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

Line-in is abnormally loud for snd-intel-hda
by GitHub issues - opened 22 Mar '22

22 Mar '22

alsa-project/alsa-lib issue #219 was opened from mpeter50: When I'm on Windows, I sometimes use the line-in Jack to connect my phone to it, so I can either listen to it's sounds along with my PCs sounds or use it as a microphone. I've tried to set this functionality up on my Linux installation, but it seems something is not right. The most basic example is with `arecord`: after setting line-in as the default input device with `alsamixer` I've ran `arecord -vv -fdat foo.wav` so I can check what the ALSA drivers actually "hear" without possible interference of any audio servers. While the command was running, I changed the volume of line-in using he appropriate slider of pavucontrol. When playing the recording, it is clear that while line-in is at 100%, the sound is so loud that it is distorted, and the normal volume is around 12-15% of the actual volume. I don't experience this on Windows, so this is probably not a hardware issue. If I plug a headphone directly into my phone and listen to it, the volume is normal, so this is probably not an issue on the signal producer side either. Checking with `alsamixer`, all capture controls are at 0 volume. But, the controls there might not tell the full truth for multiple reasons: - after reducing the Capture control's volume I can still hear the sound coming from line-in - the Capture control's volume is not in sync with what I see in pavucontrol and other places like KDE's volume menu - the controls Master and PCM in the Playback category does not seem to have any effect. As such, none of the controls control the actual output to my headphone - the "default" card has a single slider, it is named Master and this actually controls the master volume, at least for my headphone Screenshot relevant to the 1st and 2nd points: ![image](https://user-images.githubusercontent.com/83356418/159570560-879f2508-d48c-4718-bf7e-34ee80a4c2e6.png) Additional information: - output of the alsa-info script: [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/8327703/alsa-… - Content of `/proc/asound/card*/codec#*` files: - [/proc/asound/card0/codec#0](https://github.com/alsa-project/alsa-lib/files/8327730/card0_codec0.txt) - [/proc/asound/card1/codec#0](https://github.com/alsa-project/alsa-lib/files/8327735/card1_codec0.txt) - note: card0 is the HDMI section of my graphics card. I never use it for audio, only jack and bluetooth, so it can probably be ignored - output of `lspci -vvnn`: [lspci.txt](https://github.com/alsa-project/alsa-lib/files/8327777/lspci.txt) Issue URL : https://github.com/alsa-project/alsa-lib/issues/219 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v2] ALSA: hda/realtek: Add alc256-samsung-headphone fixup
by Matt Kramer 22 Mar '22

22 Mar '22

Changes since v1: Formatting (forgot to disable word-wrap, sorry). This fixes the near-silence of the headphone jack on the ALC256-based Samsung Galaxy Book Flex Alpha (NP730QCJ). The magic verbs were found through trial and error, using known ALC298 hacks as inspiration. The fixup is auto-enabled only when the NP730QCJ is detected. It can be manually enabled using model=alc256-samsung-headphone. Signed-off-by: Matt Kramer <mccleetus(a)gmail.com> --- Documentation/sound/hd-audio/models.rst | 4 ++++ sound/pci/hda/patch_realtek.c | 11 +++++++++++ 2 files changed, 15 insertions(+) diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index d25335993e55..9b52f50a6854 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -261,6 +261,10 @@ alc-sense-combo huawei-mbx-stereo Enable initialization verbs for Huawei MBX stereo speakers; might be risky, try this at your own risk +alc298-samsung-headphone + Samsung laptops with ALC298 +alc256-samsung-headphone + Samsung laptops with ALC256 ALC66x/67x/892 ============== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1d14be4ee31d..f6ee67f41c45 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6958,6 +6958,7 @@ enum { ALC236_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, + ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, @@ -8286,6 +8287,14 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x08}, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2fcf}, + { } + }, + }, [ALC295_FIXUP_ASUS_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -9099,6 +9108,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), @@ -9445,6 +9455,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, {.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"}, + {.id = ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc256-samsung-headphone"}, {.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"}, {.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"}, {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, -- 2.35.1

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[PATCH BUGFIX] ALSA: pci: fix reading of swapped values from pcmreg in AC97 codec
by Giacomo Guiduzzi 22 Mar '22

22 Mar '22

Tests 72 and 78 for ALSA in kselftest fail due to reading inconsistent values from some devices on a VirtualBox Virtual Machine using the snd_intel8x0 driver for the AC'97 Audio Controller device. Taking for example test number 72, this is what the test reports: "Surround Playback Volume.0 expected 1 but read 0, is_volatile 0" "Surround Playback Volume.1 expected 0 but read 1, is_volatile 0" These errors repeat for each value from 0 to 31. Taking a look at these error messages it is possible to notice that the written values are read back swapped. When the write is performed, these values are initially stored in an array used to sanity-check them and write them in the pcmreg array. To write them, the two one-byte values are packed together in a two-byte variable through bitwise operations: the first value is shifted left by one byte and the second value is stored in the right byte through a bitwise OR. When reading the values back, right shifts are performed to retrieve the previously stored bytes. These shifts are executed in the wrong order, thus reporting the values swapped as shown above. This patch fixes this mistake by reversing the read operations' order. Signed-off-by: Giacomo Guiduzzi <guiduzzi.giacomo(a)gmail.com> Signed-off-by: Paolo Valente <paolo.valente(a)linaro.org> --- sound/pci/ac97/ac97_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 01f296d524ce..cb60a07d39a8 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -938,8 +938,8 @@ static int snd_ac97_ad18xx_pcm_get_volume(struct snd_kcontrol *kcontrol, struct int codec = kcontrol->private_value & 3; mutex_lock(&ac97->page_mutex); - ucontrol->value.integer.value[0] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 0) & 31); - ucontrol->value.integer.value[1] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 8) & 31); + ucontrol->value.integer.value[0] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 8) & 31); + ucontrol->value.integer.value[1] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 0) & 31); mutex_unlock(&ac97->page_mutex); return 0; } -- 2.25.1

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Re: Apogee ONEv2 keeps resetting
by Alan Stern 22 Mar '22

22 Mar '22

On Sun, Mar 20, 2022 at 02:15:40AM +0100, Petr Janecek wrote: > Hi, > I'm trying to use Apogee ONEv2. From usb dumps under > macos or windows it seems like a usb Class Audio 2.0 > device, but in linux, it keeps resetting every two seconds > or so. It keeps resetting even when the snd-usb-audio > driver is disabled, so the problem is probably at a lower > level. Can you post the log output on a system where snd-usb-audio is disabled? > The messages below are from v5.16.16, but it behaves > the same no matter what I plug it into. > > [ 253.708616] usb 3-8: new high-speed USB device number 6 using xhci_hcd > [ 253.899363] usb 3-8: New USB device found, idVendor=0c60, idProduct=0017, > bcdDevice= 1.05 > [ 253.899370] usb 3-8: New USB device strings: Mfr=1, Product=2, > SerialNumber=3 > [ 253.899373] usb 3-8: Product: ONEv2 > [ 253.899375] usb 3-8: Manufacturer: Apogee > [ 253.899377] usb 3-8: SerialNumber: 0C12FF2020204652334D513A7A2A9B > [ 253.960901] mc: Linux media interface: v0.10 Any idea where that line came from? You should try capturing a usbmon trace showing what happens when the device is plugged in and then resets. Preferably on a system where snd-usb-audio is disabled. One other thing you might try: Disable runtime PM for USB ("echo -1 >/sys/module/usbcore/parameters/autosuspend" before plugging in the device). Alan Stern

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[PATCH] ALSA: pcm: Add stream lock during PCM reset ioctl operations
by Takashi Iwai 22 Mar '22

22 Mar '22

snd_pcm_reset() is a non-atomic operation, and it's allowed to run during the PCM stream running. It implies that the manipulation of hw_ptr and other parameters might be racy. This patch adds the PCM stream lock at appropriate places in snd_pcm_*_reset() actions for covering that. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 0e4fbf5fd87b..704fdc9ebf91 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1864,11 +1864,13 @@ static int snd_pcm_do_reset(struct snd_pcm_substream *substream, int err = snd_pcm_ops_ioctl(substream, SNDRV_PCM_IOCTL1_RESET, NULL); if (err < 0) return err; + snd_pcm_stream_lock_irq(substream); runtime->hw_ptr_base = 0; runtime->hw_ptr_interrupt = runtime->status->hw_ptr - runtime->status->hw_ptr % runtime->period_size; runtime->silence_start = runtime->status->hw_ptr; runtime->silence_filled = 0; + snd_pcm_stream_unlock_irq(substream); return 0; } @@ -1876,10 +1878,12 @@ static void snd_pcm_post_reset(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_stream_lock_irq(substream); runtime->control->appl_ptr = runtime->status->hw_ptr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); + snd_pcm_stream_unlock_irq(substream); } static const struct action_ops snd_pcm_action_reset = { -- 2.31.1

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[PATCH] ALSA: hda/realtek: Add alc256-samsung-headphone fixup
by Matt Kramer 22 Mar '22

22 Mar '22

This fixes the near-silence of the headphone jack on the ALC256-based Samsung Galaxy Book Flex Alpha (NP730QCJ). The magic verbs were found through trial and error, using known ALC298 hacks as inspiration. The fixup is auto-enabled only when the NP730QCJ is detected. It can be manually enabled using model=alc256-samsung-headphone. Signed-off-by: Matt Kramer <mccleetus at gmail.com> --- Documentation/sound/hd-audio/models.rst | 4 ++++ sound/pci/hda/patch_realtek.c | 11 +++++++++++ 2 files changed, 15 insertions(+) diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd- audio/models.rst index d25335993e55..9b52f50a6854 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -261,6 +261,10 @@ alc-sense-combo huawei-mbx-stereo Enable initialization verbs for Huawei MBX stereo speakers; might be risky, try this at your own risk +alc298-samsung-headphone + Samsung laptops with ALC298 +alc256-samsung-headphone + Samsung laptops with ALC256 ALC66x/67x/892 ============== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1d14be4ee31d..f6ee67f41c45 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6958,6 +6958,7 @@ enum { ALC236_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, + ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, @@ -8286,6 +8287,14 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x08}, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2fcf}, + { } + }, + }, [ALC295_FIXUP_ASUS_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -9099,6 +9108,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE- X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ- X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), @@ -9445,6 +9455,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256- medion-headset"}, {.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298- samsung-headphone"}, + {.id = ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc256- samsung-headphone"}, {.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi- headset"}, {.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"}, {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, -- 2.35.1

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[v5 0/4] ASoC: mediatek: mt8192: support rt1015p_rt5682s
by Jiaxin Yu 22 Mar '22

22 Mar '22

The series reuses mt8192-mt6359-rt10150rt5682.c for supporting machine driver with rt1015p speaker amplifier and rt5682s headset codec. Changes form v4: - split a large patch into three small patches for easy reviewing - correct coding style Changes from v3: - fix build error: too many arguments for format [-Werror-format-extra-args] Changes from v2: - fix build warnings such as "data argument not used by format string" Changes from v1: - uses the snd_soc_of_get_dai_link_codecs to complete the configuration of dai_link's codecs - uses definitions to simplifies card name and compatible name Jiaxin Yu (4): ASoC: dt-bindings: mt8192-mt6359: add new compatible and new properties ASoC: mediatek: mt8192: refactor for I2S3 DAI link of speaker ASoC: mediatek: mt8192: refactor for I2S8/I2S9 DAI links of headset ASoC: mediatek: mt8192: support rt1015p_rt5682s .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 29 +++ sound/soc/mediatek/Kconfig | 1 + .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 201 +++++++++++------- 3 files changed, 153 insertions(+), 78 deletions(-) -- 2.18.0

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[PATCH v4 00/16] Support external boost at CS35l41 ASoC driver
by Lucas Tanure 22 Mar '22

22 Mar '22

Move the support for CS35L41 external boost to its shared library for ASoC use. This move resulted in cs35l41_hda_reg_sequence being removed, and its steps were broken down into regmap writes or functions from the library. And hardware configuration struct was unified for its use in the shared lib. While at it, some minor bugs were found and fixed it. v4 changelog: - Separated GPIO 1 and 2 function enums v3 changelog: - Remove patches already accepted - Improved logic in documentation patch - Documentation patch goes before its code - Fixed missing Signed-off-by - Fixed subject for HDA patches v2 changelog: - Instead of removing the log, playback actions will log the last regmap access. - Documentation patch with the correct subject line and fixed bug reported by Rob Herring on the provided example. Previous versions: v1: https://lkml.org/lkml/2022/3/3/759 v2: https://lkml.org/lkml/2022/3/4/743 v3: https://lkml.org/lkml/2022/3/8/975 David Rhodes (1): ASoC: dt-bindings: cs35l41: Document CS35l41 External Boost Lucas Tanure (15): sound: cs35l41: Unify hardware configuration sound: cs35l41: Check hw_config before using it sound: cs35l41: Move cs35l41_gpio_config to shared lib ALSA: hda: cs35l41: Fix I2S params comments ALSA: hda: cs35l41: Always configure the DAI ALSA: hda: cs35l41: Add Boost type flag hda: cs35l41: Put the device into safe mode for external boost hda: cs35l41: Mute the device before shutdown sound: cs35l41: Enable Internal Boost in shared lib ALSA: hda: cs35l41: Move boost config to initialization code ALSA: hda: cs35l41: Remove cs35l41_hda_reg_sequence struct ALSA: hda: cs35l41: Reorganize log for playback actions ALSA: hda: cs35l41: Handle all external boost setups the same way ALSA: hda: cs35l41: Move external boost handling to lib for ASoC use ASoC: cs35l41: Support external boost .../bindings/sound/cirrus,cs35l41.yaml | 44 ++- include/sound/cs35l41.h | 59 +++- sound/pci/hda/cs35l41_hda.c | 295 ++++++------------ sound/pci/hda/cs35l41_hda.h | 27 +- sound/soc/codecs/cs35l41-i2c.c | 4 +- sound/soc/codecs/cs35l41-lib.c | 190 ++++++++++- sound/soc/codecs/cs35l41-spi.c | 4 +- sound/soc/codecs/cs35l41.c | 166 +++++----- sound/soc/codecs/cs35l41.h | 5 +- 9 files changed, 443 insertions(+), 351 deletions(-) -- 2.35.1

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pcm: plugin - fix avail_min calculation on rate plugin
by GitHub pull_request - opened 22 Mar '22

22 Mar '22

alsa-project/alsa-lib pull request #218 was opened from aditpape: commit 88e4ae27, ff1f669d introduced a dynamic recalculation of the slave's avail_min value. The calculated avail_min setting did not take into account, that the avail_min value depends on the used sampling rate and must be adapted accordingly if the slave is using a different sampling rate. That leads to too large/too small calculated avail_min settings and inaccurate period wake-up events if a rate converter plugin is used. This patch is adapting the avail_min calculation to consider a different sampling rate between actual pcm and it's slave. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/218 Patch URL : https://github.com/alsa-project/alsa-lib/pull/218.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 1/2] support either TAS2764 or TAS2780 from TI-yaml
by Raphael-Xu 22 Mar '22

22 Mar '22

Signed-off-by: Raphael-Xu <13691752556(a)139.com> --- .../sound/{tas2764.yaml => tas27xx.yaml} | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) rename Documentation/devicetree/bindings/sound/{tas2764.yaml => tas27xx.yaml} (72%) diff --git a/Documentation/devicetree/bindings/sound/tas2764.yaml b/Documentation/devicetree/bindings/sound/tas27xx.yaml similarity index 72% rename from Documentation/devicetree/bindings/sound/tas2764.yaml rename to Documentation/devicetree/bindings/sound/tas27xx.yaml index 5bf8c76ecda1..191f4bceb4ee 100644 --- a/Documentation/devicetree/bindings/sound/tas2764.yaml +++ b/Documentation/devicetree/bindings/sound/tas27xx.yaml @@ -1,25 +1,27 @@ # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) -# Copyright (C) 2020 Texas Instruments Incorporated +# Copyright (C) 2020-2022 Texas Instruments Incorporated %YAML 1.2 --- -$id: "http://devicetree.org/schemas/sound/tas2764.yaml#" +$id: "http://devicetree.org/schemas/sound/tas27xx.yaml#" $schema: "http://devicetree.org/meta-schemas/core.yaml#" -title: Texas Instruments TAS2764 Smart PA +title: Texas Instruments TAS2764/TAS2780 Smart PA maintainers: - - Dan Murphy <dmurphy(a)ti.com> + - Shenghao Ding <shenghao-ding(a)ti.com> description: | - The TAS2764 is a mono, digital input Class-D audio amplifier optimized for - efficiently driving high peak power into small loudspeakers. - Integrated speaker voltage and current sense provides for - real time monitoring of loudspeaker behavior. + The TAS2764/TAS2780 is a mono, digital input Class-D audio amplifier + optimized for efficiently driving high peak power into small + loudspeakers. Integrated speaker voltage and current sense provides + for real time monitoring of loudspeaker behavior. properties: compatible: enum: - ti,tas2764 + or + - ti,tas2780 reg: maxItems: 1 -- 2.35.1

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