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

[PATCH AUTOSEL 6.4 10/58] ASoC: Intel: sof_sdw: add quirk for MTL RVP
by Sasha Levin 24 Jul '23

24 Jul '23

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 289e1df00e49a229a1c924c059242e759a552f01 ] We should use RT711_JD2_100K for on board rt711. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Link: https://lore.kernel.org/r/20230512173305.65399-4-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 144f082c63fda..db99f7fefdfc7 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -415,6 +415,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(RT711_JD1 | SOF_SDW_TGL_HDMI), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Meteor Lake Client Platform"), + }, + .driver_data = (void *)(RT711_JD2_100K), + }, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 6.4 09/58] ALSA: emu10k1: roll up loops in DSP setup code for Audigy
by Sasha Levin 24 Jul '23

24 Jul '23

From: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> [ Upstream commit 8cabf83c7aa54530e699be56249fb44f9505c4f3 ] There is no apparent reason for the massive code duplication. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> Link: https://lore.kernel.org/r/20230510173917.3073107-3-oswald.buddenhagen@gmx.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/emu10k1/emufx.c | 112 +++----------------------------------- 1 file changed, 9 insertions(+), 103 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 3f64ccab0e632..fba19a854f27a 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1559,14 +1559,8 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) gpr += 2; /* Master volume (will be renamed later) */ - A_OP(icode, &ptr, iMAC0, A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS)); + for (z = 0; z < 8; z++) + A_OP(icode, &ptr, iMAC0, A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS)); snd_emu10k1_init_mono_control(&controls[nctl++], "Wave Master Playback Volume", gpr, 0); gpr += 2; @@ -1653,102 +1647,14 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) dev_dbg(emu->card->dev, "emufx.c: gpr=0x%x, tmp=0x%x\n", gpr, tmp); */ - /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ - /* A_P16VIN(0) is delayed by one sample, - * so all other A_P16VIN channels will need to also be delayed - */ - /* Left ADC in. 1 of 2 */ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) ); - /* Right ADC in 1 of 2 */ - gpr_map[gpr++] = 0x00000000; - /* Delaying by one sample: instead of copying the input - * value A_P16VIN to output A_FXBUS2 as in the first channel, - * we use an auxiliary register, delaying the value by one - * sample - */ - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000); - /* For 96kHz mode */ - /* Left ADC in. 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000); - /* Right ADC in 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000); - /* Pavel Hofman - we still have voices, A_FXBUS2s, and - * A_P16VINs available - - * let's add 8 more capture channels - total of 16 - */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x10)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x12)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x14)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x16)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x18)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1a)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1c)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1e)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf), - A_C_00000000, A_C_00000000); + /* A_P16VIN(0) is delayed by one sample, so all other A_P16VIN channels + * will need to also be delayed; we use an auxiliary register for that. */ + for (z = 1; z < 0x10; z++) { + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr), A_FXBUS2(z * 2) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr), A_P16VIN(z), A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + } } #if 0 -- 2.39.2

1 0

[PATCH] ASoC: rt5668: add missed regulator_bulk_disable
by Zhang Shurong 23 Jul '23

23 Jul '23

The driver forgets to call regulator_bulk_disable() Add the missed call to fix it. Fixes: d59fb2856223 ("ASoC: rt5668: add rt5668B codec driver") Signed-off-by: Zhang Shurong <zhang_shurong(a)foxmail.com> --- sound/soc/codecs/rt5668.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index f04c810fd710..d66c9e9b9185 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -2452,6 +2452,13 @@ static void rt5668_calibrate(struct rt5668_priv *rt5668) } +static void rt5668_i2c_disable_regulators(void *data) +{ + struct rt5668_priv *rt5668 = data; + + regulator_bulk_disable(ARRAY_SIZE(rt5668->supplies), rt5668->supplies); +} + static int rt5668_i2c_probe(struct i2c_client *i2c) { struct rt5668_platform_data *pdata = dev_get_platdata(&i2c->dev); @@ -2490,6 +2497,11 @@ static int rt5668_i2c_probe(struct i2c_client *i2c) return ret; } + ret = devm_add_action_or_reset(&i2c->dev, rt5668_i2c_disable_regulators, + rt5668); + if (ret) + return ret; + ret = regulator_bulk_enable(ARRAY_SIZE(rt5668->supplies), rt5668->supplies); if (ret != 0) { -- 2.41.0

1 0

alsactl: add define to compile with glibc 2.38
by GitHub pull_request - opened 22 Jul '23

22 Jul '23

alsa-project/alsa-utils pull request #225 was opened from heitbaum: strlcat and strlcpy have been added to glibc 2.38. update the defines to use the glibc versions, and not conflict with string.h. ref: - https://sourceware.org/git/?p=glibc.git;a=commit;h=454a20c8756c9c1d55419153… Request URL : https://github.com/alsa-project/alsa-utils/pull/225 Patch URL : https://github.com/alsa-project/alsa-utils/pull/225.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

USB soundcard no longer working after update from 1.2.8 to 1.2.9
by GitHub issues - opened 22 Jul '23

22 Jul '23

alsa-project/alsa-lib issue #336 was opened from bothesven: I use a "Teufel Cinebar Ultima" as a usb sound card with driver snd-usb-audio. This works with alsa-lib, alsa-ucm-conf and lib32-alsa-lib in version 1.2.8. When these libraries are updated to 1.2.9, any player just hangs when this sound card is selected as output. [alsa-info.txt.gz](https://github.com/alsa-project/alsa-lib/files/12134686/a… Issue URL : https://github.com/alsa-project/alsa-lib/issues/336 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: fsl_spdif: Silence output on stop
by Matus Gajdos 21 Jul '23

21 Jul '23

Clear TX registers on stop to prevent the SPDIF interface from sending last written word over and over again. Fixes: a2388a498ad2 ("ASoC: fsl: Add S/PDIF CPU DAI driver") Signed-off-by: Matus Gajdos <matuszpd(a)gmail.com> --- sound/soc/fsl/fsl_spdif.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 95e639711eba..95bb8b10494a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -755,6 +755,8 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: regmap_update_bits(regmap, REG_SPDIF_SCR, dmaen, 0); regmap_update_bits(regmap, REG_SPDIF_SIE, intr, 0); + regmap_write(regmap, REG_SPDIF_STL, 0x0); + regmap_write(regmap, REG_SPDIF_STR, 0x0); break; default: return -EINVAL; -- 2.25.1

3 2

[PATCH v5 0/6] Add cs42l43 PC focused SoundWire CODEC
by Charles Keepax 21 Jul '23

21 Jul '23

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. 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 | 1189 +++++++++ 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 | 281 ++ 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 | 969 +++++++ sound/soc/codecs/cs42l43-sdw.c | 74 + sound/soc/codecs/cs42l43.c | 2278 +++++++++++++++++ sound/soc/codecs/cs42l43.h | 131 + 26 files changed, 7636 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

2 11

[PATCH 2/2] ASoC: rt1316: fix key tone missing
by shumingf＠realtek.com 21 Jul '23

21 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> This patch disables the soft ramp to output immediately. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1316-sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index ce2246a2591c..feea11149d93 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -134,7 +134,7 @@ static const struct reg_sequence rt1316_blind_write[] = { { 0xc093, 0x80 }, { 0xc09d, 0x80 }, { 0xc0b0, 0x77 }, - { 0xc010, 0xa5 }, + { 0xc010, 0x25 }, { 0xc050, 0x83 }, { 0x2f55, 0x03 }, { 0x3217, 0xb5 }, -- 2.34.1

3 2

[PATCH v1 00/11] Fix support for System Suspend for CS35L41 HDA
by Stefan Binding 21 Jul '23

21 Jul '23

There have been a couple of customer reports of intermittant issues after system resume, where sometimes the DSP firmware stops responding. Investigations into this issue show that there is a race between receiving a prepare from the HDA core, and the firmware reload which is started by the system resume. This can causes the Global Enable on the CS35L41 to be enabled during the firmware load, which can sometimes cause issues in the DSP. The existing system resume behaviour also did not resume the audio, if audio was previously playing when it was suspended. In addition, during investigation, it was found there were additional problems in the System Resume sequence, as well as the Playback sequence with External Boost, where the driver does not correctly follow its enable sequence for this mode. This can cause additional issues such as pops and clicks. This chain intends to correct the sequences for playback and system suspend/resume so that the driver: obeys the external boost enable sequence; resumes audio on system resume; and avoids the race condition on firmware load and playback during system resume. Stefan Binding (11): ALSA: cs35l41: Use mbox command to enable speaker output for external boost ALSA: hda: cs35l41: Check mailbox status of pause command after firmware load ALSA: hda: cs35l41: Ensure we correctly re-sync regmap before system suspending. ALSA: hda: cs35l41: Ensure we pass up any errors during system suspend. ALSA: hda: cs35l41: Move Play and Pause into separate functions ALSA: hda: hda_component: Add pre and post playback hooks to hda_component ALSA: hda: cs35l41: Use pre and post playback hooks ALSA: hda: cs35l41: Rework System Suspend to ensure correct call separation ALSA: hda/realtek: Support pre-/post- playback hooks for cs35l41 ALSA: hda: cs35l41: Add device_link between HDA and cs35l41_hda ALSA: hda: cs35l41: Ensure amp is only unmuted during playback include/sound/cs35l41.h | 5 +- sound/pci/hda/cs35l41_hda.c | 288 +++++++++++++++++++++++++-------- sound/pci/hda/hda_component.h | 2 + sound/pci/hda/patch_realtek.c | 10 +- sound/soc/codecs/cs35l41-lib.c | 118 ++++++++++++-- sound/soc/codecs/cs35l41.c | 18 +-- 6 files changed, 343 insertions(+), 98 deletions(-) -- 2.34.1

3 16

[PATCH 0/2] ALSA: hda/hdmi: Add Intel Lunar Lake support
by Peter Ujfalusi 21 Jul '23

21 Jul '23

Hi, after fixing the ordering, add the HDMI codec ID for LNL Regards, Peter --- Kai Vehmanen (2): ALSA: hda/hdmi: keep codec entries in numerical order ALSA: hda: add HDMI codec ID for Intel LNL sound/pci/hda/patch_hdmi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) -- 2.41.0

2 3