June 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: cs35l45: Make bmc150_magn_remove() return void
by Uwe Kleine-König 07 Jun '22

07 Jun '22

cs35l45_remove() always returns zero. Make it return no value which makes it easier to see in the callers that there is no error to handle. Also the return value of i2c driver remove callbacks is ignored anyway. This prepares making i2c remove callbacks return void, too. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/cs35l45-i2c.c | 4 +++- sound/soc/codecs/cs35l45.c | 4 +--- sound/soc/codecs/cs35l45.h | 2 +- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs35l45-i2c.c b/sound/soc/codecs/cs35l45-i2c.c index 18ebf1e14b91..53a57320d0ab 100644 --- a/sound/soc/codecs/cs35l45-i2c.c +++ b/sound/soc/codecs/cs35l45-i2c.c @@ -40,7 +40,9 @@ static int cs35l45_i2c_remove(struct i2c_client *client) { struct cs35l45_private *cs35l45 = i2c_get_clientdata(client); - return cs35l45_remove(cs35l45); + cs35l45_remove(cs35l45); + + return 0; } static const struct of_device_id cs35l45_of_match[] = { diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 7822626d3693..64a3abe14010 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -669,7 +669,7 @@ int cs35l45_probe(struct cs35l45_private *cs35l45) } EXPORT_SYMBOL_GPL(cs35l45_probe); -int cs35l45_remove(struct cs35l45_private *cs35l45) +void cs35l45_remove(struct cs35l45_private *cs35l45) { pm_runtime_disable(cs35l45->dev); @@ -677,8 +677,6 @@ int cs35l45_remove(struct cs35l45_private *cs35l45) regulator_disable(cs35l45->vdd_a); /* VDD_BATT must be the last to power-off */ regulator_disable(cs35l45->vdd_batt); - - return 0; } EXPORT_SYMBOL_GPL(cs35l45_remove); diff --git a/sound/soc/codecs/cs35l45.h b/sound/soc/codecs/cs35l45.h index 4e266d19cd1c..680891bcfce9 100644 --- a/sound/soc/codecs/cs35l45.h +++ b/sound/soc/codecs/cs35l45.h @@ -212,6 +212,6 @@ extern const struct regmap_config cs35l45_spi_regmap; int cs35l45_apply_patch(struct cs35l45_private *cs43l45); unsigned int cs35l45_get_clk_freq_id(unsigned int freq); int cs35l45_probe(struct cs35l45_private *cs35l45); -int cs35l45_remove(struct cs35l45_private *cs35l45); +void cs35l45_remove(struct cs35l45_private *cs35l45); #endif /* CS35L45_H */ base-commit: 0d463d016000d68d7e982720b5e4380b2d83409a -- 2.35.1

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[PATCH] ASoC: amd: acp: fix typo in comment
by Julia Lawall 07 Jun '22

07 Jun '22

Spelling mistake (triple letters) in comment. Detected with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- sound/soc/amd/acp/acp-pdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-pdm.c b/sound/soc/amd/acp/acp-pdm.c index 424c6e0bb9d6..7a0b26a30051 100644 --- a/sound/soc/amd/acp/acp-pdm.c +++ b/sound/soc/amd/acp/acp-pdm.c @@ -174,7 +174,7 @@ static void acp_dmic_dai_shutdown(struct snd_pcm_substream *substream, struct acp_dev_data *adata = dev_get_drvdata(dev); u32 ext_int_ctrl; - /* Disable DMIC interrrupts */ + /* Disable DMIC interrupts */ ext_int_ctrl = readl(adata->acp_base + ACP_EXTERNAL_INTR_CNTL); ext_int_ctrl |= ~PDM_DMA_INTR_MASK; writel(ext_int_ctrl, adata->acp_base + ACP_EXTERNAL_INTR_CNTL);

2 1

[PATCH] ASoC: stm32: dfsdm: fix typo in comment
by Julia Lawall 07 Jun '22

07 Jun '22

Spelling mistake (triple letters) in comment. Detected with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- sound/soc/stm/stm32_adfsdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 6ee714542b84..122805160e70 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -296,7 +296,7 @@ static int stm32_adfsdm_pcm_new(struct snd_soc_component *component, static int stm32_adfsdm_dummy_cb(const void *data, void *private) { /* - * This dummmy callback is requested by iio_channel_get_all_cb() API, + * This dummy callback is requested by iio_channel_get_all_cb() API, * but the stm32_dfsdm_get_buff_cb() API is used instead, to optimize * DMA transfers. */

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[PATCH v2] ASoC: qcom: soundwire: Add support for controlling audio CGCR from HLOS
by Srinivasa Rao Mandadapu 07 Jun '22

07 Jun '22

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> --- This patch set depends on: -- Clock driver patches for CGCR reset control support. --- https://patchwork.kernel.org/project/linux-arm-msm/list/?series=638002 --- https://patchwork.kernel.org/project/linux-arm-msm/list/?series=637998 Changes since v1: -- Add audio cgcr reset control in runtime PM resume handler. -- Update dependency list. drivers/soundwire/qcom.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index da1ad7e..445e481 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -13,6 +13,7 @@ #include <linux/of_device.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <linux/slab.h> #include <linux/pm_wakeirq.h> #include <linux/slimbus.h> @@ -142,6 +143,7 @@ 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 @@ -656,6 +658,8 @@ 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 */ @@ -1333,6 +1337,10 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.compute_params = &qcom_swrm_compute_params; ctrl->bus.clk_stop_timeout = 300; + 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) goto err_clk; @@ -1486,6 +1494,8 @@ static int __maybe_unused swrm_runtime_resume(struct device *dev) qcom_swrm_get_device_status(ctrl); sdw_handle_slave_status(&ctrl->bus, ctrl->status); } else { + reset_control_reset(ctrl->audio_cgcr); + ctrl->reg_write(ctrl, SWRM_MCP_BUS_CTRL, SWRM_MCP_BUS_CLK_START); ctrl->reg_write(ctrl, SWRM_INTERRUPT_CLEAR, SWRM_INTERRUPT_STATUS_MASTER_CLASH_DET); -- 2.7.4

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[PATCH] ASoC: fsl_asrc_dma: enable dual fifo for ASRC P2P
by Shengjiu Wang 07 Jun '22

07 Jun '22

The SSI and SPDIF has dual fifos, enhance P2P for these case with using the sdma_peripheral_config struct Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc_dma.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 5038faf035cb..aaf7993935b7 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -129,6 +129,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct snd_pcm_hw_params *params) { enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + enum sdma_peripheral_type be_peripheral_type = IMX_DMATYPE_SSI; struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct snd_dmaengine_dai_dma_data *dma_params_fe = NULL; @@ -139,6 +140,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct snd_soc_component *component_be = NULL; struct fsl_asrc *asrc = pair->asrc; struct dma_slave_config config_fe, config_be; + struct sdma_peripheral_config audio_config; enum asrc_pair_index index = pair->index; struct device *dev = component->dev; struct device_node *of_dma_node; @@ -221,6 +223,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, /* Get DMA request of Back-End */ tmp_data = tmp_chan->private; pair->dma_data.dma_request = tmp_data->dma_request; + be_peripheral_type = tmp_data->peripheral_type; if (!be_chan) dma_release_channel(tmp_chan); @@ -268,6 +271,17 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, config_be.dst_addr_width = buswidth; config_be.dst_maxburst = dma_params_be->maxburst; + memset(&audio_config, 0, sizeof(audio_config)); + config_be.peripheral_config = &audio_config; + config_be.peripheral_size = sizeof(audio_config); + + if (tx && (be_peripheral_type == IMX_DMATYPE_SSI_DUAL || + be_peripheral_type == IMX_DMATYPE_SPDIF)) + audio_config.n_fifos_dst = 2; + if (!tx && (be_peripheral_type == IMX_DMATYPE_SSI_DUAL || + be_peripheral_type == IMX_DMATYPE_SPDIF)) + audio_config.n_fifos_src = 2; + if (tx) { config_be.src_addr = asrc->paddr + asrc->get_fifo_addr(OUT, index); config_be.dst_addr = dma_params_be->addr; -- 2.17.1

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[PATCH] ALSA: hda/realtek - Add HW8326 support
by huangwenhui 07 Jun '22

07 Jun '22

Added the support of new Huawei codec HW8326. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..775474f00141 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -651,6 +651,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x1095, "Silicon Image" }, { 0x10de, "Nvidia" }, { 0x10ec, "Realtek" }, + { 0x19e5, "Huawei" }, { 0x1102, "Creative" }, { 0x1106, "VIA" }, { 0x111d, "IDT" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f3ad454b3fbf..0f32c0d17a3f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10095,6 +10107,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11479,6 +11492,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0236, "ALC236", patch_alc269), HDA_CODEC_ENTRY(0x10ec0245, "ALC245", patch_alc269), HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269), HDA_CODEC_ENTRY(0x10ec0257, "ALC257", patch_alc269), HDA_CODEC_ENTRY(0x10ec0260, "ALC260", patch_alc260), -- 2.20.1

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[PATCH] ALSA: hda/conexant - Fix loopback issue with CX20632
by huangwenhui 07 Jun '22

07 Jun '22

On a machine with CX20632,Alsamixer doesn't have 'Loopback Mixing' and 'Line'. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> --- sound/pci/hda/patch_conexant.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index aa360a0af284..1248d1a51cf0 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1052,6 +1052,13 @@ static int patch_conexant_auto(struct hda_codec *codec) snd_hda_pick_fixup(codec, cxt5051_fixup_models, cxt5051_fixups, cxt_fixups); break; + case 0x14f15098: + codec->pin_amp_workaround = 1; + spec->gen.mixer_nid = 0x22; + spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; + snd_hda_pick_fixup(codec, cxt5066_fixup_models, + cxt5066_fixups, cxt_fixups); + break; case 0x14f150f2: codec->power_save_node = 1; fallthrough; -- 2.20.1

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[PATCH] ALSA: hda: MTL: add HD Audio PCI ID and HDMI codec vendor ID
by Pierre-Louis Bossart 07 Jun '22

07 Jun '22

From: Yong Zhi <yong.zhi(a)intel.com> Add HD Audio PCI ID for Intel Meteorlake platform. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 3 +++ sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0a83eb6b88b1f..2c2257ec50929 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2525,6 +2525,9 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, { PCI_DEVICE(0x8086, 0x51cf), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Meteorlake-P */ + { PCI_DEVICE(0x8086, 0x7E28), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 31fe417955712..6c209cd26c0ca 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4554,6 +4554,7 @@ HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x8086281f, "Raptorlake-P HDMI", patch_i915_adlp_hdmi), +HDA_CODEC_ENTRY(0x8086281d, "Meteorlake HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- 2.34.1

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ASUS ROG Maximus Z690 Hero (ALC4082) - No audio from microphone, no audio on the front panel
by GitHub issues - edited 06 Jun '22

06 Jun '22

alsa-project/alsa-ucm-conf issue #165 was edited from Lastique: I'm having a problem with getting audio from the microphone connected to the rear panel jack. Also, if I connect the headset to the front panel headphone/mic jacks, I get neither audio out or in. My main concern is the rear panel jack though. The motherboard is ASUS ROG Maximus Z690 Hero, ALC4082 USB Audio chipset. I have tried this: 1. Installed linux kernel 5.18.0. 2. Copied `ucm2` from alsa-ucm-conf master (rev ef3b6513b6581705d36432428a7f6a4ae23b1276 which is after https://github.com/alsa-project/alsa-ucm-conf/pull/143 was merged) to `/usr/share/alsa`. 3. In `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` added hardware id of my USB Audio device (`0b05:1a27`) to the `realtek-alc4080` rule. Also added the hardware id in `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf`, in the `asus-rog-usb` block condition. 4. Rebooted. 5. The headset is connected to the rear panel (Front out and Mic in jacks; also tried Line in). There are three devices for the USB Audio card, and there are three volume controls in alsamixer: "Line", "Mic" and "Analog In". The problem is that there is no audio coming from the first two devices (which are presumably "Line" and "Mic"). I can record audio from the third device ("Analog In") when the microphone is connected to Mic jack. Unfortunately, "Analog In" cannot be selected in PulseAudio for some reason (I created a [bug](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1364) for PA), which means no audio can be recorded. I also tried to connect the headset to the front panel connectors, but got no audio output nor capture. This is not my intended configuration though, I normally connect the headset to the rear panel. `arecord` output: [arecord.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838239/ar… `lsusb` output: [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838250/lsus… `alsa-info.sh` output: [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8846120/… This is on Kubuntu 22.04. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/165 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 0/2] ALSA: usb-audio: Fix regressions for Saffire 6
by Takashi Iwai 06 Jun '22

06 Jun '22

Hi, this is a small patch set to address the still remaining regressions for Focusrite Saffire 6. Takashi === Takashi Iwai (2): ALSA: usb-audio: Skip generic sync EP parse for secondary EP ALSA: usb-audio: Set up (implicit) sync for Saffire 6 sound/usb/pcm.c | 5 ++++- sound/usb/quirks-table.h | 7 ++++++- 2 files changed, 10 insertions(+), 2 deletions(-) -- 2.35.3

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