September 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: rt5682: Have global name clock option for parent clk
by Akshu Agrawal 07 Sep '20

07 Sep '20

When adding parent clk(mclk) to wclk, this adds fallback option for the case where global clk name is used. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/codecs/rt5682.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 93ebf0279b62..26928bc49bcb 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2780,6 +2780,7 @@ static int rt5682_register_dai_clks(struct snd_soc_component *component) if (rt5682->mclk) { init.parent_data = &(struct clk_parent_data){ .fw_name = "mclk", + .name = __clk_get_name(rt5682->mclk), }; init.num_parents = 1; } -- 2.20.1

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[PATCH] [RFC] ASoC: core: Do not cleanup uninitialized dais on soc_pcm_open failure
by Cezary Rojewski 07 Sep '20

07 Sep '20

Introduce for_each_rtd_dais_rollback macro which behaves exactly like for_each_codec_dais_rollback and its cpu_dais equivalent but for all dais instead. Use newly added macro to fix soc_pcm_open error path and prevent uninitialized dais from being cleaned-up. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-pcm.c | 2 +- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index b94ebff12350..09a7d8409ade 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1193,6 +1193,8 @@ struct snd_soc_pcm_runtime { ((i) < (rtd)->num_cpus + (rtd)->num_codecs) && \ ((dai) = (rtd)->dais[i]); \ (i)++) +#define for_each_rtd_dais_rollback(rtd, i, dai) \ + for (; (--(i) >= 0) && ((dai) = (rtd)->dais[i]);) void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 00ac1cbf6f88..4c9d4cd8cf0b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -812,7 +812,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) return 0; config_err: - for_each_rtd_dais(rtd, i, dai) + for_each_rtd_dais_rollback(rtd, i, dai) snd_soc_dai_shutdown(dai, substream); snd_soc_link_shutdown(substream); -- 2.17.1

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[PATCH 00/11] ALSA: Kill tasklet usage
by Takashi Iwai 07 Sep '20

07 Sep '20

Hi, here is a patch set to convert the tasklet usage in sound tree with either the threaded irq or the dedicated work. It's applied after the tasket API conversion series, found in topic/tasklet-convert branch of sound git tree (which will be included in the next pull request for 5.9-rc4). This contains only non-ASoC changes, the changes for ASoC will follow at next. Takashi === Takashi Iwai (11): ALSA: pcsp: Replace tasklet with work ALSA: timer: Replace tasklet with work ALSA: usb-audio: Replace tasklet with work ALSA: ua101: Replace tasklet with work ALSA: aloop: Replace tasklet with work ALSA: hdsp: Replace tasklet with work ALSA: hdspm: Replace tasklet with work ALSA: riptide: Replace tasklet with threaded irq ALSA: asihpi: Replace tasklet with threaded irq ALSA: firewire: Replace tasklet with work ALSA: mixart: Correct comment wrt obsoleted tasklet usage include/sound/timer.h | 8 +++--- sound/core/hrtimer.c | 2 +- sound/core/timer.c | 20 +++++++------- sound/drivers/aloop.c | 23 ++++++++-------- sound/drivers/pcsp/pcsp_lib.c | 12 ++++---- sound/firewire/amdtp-stream-trace.h | 2 +- sound/firewire/amdtp-stream.c | 25 +++++++++-------- sound/firewire/amdtp-stream.h | 2 +- sound/pci/asihpi/asihpi.c | 28 ++----------------- sound/pci/asihpi/hpioctl.c | 16 +++++++++-- sound/pci/mixart/mixart.h | 2 +- sound/pci/riptide/riptide.c | 20 ++++++++------ sound/pci/rme9652/hdsp.c | 55 ++++++++++++++++++------------------- sound/pci/rme9652/hdspm.c | 13 ++++----- sound/usb/midi.c | 13 +++++---- sound/usb/misc/ua101.c | 16 +++++------ 16 files changed, 122 insertions(+), 135 deletions(-) -- 2.16.4

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[PATCH V2] ALSA: hda: fixup headset for ASUS GX502 laptop
by Luke D Jones 07 Sep '20

07 Sep '20

The GX502 requires a few steps to enable the headset i/o: pincfg, verbs to enable and unmute the amp used for headpone out, and a jacksense callback to toggle output via internal or jack using a verb. Signed-off-by: Luke D Jones <luke(a)ljones.dev> --- sound/pci/hda/patch_realtek.c | 65 +++++++++++++++++++++++++++++++++++ 1 file changed, 65 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 29f5878f0c50..16009297f771 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5975,6 +5975,40 @@ static void alc_fixup_disable_mic_vref(struct hda_codec *codec, snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); } + +static void alc294_gx502_toggle_output(struct hda_codec *codec, + struct hda_jack_callback *cb) +{ + /* The Windows driver sets the codec up in a very different way where + * it appears to leave 0x10 = 0x8a20 set. For Linux we need to toggle it + */ + if (snd_hda_jack_detect_state(codec, 0x21) == HDA_JACK_PRESENT) + alc_write_coef_idx(codec, 0x10, 0x8a20); + else + alc_write_coef_idx(codec, 0x10, 0x0a20); +} + +static void alc294_fixup_gx502_hp(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Pin 0x21: headphones/headset mic */ + if (!is_jack_detectable(codec, 0x21)) + return; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_jack_detect_enable_callback(codec, 0x21, + alc294_gx502_toggle_output); + break; + case HDA_FIXUP_ACT_INIT: + /* Make sure to start in a correct state, i.e. if + * headphones have been plugged in before powering up the system + */ + alc294_gx502_toggle_output(codec, NULL); + break; + } +} + static void alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6154,6 +6188,9 @@ enum { ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, ALC294_FIXUP_ASUS_COEF_1B, + ALC294_FIXUP_ASUS_GX502_HP, + ALC294_FIXUP_ASUS_GX502_PINS, + ALC294_FIXUP_ASUS_GX502_VERBS, ALC285_FIXUP_HP_GPIO_LED, ALC285_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED, @@ -7307,6 +7344,33 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC }, + [ALC294_FIXUP_ASUS_GX502_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, /* front HP mic */ + { 0x1a, 0x01a11830 }, /* rear external mic */ + { 0x21, 0x03211020 }, /* front HP out */ + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_GX502_VERBS + }, + [ALC294_FIXUP_ASUS_GX502_VERBS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* set 0x15 to HP-OUT ctrl */ + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* unmute the 0x15 amp */ + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_GX502_HP + }, + [ALC294_FIXUP_ASUS_GX502_HP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc294_fixup_gx502_hp, + }, [ALC294_FIXUP_ASUS_COEF_1B] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -7588,6 +7652,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), -- 2.26.2

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[PATCH] ALSA: hda: fixup headset for ASUS GX502 laptop
by Luke Jones 07 Sep '20

07 Sep '20

From: Luke D Jones <luke(a)ljones.dev> The GX502 requires a few steps to enable the headset i/o: pincfg, verbs to enable and unmute the amp used for headpone out, and a jacksense callback to toggle output via internal or jack using a verb. Signed-off-by: Luke Jones <luke(a)ljones.dev> --- sound/pci/hda/patch_realtek.c | 70 ++++++++++++++++++++++++++++++++++- 1 file changed, 69 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c521a1f17096..d2052a6e3b13 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5424,7 +5424,7 @@ static void alc_fixup_headset_mode_alc255_no_hp_mic(struct hda_codec *codec, struct alc_spec *spec = codec->spec; spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; alc255_set_default_jack_type(codec); - } + } else alc_fixup_headset_mode(codec, fix, action); } @@ -5993,6 +5993,41 @@ static void alc_fixup_disable_mic_vref(struct hda_codec *codec, snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); } + +static void alc294_gx502_toggle_output(struct hda_codec *codec, + struct hda_jack_callback *cb) +{ + /* The Windows driver sets the codec up in a very different way where + * it appears to leave 0x10 = 0x8a20 set. For Linux we need to toggle it + */ + if (snd_hda_jack_detect_state(codec, 0x21) == HDA_JACK_PRESENT) { + alc_write_coef_idx(codec, 0x10, 0x8a20); + } else { + alc_write_coef_idx(codec, 0x10, 0x0a20); + } +} + +static void alc294_fixup_gx502_hp(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Pin 0x21: headphones/headset mic */ + if (!is_jack_detectable(codec, 0x21)) + return; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_jack_detect_enable_callback(codec, 0x21, + alc294_gx502_toggle_output); + break; + case HDA_FIXUP_ACT_INIT: + /* Make sure to start in a correct state, i.e. if + * headphones have been plugged in before powering up the system + */ + alc294_gx502_toggle_output(codec, NULL); + break; + } +} + static void alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6172,6 +6207,9 @@ enum { ALC285_FIXUP_THINKPAD_X1_GEN7, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, + ALC294_FIXUP_ASUS_GX502_HP, + ALC294_FIXUP_ASUS_GX502_PINS, + ALC294_FIXUP_ASUS_GX502_VERBS, ALC294_FIXUP_ASUS_COEF_1B, ALC285_FIXUP_HP_GPIO_LED, ALC285_FIXUP_HP_MUTE_LED, @@ -7338,6 +7376,35 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC }, + [ALC294_FIXUP_ASUS_GX502_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, /* front HP mic */ + { 0x1a, 0x01a11830 }, //0x00a11030 }, /* rear external mic */ + { 0x21, 0x03211020 }, /* HP out */ + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_GX502_VERBS + }, + [ALC294_FIXUP_ASUS_GX502_VERBS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* set 0x15 to HP-OUT ctrl */ + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* unmute the 0x15 amp */ + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 }, + /* set 0x0a input converter to digital */ + { 0x0a, 0x70d, 0x01 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_GX502_HP + }, + [ALC294_FIXUP_ASUS_GX502_HP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc294_fixup_gx502_hp, + }, [ALC294_FIXUP_ASUS_COEF_1B] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -7711,6 +7778,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), -- 2.26.2

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[PATCH] ALSA: hda: release resource when snd_hdac_device_init is failed
by Kai Vehmanen 07 Sep '20

07 Sep '20

From: Rander Wang <rander.wang(a)intel.com> When snd_hdac_device_init is failed, the codec is released by kfree immediately without releasing some resources. The vendor_name should be free if the memory is allocated and the runtime pm status should be restored, especially the runtime pm status of parent device. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/hda/hdac_device.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..8d9d37225701 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -109,12 +109,16 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, codec->vendor_id & 0xffff); if (!codec->chip_name) { err = -ENOMEM; - goto error; + goto error_chip; } return 0; + error_chip: + kfree(codec->vendor_name); error: + pm_runtime_put_noidle(&codec->dev); + pm_runtime_set_suspended(&codec->dev); put_device(&codec->dev); return err; } -- 2.27.0

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[RFC] soc_pcm_open: error path behavior change since v5.6
by Cezary Rojewski 07 Sep '20

07 Sep '20

Hello, Some time ago negative-tests found out that behavior of soc_pcm_open has changed, quite sure this might be a regression hence my email. Till v5.6 soc_pcm_open was invoking ::shutdown() for cpu_dai in error path only if ::startup() succeeded first (label: 'out'). After addition of commit: 5d9fa03e6c3514266fa94851ab1b6dd6e0595a13 ASoC: soc-pcm: tidyup soc_pcm_open() order this is no longer true. ::shutdown() can and will be invoked for given cpu_dai despite ::startup()'s failure. This complicated further since the merging of cpu & codec dais. The same applies to codec_dais: notice the usage of for_each_rtd_codec_dai_rollback macro instead of for_each_rtd_dais in error path in v5.6. Should dai's ::shutdown() be introducing some kind of state-check from now on? - similarly to how developers deal with some of the core pcm operations e.g.: ::prepare() (as it may get invoked multiple times in a row so check is there to prevent redundancy). Or, perhaps behavior change should be reverted with ::shutdown() routine again being called only after successful ::startup()? Czarek

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[PATCH v2 00/20] Add Allwinner H3/H5/H6/A64 HDMI audio
by Clément Péron 05 Sep '20

05 Sep '20

Hi, Regarding the I2S LRCK polarity, Maxime Ripard test it and found that the LRCK is fine: https://lkml.org/lkml/2020/7/29/581. So the patch introduce this modification has been reverted. I have tested this on Allwinner H6 but this should be tested on older platform like sun4i family. Regards, Clement Change since v1: - rebase on next-20200828 - add revert LRCK polarity - remove all simple-audio-card,frame-inversion in dts - add Ondrej patches for Orange Pi board - Add arm64 defconfig patch Clément Péron (4): Revert "ASoC: sun4i-i2s: Fix the LRCK polarity" ASoC: sun4i-i2s: Fix sun8i volatile regs arm64: dts: allwinner: h6: Enable HDMI sound for Beelink GS1 arm64: defconfig: Enable Allwinner i2s driver Jernej Skrabec (3): ASoC: sun4i-i2s: Add support for H6 I2S dt-bindings: ASoC: sun4i-i2s: Add H6 compatible arm64: dts: allwinner: h6: Add HDMI audio node Marcus Cooper (10): ASoC: sun4i-i2s: Adjust LRCLK width ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support arm: dts: sunxi: h3/h5: Add DAI node for HDMI arm: dts: sunxi: h3/h5: Add HDMI audio arm64: dts: allwinner: a64: Add DAI node for HDMI arm64: dts: allwinner: a64: Add HDMI audio arm: sun8i: h3: Add HDMI audio to Orange Pi 2 arm: sun8i: h3: Add HDMI audio to Beelink X2 arm64: dts: allwinner: a64: Add HDMI audio to Pine64 Ondrej Jirman (3): arm64: dts: allwinner: Enable HDMI audio on Orange Pi PC 2 ARM: dts: sun8i-h3: Enable HDMI audio on Orange Pi PC/One arm64: dts: sun50i-h6-orangepi-3: Enable HDMI audio .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm/boot/dts/sun8i-h3-beelink-x2.dts | 8 + arch/arm/boot/dts/sun8i-h3-orangepi-2.dts | 8 + arch/arm/boot/dts/sun8i-h3-orangepi-one.dts | 8 + arch/arm/boot/dts/sun8i-h3-orangepi-pc.dts | 8 + arch/arm/boot/dts/sunxi-h3-h5.dtsi | 32 ++ .../boot/dts/allwinner/sun50i-a64-pine64.dts | 8 + arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 33 ++ .../dts/allwinner/sun50i-h5-orangepi-pc2.dts | 8 + .../dts/allwinner/sun50i-h6-beelink-gs1.dts | 8 + .../dts/allwinner/sun50i-h6-orangepi-3.dts | 8 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 32 ++ arch/arm64/configs/defconfig | 1 + sound/soc/sunxi/sun4i-i2s.c | 285 ++++++++++++++++-- 14 files changed, 431 insertions(+), 18 deletions(-) -- 2.25.1

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[PATCH 00/16] Add Allwinner H3/H5/H6/A64 HDMI audio
by Clément Péron 04 Sep '20

04 Sep '20

Hi, This a merge of serie: - Add Add H6 I2S support https://patchwork.kernel.org/cover/11497007/ - Add Allwinner H3/H5/A64 HDMI audio https://patchwork.kernel.org/cover/11510511/ I merge both serie because there is a similar issue regarding the I2S polarity. This need to be investigated under a scope and see if the comment is still true for Allwinner H6 and previous SoC. LibreElec team found that we have to introduce the "simple-audio-card,frame-inversion" property with recent mainline change. In order to make HDMI audio work. Maybe the I2S polarity is good or maybe the silicium has an issue but not present for HDMI I2S or not present in TDM mode... I cannot do it myself, so if someone want to to do it please feel free. Regarding the discussion we had here: https://patchwork.kernel.org/patch/11510521/ I didn't switch to generic hdmi card-name and used name like: sun8i-h3-hdmi, sun50i-h6-hdmi, etc.... Despite this wrong comment the rest of the serie introduce some fix that should be merged even if it's without the H6 support. Regards Clement Change since v1: - drop allwinner,playback-channels property - use coherent hdmi,card-name - indentation fix - collect tags Clément Péron (2): ASoC: sun4i-i2s: Fix sun8i volatile regs arm64: dts: allwinner: h6: Enable HDMI sound for Beelink GS1 Jernej Skrabec (3): ASoC: sun4i-i2s: Add support for H6 I2S dt-bindings: ASoC: sun4i-i2s: Add H6 compatible arm64: dts: allwinner: h6: Add HDMI audio node Marcus Cooper (11): ASoC: sun4i-i2s: Adjust LRCLK width ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support ASoC: sun4i-i2s: Adjust regmap settings arm: dts: sunxi: h3/h5: Add DAI node for HDMI arm: dts: sunxi: h3/h5: Add HDMI audio arm64: dts: allwinner: a64: Add DAI node for HDMI arm64: dts: allwinner: a64: Add HDMI audio arm: sun8i: h3: Add HDMI audio to Orange Pi 2 arm: sun8i: h3: Add HDMI audio to Beelink X2 arm64: dts: allwinner: a64: Add HDMI audio to Pine64 .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm/boot/dts/sun8i-h3-beelink-x2.dts | 8 + arch/arm/boot/dts/sun8i-h3-orangepi-2.dts | 8 + arch/arm/boot/dts/sunxi-h3-h5.dtsi | 33 ++ .../boot/dts/allwinner/sun50i-a64-pine64.dts | 8 + arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 35 +++ .../dts/allwinner/sun50i-h6-beelink-gs1.dts | 8 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 33 ++ sound/soc/sunxi/sun4i-i2s.c | 281 +++++++++++++++++- 9 files changed, 410 insertions(+), 6 deletions(-) -- 2.25.1

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[PATCH v3 0/5] Qualcomm's lpass-hdmi ASoC driver to support audio over dp port
by Srinivasa Rao Mandadapu 04 Sep '20

04 Sep '20

These patches are to support audio over DP port on Qualcomm's SC7180 LPASS Asoc. It includes machine driver, cpu driver, platform driver updates for HDMI path support, device tree documention, lpass variant structure optimization and configuration changes. These patches depends on the DP patch series https://patchwork.kernel.org/project/dri-devel/list/?series=332029 Changes Since v2: -- Audio buffer size(i.e. LPASS_PLATFORM_BUFFER_SIZE) in lpass-platform.c increased. V Sujith Kumar Reddy (5): ASoC: Add sc7180-lpass binding header hdmi define ASoC: dt-bindings: Add dt binding for lpass hdmi ASoC: qcom: Add support for lpass hdmi driver ASoC: qcom: Add support for audio over DP ASoC: qcom: Optimise lpass variant structure .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 51 +- include/dt-bindings/sound/sc7180-lpass.h | 1 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 25 +- sound/soc/qcom/lpass-cpu.c | 92 ++- sound/soc/qcom/lpass-hdmi.c | 685 +++++++++++++++++++++ sound/soc/qcom/lpass-hdmi.h | 129 ++++ sound/soc/qcom/lpass-ipq806x.c | 25 +- sound/soc/qcom/lpass-lpaif-reg.h | 51 +- sound/soc/qcom/lpass-platform.c | 287 +++++++-- sound/soc/qcom/lpass-sc7180.c | 147 ++++- sound/soc/qcom/lpass.h | 123 +++- 13 files changed, 1472 insertions(+), 151 deletions(-) create mode 100644 sound/soc/qcom/lpass-hdmi.c create mode 100644 sound/soc/qcom/lpass-hdmi.h -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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