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[PATCH 01/54] ASoC: dt-bindings: Add WM8978 Binding
by Maxime Ripard 27 Jul '21

27 Jul '21

Even though we had the wm8978 driver for some time and a number of boards using it already, we never had a binding for it. Let's add it based on what the driver expects and the boards are providing. Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- .../devicetree/bindings/sound/wlf,wm8978.yaml | 58 +++++++++++++++++++ 1 file changed, 58 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8978.yaml diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml new file mode 100644 index 000000000000..6761380261e3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8978.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8978 Codec Device Tree Bindings + +maintainers: + - Mark Brown <broonie(a)kernel.org> + +properties: + '#sound-dai-cells': + const: 0 + + compatible: + const: wlf,wm8978 + + reg: + maxItems: 1 + + spi-max-frequency: + maximum: 526000 + +required: + - '#sound-dai-cells' + - compatible + - reg + +additionalProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0>; + spi-max-frequency = <500000>; + }; + }; + + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0>; + }; + }; + +... -- 2.31.1

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Pulseaudio craches on start with SegFault error
by GitHub issues - opened 27 Jul '21

27 Jul '21

alsa-project/alsa-plugins issue #31 was opened from jmtrivial: Dear alsa-project developers, I am using a debian (testing) on my computer, together with deb-multimedia.org repository. As some other users with external USB audio devices, I recently discovered a segfault: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=991337 I discussed withe Christian Marillat (maintainer of the deb-multimedia repository) and we tried a couple of alsa versions, but the bug is still present if I'm using a recent libasound2-plugins such as this one: https://www.deb-multimedia.org/tests/libasound2-plugins_1.2.2+git4219f4b-dm… It does not crash with libasound2-plugins 1.2.2-2 from debian testing. You will found attached the log file from pulseaudio -vvv. [pulseaudio.log](https://github.com/alsa-project/alsa-plugins/files/6884875/… Issue URL : https://github.com/alsa-project/alsa-plugins/issues/31 Repository URL: https://github.com/alsa-project/alsa-plugins

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[PATCH] ALSA: seq: Fix comments of wrong client number for MIDI Passthrough
by Takashi Iwai 27 Jul '21

27 Jul '21

MIDI Passthrough sequencer client is assigned always to the fixed number 14, while it's wrongly documented in the comments as if 62, which was an old number that was used during development. Fix all those numbers. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_dummy.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index ac760b1e3d12..8c18d8c4177e 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -20,15 +20,15 @@ are redirected to output port immediately. The routing can be done via aconnect program in alsa-utils. - Each client has a static client number 62 (= SNDRV_SEQ_CLIENT_DUMMY). + Each client has a static client number 14 (= SNDRV_SEQ_CLIENT_DUMMY). If you want to auto-load this module, you may add the following alias in your /etc/conf.modules file. - alias snd-seq-client-62 snd-seq-dummy + alias snd-seq-client-14 snd-seq-dummy - The module is loaded on demand for client 62, or /proc/asound/seq/ + The module is loaded on demand for client 14, or /proc/asound/seq/ is accessed. If you don't need this module to be loaded, alias - snd-seq-client-62 as "off". This will help modprobe. + snd-seq-client-14 as "off". This will help modprobe. The number of ports to be created can be specified via the module parameter "ports". For example, to create four ports, add the -- 2.26.2

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[PATCH] ALSA: usb-audio: Add registration quirk for JBL Quantum 600
by Alexander Tsoy 27 Jul '21

27 Jul '21

Apparently JBL Quantum 600 has multiple hardware revisions. Apply registration quirk to another device id as well. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e7accd87e063..326d1b0ea5e6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1899,6 +1899,7 @@ static const struct registration_quirk registration_quirks[] = { REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */ REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */ REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */ + REG_QUIRK_ENTRY(0x0ecb, 0x203c, 2), /* JBL Quantum 600 */ REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */ { 0 } /* terminator */ }; -- 2.31.1

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MIDI Through Port Number in ALSA Sequencer
by rv71＠onet.eu 27 Jul '21

27 Jul '21

Hello, ALSA Sequencer always creates MIDI Through Port with the id 14:0. However, I can't find where it is defined that this special port will have this particular id/number. I also don't see any definition in seq.h regarding this ( unlike the timer and announce special port, for example). Can you explain? Thanks Robert

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[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 27 Jul '21

27 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 27 Jul '21

27 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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