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

[PATCH AUTOSEL 5.8 11/53] ALSA: hda: Fix 2 channel swapping for Tegra
by Sasha Levin 07 Sep '20

07 Sep '20

From: Mohan Kumar <mkumard(a)nvidia.com> [ Upstream commit 216116eae43963c662eb84729507bad95214ca6b ] The Tegra HDA codec HW implementation has an issue related to not swapping the 2 channel Audio Sample Packet(ASP) channel mapping. Whatever the FL and FR mapping specified the left channel always comes out of left speaker and right channel on right speaker. So add condition to disallow the swapping of FL,FR during the playback. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> Acked-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/20200825052415.20626-2-mkumard@nvidia.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f0c6d2907e396..9442127d9b15d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3670,6 +3670,7 @@ static int tegra_hdmi_build_pcms(struct hda_codec *codec) static int patch_tegra_hdmi(struct hda_codec *codec) { + struct hdmi_spec *spec; int err; err = patch_generic_hdmi(codec); @@ -3677,6 +3678,10 @@ static int patch_tegra_hdmi(struct hda_codec *codec) return err; codec->patch_ops.build_pcms = tegra_hdmi_build_pcms; + spec = codec->spec; + spec->chmap.ops.chmap_cea_alloc_validate_get_type = + nvhdmi_chmap_cea_alloc_validate_get_type; + spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; return 0; } -- 2.25.1

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[PATCH 00/10] USB: new USB control message helper functions
by Greg Kroah-Hartman 07 Sep '20

07 Sep '20

In a recent discussion about a USB networking bug found by syzbot, and fixed by Himadri Pandya, the design of the existing usb_control_msg() call was brought up as not being the "nicest" thing to use by Dmitry Vyukov: https://lore.kernel.org/r/CACT4Y+YbDODLRFn8M5QcY4CazhpeCaunJnP_udXtAs0rYoAS… The function makes it hard to get right, in that it will return the number of bytes sent/received, but almost no one checks to see if a short read/write happens. With a malicious, or broken, USB device, this can cause drivers to act on data that they did not anticipate, and sometimes copy internal kernel data out to userspace. So let's fix this up by creating two new functions, usb_control_msg_send() and usb_control_msg_recv(). These functions either complete the full transation, or they return an error, a short send/recv is now an error. They also accept data off of the stack, saving individual drivers the pain of having to constantly allocate memory on their own for tiny messages, thereby saving overall kernel code space. The api also does not require a raw USB "pipe" to be sent to the function, as we know the direction, so just pass in the endpoint number instead, again making it easier on the USB driver author to use. This series first takes a helper function out of the sound core for verifying USB endpoints to be able to use internally, and then adds the new functions, converts over some internal USB code to use them, and then starts to clean up some drivers using these new functions, as an example of the savings that can happen by using these functions. Thanks to Dmitry and Himadri for the idea on how to do all of this. greg k-h Greg Kroah-Hartman (10): USB: move snd_usb_pipe_sanity_check into the USB core USB: add usb_control_msg_send() and usb_control_msg_recv() USB: core: message.c: use usb_control_msg_send() in a few places USB: core: hub.c: use usb_control_msg_send() in a few places USB: legousbtower: use usb_control_msg_recv() sound: usx2y: move to use usb_control_msg_send() sound: 6fire: move to use usb_control_msg_send() and usb_control_msg_recv() sound: line6: move to use usb_control_msg_send() and usb_control_msg_recv() sound: hiface: move to use usb_control_msg_send() Bluetooth: ath3k: use usb_control_msg_send() and usb_control_msg_recv() drivers/bluetooth/ath3k.c | 90 +++++------------ drivers/usb/core/hub.c | 128 +++++++++--------------- drivers/usb/core/message.c | 171 ++++++++++++++++++++++++++++---- drivers/usb/core/urb.c | 29 ++++-- drivers/usb/misc/legousbtower.c | 60 ++++------- include/linux/usb.h | 7 ++ sound/usb/6fire/firmware.c | 38 +++---- sound/usb/helper.c | 16 +-- sound/usb/helper.h | 1 - sound/usb/hiface/pcm.c | 14 ++- sound/usb/line6/driver.c | 69 +++++-------- sound/usb/line6/podhd.c | 17 ++-- sound/usb/line6/toneport.c | 8 +- sound/usb/mixer_scarlett_gen2.c | 2 +- sound/usb/quirks.c | 12 +-- sound/usb/usx2y/us122l.c | 42 ++------ 16 files changed, 345 insertions(+), 359 deletions(-) -- 2.28.0

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[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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