acpi-dev.c | 16 +- sound/soc/sof/sof-audio.c | 120 ------------ sound/soc/sof/sof-of-dev.c | 36 +++- sound/soc/sof/sof-of-dev.h | 9 + sound/soc/sof/sof-pci-dev.c | 102 ++-------- sound/soc/sof/sof-priv.h | 9 +- 19 files changed, 671 insertions(+), 278 deletions(-) create mode 100644 sound/soc/sof/fw-file-profile.c -- 2.43.0

2 14

[PATCH] ASoC: qcom: audioreach: Add 4 channel support
by Krzysztof Kozlowski 30 Nov '23

30 Nov '23

Add support four channel streams. Map channel 3 and 4 to left/right surround ("quad(side)" from ffmpeg standard channel list) to match what is in qdsp6/q6dsp-common.c driver. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- quad(side): https://trac.ffmpeg.org/wiki/AudioChannelManipulation#Listchannelnamesandst… --- sound/soc/qcom/qdsp6/audioreach.c | 25 ++++++++++++++++++++++--- 1 file changed, 22 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c index 5974c7929dd3..4dd2a2e261c9 100644 --- a/sound/soc/qcom/qdsp6/audioreach.c +++ b/sound/soc/qcom/qdsp6/audioreach.c @@ -833,6 +833,11 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph, } else if (num_channels == 2) { media_format->channel_mapping[0] = PCM_CHANNEL_L; media_format->channel_mapping[1] = PCM_CHANNEL_R; + } else if (num_channels == 4) { + media_format->channel_mapping[0] = PCM_CHANNEL_FL; + media_format->channel_mapping[1] = PCM_CHANNEL_FR; + media_format->channel_mapping[2] = PCM_CHANNEL_LS; + media_format->channel_mapping[3] = PCM_CHANNEL_RS; } rc = q6apm_send_cmd_sync(graph->apm, pkt, 0); @@ -869,6 +874,11 @@ static int audioreach_set_compr_media_format(struct media_format *media_fmt_hdr, } else if (mcfg->num_channels == 2) { mp3_cfg->channel_mapping[0] = PCM_CHANNEL_L; mp3_cfg->channel_mapping[1] = PCM_CHANNEL_R; + } else if (mcfg->num_channels == 4) { + mp3_cfg->channel_mapping[0] = PCM_CHANNEL_FL; + mp3_cfg->channel_mapping[1] = PCM_CHANNEL_FR; + mp3_cfg->channel_mapping[2] = PCM_CHANNEL_LS; + mp3_cfg->channel_mapping[3] = PCM_CHANNEL_RS; } break; case SND_AUDIOCODEC_AAC: @@ -1057,7 +1067,7 @@ static int audioreach_pcm_set_media_format(struct q6apm_graph *graph, int rc, payload_size; struct gpr_pkt *pkt; - if (num_channels > 2) { + if (num_channels > 4) { dev_err(graph->dev, "Error: Invalid channels (%d)!\n", num_channels); return -EINVAL; } @@ -1094,7 +1104,11 @@ static int audioreach_pcm_set_media_format(struct q6apm_graph *graph, } else if (num_channels == 2) { media_cfg->channel_mapping[0] = PCM_CHANNEL_L; media_cfg->channel_mapping[1] = PCM_CHANNEL_R; - + } else if (num_channels == 4) { + media_cfg->channel_mapping[0] = PCM_CHANNEL_FL; + media_cfg->channel_mapping[1] = PCM_CHANNEL_FR; + media_cfg->channel_mapping[2] = PCM_CHANNEL_LS; + media_cfg->channel_mapping[3] = PCM_CHANNEL_RS; } rc = q6apm_send_cmd_sync(graph->apm, pkt, 0); @@ -1116,7 +1130,7 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph, struct gpr_pkt *pkt; void *p; - if (num_channels > 2) { + if (num_channels > 4) { dev_err(graph->dev, "Error: Invalid channels (%d)!\n", num_channels); return -EINVAL; } @@ -1158,6 +1172,11 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph, else if (num_channels == 2) { cfg->channel_mapping[0] = PCM_CHANNEL_L; cfg->channel_mapping[1] = PCM_CHANNEL_R; + } else if (num_channels == 4) { + cfg->channel_mapping[0] = PCM_CHANNEL_FL; + cfg->channel_mapping[1] = PCM_CHANNEL_FR; + cfg->channel_mapping[2] = PCM_CHANNEL_LS; + cfg->channel_mapping[3] = PCM_CHANNEL_RS; } } else { rc = audioreach_set_compr_media_format(header, p, mcfg); -- 2.34.1

2 3

[PATCH 1/2] ASoC: qcom: Add helper for allocating Soundwire stream runtime
by Krzysztof Kozlowski 30 Nov '23

30 Nov '23

Newer Qualcomm SoC soundcards will need to allocate Soundwire stream runtime in their startup op. The code will be exactly the same for all soundcards, so add a helper for that. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/qcom/sdw.c | 45 +++++++++++++++++++++++++++++++++++++++++++- sound/soc/qcom/sdw.h | 1 + 2 files changed, 45 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/sdw.c b/sound/soc/qcom/sdw.c index dd275123d31d..77dbe0c28b29 100644 --- a/sound/soc/qcom/sdw.c +++ b/sound/soc/qcom/sdw.c @@ -1,5 +1,5 @@ // SPDX-License-Identifier: GPL-2.0 -// Copyright (c) 2018, Linaro Limited. +// Copyright (c) 2018-2023, Linaro Limited. // Copyright (c) 2018, The Linux Foundation. All rights reserved. #include <dt-bindings/sound/qcom,q6afe.h> @@ -7,6 +7,49 @@ #include <sound/soc.h> #include "sdw.h" +/** + * qcom_snd_sdw_startup() - Helper to start Soundwire stream for SoC audio card + * @substream: The PCM substream from audio, as passed to snd_soc_ops->startup() + * + * Helper for the SoC audio card (snd_soc_ops->startup()) to allocate and set + * Soundwire stream runtime to each codec DAI. + * + * The shutdown() callback should call sdw_release_stream() on the same + * sdw_stream_runtime. + * + * Return: 0 or errno + */ +int qcom_snd_sdw_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + struct sdw_stream_runtime *sruntime; + struct snd_soc_dai *codec_dai; + int ret, i; + + sruntime = sdw_alloc_stream(cpu_dai->name); + if (!sruntime) + return -ENOMEM; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + ret = snd_soc_dai_set_stream(codec_dai, sruntime, + substream->stream); + if (ret < 0 && ret != -ENOTSUPP) { + dev_err(rtd->dev, "Failed to set sdw stream on %s\n", + codec_dai->name); + goto err_set_stream; + } + } + + return 0; + +err_set_stream: + sdw_release_stream(sruntime); + + return ret; +} +EXPORT_SYMBOL_GPL(qcom_snd_sdw_startup); + int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream, struct sdw_stream_runtime *sruntime, bool *stream_prepared) diff --git a/sound/soc/qcom/sdw.h b/sound/soc/qcom/sdw.h index d74cbb84da13..392e3455f1b1 100644 --- a/sound/soc/qcom/sdw.h +++ b/sound/soc/qcom/sdw.h @@ -6,6 +6,7 @@ #include <linux/soundwire/sdw.h> +int qcom_snd_sdw_startup(struct snd_pcm_substream *substream); int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream, struct sdw_stream_runtime *runtime, bool *stream_prepared); -- 2.34.1

3 8

[PATCH 00/10] ASoC: Convert Cirrus codecs to GPIO descriptors
by Linus Walleij 30 Nov '23

30 Nov '23

This series walks over the Cirrus Logic ASoC drivers and clean out the use of legacy GPIO numbers and legacy GPIO APIs. The CS4271 affects an ASoC driver for EP93xx which Nikita is actively working on moving over to device tree, so I don't know about that patch specifically, but I think the collision would be max "the file was deleted". Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Linus Walleij (10): ASoC: cs35l32: Drop legacy include ASoC: cs35l33: Fix GPIO name and drop legacy include ASoC: cs35l34: Fix GPIO name and drop legacy include ASoC: cs35l35: Drop legacy includes ASoC: cs35l36: Drop legacy includes ASoC: cs4271: Convert to GPIO descriptors ASoC: cirrus: edb93xx: Drop legacy include ASoC: cs42l42: Drop legacy include ASoC: cs43130: Drop legacy includes ASoC: cs4349: Drop legacy include arch/arm/mach-ep93xx/edb93xx.c | 32 +++++++++++++++++++++++++---- arch/arm/mach-ep93xx/vision_ep9307.c | 12 ++++++++++- include/sound/cs4271.h | 1 - sound/soc/cirrus/edb93xx.c | 1 - sound/soc/codecs/cs35l32.c | 1 - sound/soc/codecs/cs35l33.c | 4 +--- sound/soc/codecs/cs35l34.c | 4 +--- sound/soc/codecs/cs35l35.c | 2 -- sound/soc/codecs/cs35l36.c | 2 -- sound/soc/codecs/cs4271.c | 39 ++++++++++++------------------------ sound/soc/codecs/cs42l42.c | 1 - sound/soc/codecs/cs42l42.h | 2 +- sound/soc/codecs/cs43130.c | 2 -- sound/soc/codecs/cs4349.c | 1 - 14 files changed, 55 insertions(+), 49 deletions(-) --- base-commit: 267aea213ae042f779a8054401a8a5f301518605 change-id: 20231129-descriptors-sound-cirrus-522d9061808e Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

4 14

[PATCH 0/2] ASoC: SOF: ipc4-topology: Correct IPC message struct layouts
by Peter Ujfalusi 29 Nov '23

29 Nov '23

Hi, The SRC and GAIN module defined it's kernel side private struct in a way that part of that is used as IPC message without making sure of proper alignment and padding, which happens to work but it is better to be not 'clever' about these sort of things. The two patch corects this overlooked detail. Regards, Peter --- Peter Ujfalusi (2): ASoC: SOF: ipc4-topology: Correct data structures for the SRC module ASoC: SOF: ipc4-topology: Correct data structures for the GAIN module sound/soc/sof/ipc4-control.c | 20 +++++++------- sound/soc/sof/ipc4-topology.c | 52 +++++++++++++++++------------------ sound/soc/sof/ipc4-topology.h | 34 +++++++++++++++++------ 3 files changed, 61 insertions(+), 45 deletions(-) -- 2.43.0

2 3