July 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH 00/27] ALSA: hda/cirrus: Split generic cirrus HDA codecs and CS8490 bridge into separate modules.
by Vitaly Rodionov 27 Jul '21

27 Jul '21

This series of patches splits generic cirrus HDA codecs and CS8490 bridge into separate modules, adds support for multiple companion codecs connected to CS8409, and also adds support for new DELL HW platform. CS8409 part is not really a HDA codec, it is a HDA bridge where companion codecs (up to 16) can be attached. With growing number of supported configurations and platforms, patch_cirrus is getting less and less transparent and maintainable. So, the logical step is to separate generic Cirrus HDA codecs support and Cirrus HDA bridge support. Lots of improvements to existing functionality, code clean-up and refactoring, remove duplicated/redundant code, improve I2C functions etc. Add support for new DELL HW platform with 2 CS42L42 codecs for front and rear jacks. Lucas Tanure (12): ALSA: hda/cirrus: Move CS8409 HDA bridge to separate module ALSA: hda/cs8409: Move arrays of configuration to a new file ALSA: hda/cs8409: Disable unsolicited response for the first boot ALSA: hda/cs8409: Prevent I2C access during suspend time ALSA: hda/cs8409: Generalize volume controls ALSA: hda/cs8409: Dont disable I2C clock between consecutive accesses ALSA: hda/cs8409: Avoid setting the same I2C address for every access ALSA: hda/cs8409: Avoid re-setting the same page as the last access ALSA: hda/cs8409: Support i2c bulk read/write functions ALSA: hda/cs8409: Separate CS8409, CS42L42 and project functions ALSA: hda/cs8409: Move codec properties to its own struct ALSA: hda/cs8409: Add support for dolphin Stefan Binding (15): ALSA: hda/cs8409: Use enums for register names and coefficients ALSA: hda/cs8409: Mask all CS42L42 interrupts on initialization ALSA: hda/cs8409: Reduce HS pops/clicks for Cyborg ALSA: hda/cs8409: Disable unnecessary Ring Sense for Cyborg/Warlock/Bullseye ALSA: hda/cs8409: Disable unsolicited responses during suspend ALSA: hda/cs8409: Mask CS42L42 wake events ALSA: hda/cs8409: Simplify CS42L42 jack detect. ALSA: hda/cs8409: Support multiple sub_codecs for Suspend/Resume/Unsol events ALSA: hda/cs8409: Add Support to disable jack type detection for CS42L42 ALSA: hda/cs8409: Enable Full Scale Volume for Line Out Codec on Dolphin ALSA: hda/cs8409: Set fixed sample rate of 48kHz for CS42L42 ALSA: hda/cs8409: Use timeout rather than retries for I2C transaction waits ALSA: hda/cs8409: Remove unnecessary delays ALSA: hda/cs8409: Follow correct CS42L42 power down sequence for suspend ALSA: hda/cs8409: Unmute/Mute codec when stream starts/stops sound/pci/hda/Kconfig | 11 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_cirrus.c | 1074 ----------------------- sound/pci/hda/patch_cs8409-tables.c | 560 ++++++++++++ sound/pci/hda/patch_cs8409.c | 1234 +++++++++++++++++++++++++++ sound/pci/hda/patch_cs8409.h | 369 ++++++++ 6 files changed, 2176 insertions(+), 1074 deletions(-) create mode 100644 sound/pci/hda/patch_cs8409-tables.c create mode 100644 sound/pci/hda/patch_cs8409.c create mode 100644 sound/pci/hda/patch_cs8409.h -- 2.25.1

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[PATCH] ALSA: hda/realtek: Fix headset mic for Acer SWIFT SF314-56 (ALC256)
by Nikos Liolios 27 Jul '21

27 Jul '21

The issue on Acer SWIFT SF314-56 is that headset microphone doesn't work. The following quirk fixed headset microphone issue. The fixup was found by trial and error. Note that the fixup of SF314-54/55 (ALC256_FIXUP_ACER_HEADSET_MIC) was not successful on my SF314-56. Signed-off-by: Nikos Liolios <liolios.nk(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index caaf0e8aa..14e1ab7c7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8274,6 +8274,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x129c, "Acer SWIFT SF314-55", ALC256_FIXUP_ACER_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1300, "Acer SWIFT SF314-56", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), -- 2.31.1

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Writing a driver for multiple Codecs with one mcAsp
by Kabaz Shlomo 27 Jul '21

27 Jul '21

Hello , I am trying to add to the DM8148 board functionality that support multiple Codecs for One McAsp . The development env is DVRRDK4 of TI . Does any one know how can I do it ? OR what are the concept of adding such support ? Regards , Shlomo Kabaz . The information in this e-mail transmission contains proprietary and business sensitive information. Unauthorized interception of this e-mail may constitute a violation of law. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. You are also asked to contact the sender by reply email and immediately destroy all copies of the original message.

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[PATCH] ASoC: rt1015: Remove unnecessary flush work on rt1015 driver
by Jack Yu 26 Jul '21

26 Jul '21

Remove workqueue of flush work in rt1015 driver since we don't need it after internal discussion, and there is no impact on performance without this workqueue. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 39 +-------------------------------------- sound/soc/codecs/rt1015.h | 2 -- 2 files changed, 1 insertion(+), 40 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 9238f12999aa..b3e69bf245dd 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -547,31 +547,6 @@ static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol, return 0; } -static void rt1015_flush_work(struct work_struct *work) -{ - struct rt1015_priv *rt1015 = container_of(work, struct rt1015_priv, - flush_work.work); - struct snd_soc_component *component = rt1015->component; - unsigned int val, i; - - for (i = 0; i < 200; ++i) { - usleep_range(1000, 1500); - dev_dbg(component->dev, "Flush DAC (retry:%u)\n", i); - regmap_read(rt1015->regmap, RT1015_CLK_DET, &val); - if (val & 0x800) - break; - } - - regmap_write(rt1015->regmap, RT1015_SYS_RST1, 0x0597); - regmap_write(rt1015->regmap, RT1015_SYS_RST1, 0x05f7); - regmap_write(rt1015->regmap, RT1015_MAN_I2C, 0x0028); - - if (val & 0x800) - dev_dbg(component->dev, "Flush DAC completed.\n"); - else - dev_warn(component->dev, "Fail to flush DAC data.\n"); -} - static const struct snd_kcontrol_new rt1015_snd_controls[] = { SOC_SINGLE_TLV("DAC Playback Volume", RT1015_DAC1, RT1015_DAC_VOL_SFT, 127, 0, dac_vol_tlv), @@ -630,10 +605,6 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, } break; - case SND_SOC_DAPM_POST_PMU: - regmap_write(rt1015->regmap, RT1015_MAN_I2C, 0x00a8); - break; - case SND_SOC_DAPM_POST_PMD: if (rt1015->bypass_boost == RT1015_Enable_Boost) { snd_soc_component_write(component, @@ -653,8 +624,6 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, RT1015_SYS_RST2, 0x0b9a); } rt1015->dac_is_used = 0; - - cancel_delayed_work_sync(&rt1015->flush_work); break; default: @@ -687,8 +656,6 @@ static int rt1015_amp_drv_event(struct snd_soc_dapm_widget *w, } break; case SND_SOC_DAPM_POST_PMU: - if (rt1015->hw_config == RT1015_HW_28) - schedule_delayed_work(&rt1015->flush_work, msecs_to_jiffies(10)); msleep(rt1015->pdata.power_up_delay_ms); break; default: @@ -702,7 +669,7 @@ static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = { NULL, 0), SND_SOC_DAPM_AIF_IN("AIFRX", "AIF Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, - r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_OUT_DRV_E("Amp Drv", SND_SOC_NOPM, 0, 0, NULL, 0, rt1015_amp_drv_event, SND_SOC_DAPM_PRE_PMU | @@ -1028,7 +995,6 @@ static int rt1015_probe(struct snd_soc_component *component) snd_soc_component_get_drvdata(component); rt1015->component = component; - INIT_DELAYED_WORK(&rt1015->flush_work, rt1015_flush_work); return 0; } @@ -1037,7 +1003,6 @@ static void rt1015_remove(struct snd_soc_component *component) { struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); - cancel_delayed_work_sync(&rt1015->flush_work); regmap_write(rt1015->regmap, RT1015_RESET, 0); } @@ -1180,8 +1145,6 @@ static int rt1015_i2c_probe(struct i2c_client *i2c, return ret; } - rt1015->hw_config = (i2c->addr == 0x29) ? RT1015_HW_29 : RT1015_HW_28; - ret = regmap_read(rt1015->regmap, RT1015_DEVICE_ID, &val); if (ret) { dev_err(&i2c->dev, diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index 14344532048e..c9f636af7fd1 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -444,8 +444,6 @@ struct rt1015_priv { int bypass_boost; int dac_is_used; int cali_done; - int hw_config; - struct delayed_work flush_work; }; #endif /* __RT1015_H__ */ -- 2.31.1

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[PATCH] ASoC: Intel: sof_da7219_mx98360a: fail to initialize soundcard
by Brent Lu 26 Jul '21

26 Jul '21

The default codec for speaker amp's DAI Link is max98373 and will be overwritten in probe function if the board id is sof_da7219_mx98360a. However, the probe function does not do it because the board id is changed in earlier commit. Fixes: 1cc04d195dc2 ("ASoC: Intel: sof_da7219_max98373: shrink platform_id below 20 characters") Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/sof_da7219_max98373.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index 896251d742fe..b7b3b0bf994a 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -404,7 +404,7 @@ static int audio_probe(struct platform_device *pdev) return -ENOMEM; /* By default dais[0] is configured for max98373 */ - if (!strcmp(pdev->name, "sof_da7219_max98360a")) { + if (!strcmp(pdev->name, "sof_da7219_mx98360a")) { dais[0] = (struct snd_soc_dai_link) { .name = "SSP1-Codec", .id = 0, -- 2.17.1

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[PATCH] ASoC: tlv320aic31xx: Fix jack detection after suspend
by Mark Brown 26 Jul '21

26 Jul '21

The tlv320aic31xx driver relies on regcache_sync() to restore the register contents after going to _BIAS_OFF, for example during system suspend. This does not work for the jack detection configuration since that is configured via the same register that status is read back from so the register is volatile and not cached. This can also cause issues during init if the jack detection ends up getting set up before the CODEC is initially brought out of _BIAS_OFF, we will reset the CODEC and resync the cache as part of that process. Fix this by explicitly reapplying the jack detection configuration after resyncing the register cache during power on. This issue was found by an engineer working off-list on a product kernel, I just wrote up the upstream fix. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index b504d63385b3..e3da8f41fd1a 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1256,6 +1256,13 @@ static int aic31xx_power_on(struct snd_soc_component *component) return ret; } + /* + * The jack detection configuration is in the same register + * that is used to report jack detect status so is volatile + * and not covered by the cache sync, restore it separately. + */ + aic31xx_set_jack(component, aic31xx->jack, NULL); + return 0; } -- 2.20.1

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Writing a driver for multiple Codecs with one mcAsp
by Kabaz Shlomo 26 Jul '21

26 Jul '21

Hello , I am trying to add to the DM8148 board functionality that support multiple Codecs for One McAsp . The development env is DVRRDK4 of TI . Does any one know how can I do it ? OR what are the concept of adding such support ? Regards , Shlomo Kabaz . The information in this e-mail transmission contains proprietary and business sensitive information. Unauthorized interception of this e-mail may constitute a violation of law. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. You are also asked to contact the sender by reply email and immediately destroy all copies of the original message.

1 0

Intel HDA Driver Issue With PCM RW Interleaved
by Ryan McClue 26 Jul '21

26 Jul '21

Hi, I'm trying to play a simple PCM buffer single threaded without buffer underrun. Program source:https://gist.github.com/takaswie/d8a9553c230971d974a3dfc759b2ff67 (struct snd_pcm_sw_params).start_thresholdis altered so as to allow specifying the start of audio frame transmission and prevent underrun issues. In my application, onioctl(fd, SNDRV_PCM_IOCTL_STATUS, &status), I receive:File descriptor in bad state Output of$(dmesg)is:[ 69.013084] snd_hda_intel 0000:00:1f.3: Too big adjustment 128 From a previous response on thealsa-develmailing list: > The driver has some small number of periods to be > put at the beginning of the buffer for correcting the reported > position. Judging from the message, you seem to have set a quite high > rate and a very shot number, and that's possibly too tight to operate > properly. However, I'm not sure what to change in this code to address this issue and get it to work. I tried alteringMY_PLAYBACK_DURATIONandparams.start_thresholdbut get the same error. The program dumps the current hardware parameters: Current hardware parameters: Changed parameters: access format subformat sample-bits frame-bits channels rate period-time period-size period-bytes periods buffer-time buffer-size buffer-bytes tick-time Runtime parameters: access: readwrite-interleaved format: s16-le subformat: std sample-bits: [16, 16], integer, frame-bits: [32, 32], integer, channels: [2, 2], integer, rate: [48000, 48000], integer, period-time: (666, 667), period-size: [32, 32], integer, periods: [32, 32], integer, buffer-time: (21333, 21334), buffer-size: [1024, 1024], integer, buffer-bytes: [4096, 4096], integer, tick-time: [0, 0], integer, info: mmap mmap-valid interleaved block-transfer pause sync-start no-period-wakeup has-wall-clock has-link-atime has-link-synchronized-atime most-significant-bits: 16 rate_num: 48000 rate_den: 1 Thanks

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[PATCH] ALSA: usb-audio: Fix superfluous autosuspend recovery
by Takashi Iwai 26 Jul '21

26 Jul '21

The change to restore the autosuspend from the disabled state uses a wrong check: namely, it should have been the exact comparison of the quirk_type instead of the logical and (&&). Otherwise it matches wrongly with the other quirk types. Although re-enabling the autosuspend for the already enabled device shouldn't matter much, it's better to fix the unbalanced call. Fixes: 9799110825db ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 2f6a62416c05..a1f8c3a026f5 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -907,7 +907,7 @@ static void usb_audio_disconnect(struct usb_interface *intf) } } - if (chip->quirk_type & QUIRK_SETUP_DISABLE_AUTOSUSPEND) + if (chip->quirk_type == QUIRK_SETUP_DISABLE_AUTOSUSPEND) usb_enable_autosuspend(interface_to_usbdev(intf)); chip->num_interfaces--; -- 2.26.2

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sof-soundwire: rt1316 - DAC L and R is replaced by DAC
by GitHub pull_request - opened 26 Jul '21

26 Jul '21

alsa-project/alsa-ucm-conf pull request #109 was opened from jason77-wang: In the latest codec driver rt1316-sdw.c, the DAC L and DAC R are replaced by DAC, so we change the ucm to add the condition checking, if 'DAC L' exists, will set those mixers in the old way, otherwise, set the mixers with new name. The codec rt1316 is used on a Dell TGL soundwire audio machine. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/109 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/109.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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