May 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH alsa-lib 0/4] pcm: hw: implement explicit silencing for snd_pcm_drain
by Jaroslav Kysela 05 May '23

05 May '23

This is a complete API solution for the filling of the silence samples for the drain operation. The default bahaviour is to aligns the valid samples to one period with extra silence samples for next 1/10th of second. Introduce SNDRV_PCM_INFO_PERFECT_DRAIN and SNDRV_PCM_HW_PARAMS_NO_DRAIN_SILENCE flags to fully control the filling of the silence samples in the snd_pcm_drain call. Those flags do the bidirectional setup for this operation, so the silencing may be implemented eventually in the kernel space when there's a demand in future. Also, some applications may not require to silence the samples beyond the tail of the ring buffer at all. There is also a new pcm_hw plugin configuration field allowing the overwrite of the default (auto) behaviour. Related: https://lore.kernel.org/alsa-devel/20230420113324.877164-2-oswald.buddenhag… Related: https://lore.kernel.org/alsa-devel/20230405201219.2197789-2-oswald.buddenha… Jaroslav Kysela (4): pcm: hw: setup explicit silencing for snd_pcm_drain by default pcm: hw: add drain_silence configuration keyword pcm: hw: introduce SNDRV_PCM_INFO_PERFECT_DRAIN pcm: hw: introduce SNDRV_PCM_HW_PARAMS_DRAIN_SILENCE include/pcm.h | 1 + include/sound/uapi/asound.h | 4 ++ src/pcm/pcm.c | 88 +++++++++++++++++++++++++++++++------ src/pcm/pcm_hw.c | 74 ++++++++++++++++++++++++++++++- src/pcm/pcm_local.h | 7 +++ 5 files changed, 160 insertions(+), 14 deletions(-) -- 2.39.2

3 16

Added double quotes to variables of three `rm` commands
by GitHub pull_request - opened 05 May '23

05 May '23

alsa-project/alsa-utils pull request #210 was opened from wileyhy: lines 73, 80 and 98 Request URL : https://github.com/alsa-project/alsa-utils/pull/210 Patch URL : https://github.com/alsa-project/alsa-utils/pull/210.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 1/7] ALSA: pcm: Revert "ALSA: pcm: rewrite snd_pcm_playback_silence()"
by Oswald Buddenhagen 05 May '23

05 May '23

From: Jaroslav Kysela <perex(a)perex.cz> This reverts commit 9f656705c5faa18afb26d922cfc64f9fd103c38d. There was a regression (in the top-up mode). Unfortunately, the patch provided from the author of this commit is not easy to review. Keep the updated and new comments in headers. Also add a new comment that documents the missed API constraint which led to the regression. Reported-by: Jeff Chua <jeff.chua.linux(a)gmail.com> Link: https://lore.kernel.org/r/CAAJw_ZsbTVd3Es373x_wTNDF7RknGhCD0r+NKUSwAO7HpLAk… Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- v2: - add docu comment - commit message improvements --- sound/core/pcm_lib.c | 90 ++++++++++++++++++++++++----------------- sound/core/pcm_local.h | 3 +- sound/core/pcm_native.c | 6 +-- 3 files changed, 59 insertions(+), 40 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index d21c73944efd..3357ffae635f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -42,56 +42,74 @@ static int fill_silence_frames(struct snd_pcm_substream *substream, * * when runtime->silence_size >= runtime->boundary - fill processed area with silence immediately */ -void snd_pcm_playback_silence(struct snd_pcm_substream *substream) +void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_uframes_t new_hw_ptr) { struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_uframes_t appl_ptr = READ_ONCE(runtime->control->appl_ptr); - snd_pcm_sframes_t added, hw_avail, frames; - snd_pcm_uframes_t noise_dist, ofs, transfer; + snd_pcm_uframes_t frames, ofs, transfer; int err; - added = appl_ptr - runtime->silence_start; - if (added) { - if (added < 0) - added += runtime->boundary; - if (added < runtime->silence_filled) - runtime->silence_filled -= added; - else - runtime->silence_filled = 0; - runtime->silence_start = appl_ptr; - } - - // This will "legitimately" turn negative on underrun, and will be mangled - // into a huge number by the boundary crossing handling. The initial state - // might also be not quite sane. The code below MUST account for these cases. - hw_avail = appl_ptr - runtime->status->hw_ptr; - if (hw_avail < 0) - hw_avail += runtime->boundary; - - noise_dist = hw_avail + runtime->silence_filled; if (runtime->silence_size < runtime->boundary) { - frames = runtime->silence_threshold - noise_dist; - if (frames <= 0) + snd_pcm_sframes_t noise_dist, n; + snd_pcm_uframes_t appl_ptr = READ_ONCE(runtime->control->appl_ptr); + if (runtime->silence_start != appl_ptr) { + n = appl_ptr - runtime->silence_start; + if (n < 0) + n += runtime->boundary; + if ((snd_pcm_uframes_t)n < runtime->silence_filled) + runtime->silence_filled -= n; + else + runtime->silence_filled = 0; + runtime->silence_start = appl_ptr; + } + if (runtime->silence_filled >= runtime->buffer_size) return; + noise_dist = snd_pcm_playback_hw_avail(runtime) + runtime->silence_filled; + if (noise_dist >= (snd_pcm_sframes_t) runtime->silence_threshold) + return; + frames = runtime->silence_threshold - noise_dist; if (frames > runtime->silence_size) frames = runtime->silence_size; } else { - frames = runtime->buffer_size - noise_dist; - if (frames <= 0) - return; + /* + * This filling mode aims at free-running mode (used for example by dmix), + * which doesn't update the application pointer. + */ + if (new_hw_ptr == ULONG_MAX) { /* initialization */ + snd_pcm_sframes_t avail = snd_pcm_playback_hw_avail(runtime); + if (avail > runtime->buffer_size) + avail = runtime->buffer_size; + runtime->silence_filled = avail > 0 ? avail : 0; + runtime->silence_start = (runtime->status->hw_ptr + + runtime->silence_filled) % + runtime->boundary; + } else { + ofs = runtime->status->hw_ptr; + frames = new_hw_ptr - ofs; + if ((snd_pcm_sframes_t)frames < 0) + frames += runtime->boundary; + runtime->silence_filled -= frames; + if ((snd_pcm_sframes_t)runtime->silence_filled < 0) { + runtime->silence_filled = 0; + runtime->silence_start = new_hw_ptr; + } else { + runtime->silence_start = ofs; + } + } + frames = runtime->buffer_size - runtime->silence_filled; } - if (snd_BUG_ON(frames > runtime->buffer_size)) return; - ofs = (runtime->silence_start + runtime->silence_filled) % runtime->buffer_size; - do { + if (frames == 0) + return; + ofs = runtime->silence_start % runtime->buffer_size; + while (frames > 0) { transfer = ofs + frames > runtime->buffer_size ? runtime->buffer_size - ofs : frames; err = fill_silence_frames(substream, ofs, transfer); snd_BUG_ON(err < 0); runtime->silence_filled += transfer; frames -= transfer; ofs = 0; - } while (frames > 0); + } snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_DEVICE); } @@ -425,6 +443,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, return 0; } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + runtime->silence_size > 0) + snd_pcm_playback_silence(substream, new_hw_ptr); + if (in_interrupt) { delta = new_hw_ptr - runtime->hw_ptr_interrupt; if (delta < 0) @@ -442,10 +464,6 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, runtime->hw_ptr_wrap += runtime->boundary; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - runtime->silence_size > 0) - snd_pcm_playback_silence(substream); - update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp); return snd_pcm_update_state(substream, runtime); diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index 42fe3a4e9154..ecb21697ae3a 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -29,7 +29,8 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime); int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream); -void snd_pcm_playback_silence(struct snd_pcm_substream *substream); +void snd_pcm_playback_silence(struct snd_pcm_substream *substream, + snd_pcm_uframes_t new_hw_ptr); static inline snd_pcm_uframes_t snd_pcm_avail(struct snd_pcm_substream *substream) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 3d0c4a5b701b..94185267a7b9 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -958,7 +958,7 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (snd_pcm_running(substream)) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) - snd_pcm_playback_silence(substream); + snd_pcm_playback_silence(substream, ULONG_MAX); err = snd_pcm_update_state(substream, runtime); } snd_pcm_stream_unlock_irq(substream); @@ -1455,7 +1455,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, __snd_pcm_set_state(runtime, state); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) - snd_pcm_playback_silence(substream); + snd_pcm_playback_silence(substream, ULONG_MAX); snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART); } @@ -1916,7 +1916,7 @@ static void snd_pcm_post_reset(struct snd_pcm_substream *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); + snd_pcm_playback_silence(substream, ULONG_MAX); snd_pcm_stream_unlock_irq(substream); } -- 2.40.0.152.g15d061e6df

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[PATCH v2 1/7] ALSA: pcm: Revert "ALSA: pcm: rewrite snd_pcm_playback_silence()"
by Oswald Buddenhagen 05 May '23

05 May '23

From: Jaroslav Kysela <perex(a)perex.cz> This reverts commit 9f656705c5faa18afb26d922cfc64f9fd103c38d. There was a regression (in the top-up mode). Unfortunately, the patch provided from the author of this commit is not easy to review. Keep the updated and new comments in headers. Also add a new comment that documents the missed API constraint which led to the regression. Reported-by: Jeff Chua <jeff.chua.linux(a)gmail.com> Link: https://lore.kernel.org/r/CAAJw_ZsbTVd3Es373x_wTNDF7RknGhCD0r+NKUSwAO7HpLAk… Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- v2: - add docu comment - commit message improvements --- sound/core/pcm_lib.c | 90 ++++++++++++++++++++++++----------------- sound/core/pcm_local.h | 3 +- sound/core/pcm_native.c | 6 +-- 3 files changed, 59 insertions(+), 40 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index d21c73944efd..3357ffae635f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -42,56 +42,74 @@ static int fill_silence_frames(struct snd_pcm_substream *substream, * * when runtime->silence_size >= runtime->boundary - fill processed area with silence immediately */ -void snd_pcm_playback_silence(struct snd_pcm_substream *substream) +void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_uframes_t new_hw_ptr) { struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_uframes_t appl_ptr = READ_ONCE(runtime->control->appl_ptr); - snd_pcm_sframes_t added, hw_avail, frames; - snd_pcm_uframes_t noise_dist, ofs, transfer; + snd_pcm_uframes_t frames, ofs, transfer; int err; - added = appl_ptr - runtime->silence_start; - if (added) { - if (added < 0) - added += runtime->boundary; - if (added < runtime->silence_filled) - runtime->silence_filled -= added; - else - runtime->silence_filled = 0; - runtime->silence_start = appl_ptr; - } - - // This will "legitimately" turn negative on underrun, and will be mangled - // into a huge number by the boundary crossing handling. The initial state - // might also be not quite sane. The code below MUST account for these cases. - hw_avail = appl_ptr - runtime->status->hw_ptr; - if (hw_avail < 0) - hw_avail += runtime->boundary; - - noise_dist = hw_avail + runtime->silence_filled; if (runtime->silence_size < runtime->boundary) { - frames = runtime->silence_threshold - noise_dist; - if (frames <= 0) + snd_pcm_sframes_t noise_dist, n; + snd_pcm_uframes_t appl_ptr = READ_ONCE(runtime->control->appl_ptr); + if (runtime->silence_start != appl_ptr) { + n = appl_ptr - runtime->silence_start; + if (n < 0) + n += runtime->boundary; + if ((snd_pcm_uframes_t)n < runtime->silence_filled) + runtime->silence_filled -= n; + else + runtime->silence_filled = 0; + runtime->silence_start = appl_ptr; + } + if (runtime->silence_filled >= runtime->buffer_size) return; + noise_dist = snd_pcm_playback_hw_avail(runtime) + runtime->silence_filled; + if (noise_dist >= (snd_pcm_sframes_t) runtime->silence_threshold) + return; + frames = runtime->silence_threshold - noise_dist; if (frames > runtime->silence_size) frames = runtime->silence_size; } else { - frames = runtime->buffer_size - noise_dist; - if (frames <= 0) - return; + /* + * This filling mode aims at free-running mode (used for example by dmix), + * which doesn't update the application pointer. + */ + if (new_hw_ptr == ULONG_MAX) { /* initialization */ + snd_pcm_sframes_t avail = snd_pcm_playback_hw_avail(runtime); + if (avail > runtime->buffer_size) + avail = runtime->buffer_size; + runtime->silence_filled = avail > 0 ? avail : 0; + runtime->silence_start = (runtime->status->hw_ptr + + runtime->silence_filled) % + runtime->boundary; + } else { + ofs = runtime->status->hw_ptr; + frames = new_hw_ptr - ofs; + if ((snd_pcm_sframes_t)frames < 0) + frames += runtime->boundary; + runtime->silence_filled -= frames; + if ((snd_pcm_sframes_t)runtime->silence_filled < 0) { + runtime->silence_filled = 0; + runtime->silence_start = new_hw_ptr; + } else { + runtime->silence_start = ofs; + } + } + frames = runtime->buffer_size - runtime->silence_filled; } - if (snd_BUG_ON(frames > runtime->buffer_size)) return; - ofs = (runtime->silence_start + runtime->silence_filled) % runtime->buffer_size; - do { + if (frames == 0) + return; + ofs = runtime->silence_start % runtime->buffer_size; + while (frames > 0) { transfer = ofs + frames > runtime->buffer_size ? runtime->buffer_size - ofs : frames; err = fill_silence_frames(substream, ofs, transfer); snd_BUG_ON(err < 0); runtime->silence_filled += transfer; frames -= transfer; ofs = 0; - } while (frames > 0); + } snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_DEVICE); } @@ -425,6 +443,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, return 0; } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + runtime->silence_size > 0) + snd_pcm_playback_silence(substream, new_hw_ptr); + if (in_interrupt) { delta = new_hw_ptr - runtime->hw_ptr_interrupt; if (delta < 0) @@ -442,10 +464,6 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, runtime->hw_ptr_wrap += runtime->boundary; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - runtime->silence_size > 0) - snd_pcm_playback_silence(substream); - update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp); return snd_pcm_update_state(substream, runtime); diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index 42fe3a4e9154..ecb21697ae3a 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -29,7 +29,8 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime); int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream); -void snd_pcm_playback_silence(struct snd_pcm_substream *substream); +void snd_pcm_playback_silence(struct snd_pcm_substream *substream, + snd_pcm_uframes_t new_hw_ptr); static inline snd_pcm_uframes_t snd_pcm_avail(struct snd_pcm_substream *substream) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 3d0c4a5b701b..94185267a7b9 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -958,7 +958,7 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (snd_pcm_running(substream)) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) - snd_pcm_playback_silence(substream); + snd_pcm_playback_silence(substream, ULONG_MAX); err = snd_pcm_update_state(substream, runtime); } snd_pcm_stream_unlock_irq(substream); @@ -1455,7 +1455,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, __snd_pcm_set_state(runtime, state); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) - snd_pcm_playback_silence(substream); + snd_pcm_playback_silence(substream, ULONG_MAX); snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART); } @@ -1916,7 +1916,7 @@ static void snd_pcm_post_reset(struct snd_pcm_substream *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); + snd_pcm_playback_silence(substream, ULONG_MAX); snd_pcm_stream_unlock_irq(substream); } -- 2.40.0.152.g15d061e6df

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[PATCH] ALSA: hda/realtek: Fix mute and micmute LEDs for an HP laptop
by Kai-Heng Feng 05 May '23

05 May '23

There's another laptop that needs the fixup to enable mute and micmute LEDs. So do it accordingly. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.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 f70d6a33421d..afc562a8afbb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9478,6 +9478,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b8d, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8f, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8b96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- 2.34.1

2 1

[GIT PULL] ASoC fixes for v6.4-rc1
by Mark Brown 05 May '23

05 May '23

The following changes since commit dc801ea8ae37d54706e6f1cef140731ac5981c9c: ASoC: fsl: imx-audmix: remove dummy dai_link->platform (2023-04-21 13:45:07 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.4-rc1 for you to fetch changes up to ec6f82b4c63cc68f8dc03316e725106d242706be: ASoC: Intel: soc-acpi-cht: Add quirk for Nextbook Ares 8A tablet (2023-05-04 10:14:31 +0900) ---------------------------------------------------------------- ASoC: Fixes for v6.4 A small set of fixes and device quirks that have come in during the merge window, the Qualcomm fix seems quite large but it's mainly code motion so looks larger than it is. ---------------------------------------------------------------- Andrey Rakhmatullin (1): ASoC: amd: yc: Add Asus VivoBook Pro 14 OLED M6400RC to the quirks list for acp6x Enrico Belleri (1): ASoC: amd: yc: Add ASUS M3402RA into DMI table Hans de Goede (2): ASoC: Intel: soc-acpi-byt: Fix "WM510205" match no longer working ASoC: Intel: soc-acpi-cht: Add quirk for Nextbook Ares 8A tablet Krzysztof Kozlowski (1): ASoC: codecs: wcd938x: fix accessing regmap on unattached devices Maxim Kochetkov (1): ASoC: max98090: make it selectable sound/soc/amd/yc/acp6x-mach.c | 14 + sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wcd938x-sdw.c | 1037 ++++++++++++++++++++- sound/soc/codecs/wcd938x.c | 1003 +------------------- sound/soc/codecs/wcd938x.h | 1 + sound/soc/intel/common/soc-acpi-intel-byt-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 26 + 7 files changed, 1072 insertions(+), 1013 deletions(-)

2 1

[PATCH 0/5] rewrite snd_pcm_playback_silence() again
by Jaroslav Kysela 05 May '23

05 May '23

This is a split of changes for both patches (the first with the missing top-up mode + later fix with mixups) proposed by Oswald. The aim was to pick only real code changes. Only the first two patches after revert fixes the current silencing issues. The last two are just cleanups with the extra optimization in the last patch moving the common code to a function. Cc: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> Cc: Jeff Chua <jeff.chua.linux(a)gmail.com> Jaroslav Kysela (5): ALSA: pcm: Revert "ALSA: pcm: rewrite snd_pcm_playback_silence()" ALSA: pcm: fix playback silence - use the actual new_hw_ptr for the threshold mode ALSA: pcm: fix playback silence - correct the incremental silencing ALSA: pcm: playback silence - remove extra code ALSA: pcm: playback silence - move silence variables updates to separate function sound/core/pcm_lib.c | 91 +++++++++++++++++++++++++---------------- sound/core/pcm_local.h | 3 +- sound/core/pcm_native.c | 6 +-- 3 files changed, 61 insertions(+), 39 deletions(-) -- 2.39.2

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ucm2: USB-Audio: Add support for ALC4080 Gigabyte Z590 AORUS ULTRA
by GitHub pull_request - edited 05 May '23

05 May '23

alsa-project/alsa-ucm-conf pull request #310 was edited from BearsPunch: When installed Manjaro I had an issue my audio card was not recognized. USB ID is 0414:a00d and as I see you already have it under if.realtek-alc1220-vb condition, but I have ALC4080. So I manually changed USB-Audio.conf under If.realtek-alc4080 condition from (0414:a0(0e|1[0124])) to (0414:a0(0[de]|1[0124])) and now it works perfectly. But Ubuntu (maybe from 22.04) doesn't have this issue. Does Canonical update alsa-ucm-conf by themselves without pull request to main branch? Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/310 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/310.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v3 0/7] ASoC: mediatek: mt8188: revise AFE driver
by Trevor Wu 05 May '23

05 May '23

The series of patches consists of three major changes. First, remove redundant supply for ADDA DAI dirver. Second, revise ETDM control including APLL dynamic switch via DAPM, so APLL can be enabled when it is really required. Finally, AFE probe function is updated. Bus protection change was dropped at the previous patch because the dependent change was not accepted. Changes since v2: - remove required properties in dt-binding file Changes since v1: - replace unnecessary dev_info with dev_dbg - fix dt-binging ABI breaking behavior Trevor Wu (7): ASoC: mediatek: mt8188: remove supply AUDIO_HIRES ASoC: mediatek: mt8188: complete set_tdm_slot function ASoC: mediatek: mt8188: revise ETDM control flow ASoC: mediatek: mt8188: refine APLL control ASoC: mediatek: mt8188: combine afe component registration ASoC: mediatek: mt8188: add bus protection ASoC: dt-bindings: mediatek,mt8188-afe: add audio properties .../bindings/sound/mediatek,mt8188-afe.yaml | 12 +- sound/soc/mediatek/mt8188/mt8188-afe-clk.c | 138 ++- sound/soc/mediatek/mt8188/mt8188-afe-clk.h | 11 + sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 113 ++- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 37 - sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 922 ++++++++++-------- sound/soc/mediatek/mt8188/mt8188-reg.h | 2 + 7 files changed, 751 insertions(+), 484 deletions(-) -- 2.18.0

3 9

[PATCH] ASoC: da7219: Add Jack insertion detection polarity selection
by David Rau 05 May '23

05 May '23

Add the selection of DA7219 Jack insertion detection polarity - Default polarity (Low) - Inverted polarity (High) Signed-off-by: David Rau <David.Rau.opensource(a)dm.renesas.com> --- sound/soc/codecs/Kconfig | 24 +++++++++++++++++++++++- sound/soc/codecs/da7219-aad.c | 16 ++++++++++++++++ 2 files changed, 39 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 79d2362ad055..96f934ad52e7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -967,9 +967,31 @@ config SND_SOC_DA7218 depends on I2C config SND_SOC_DA7219 - tristate + tristate "Dialog DA7219 CODEC" depends on I2C +if SND_SOC_DA7219 + +choice + bool "DA7219 Jack insertion detection polarity selection" + default DA7219_DEFAULT_JACK_INSERTION_POLARITY if (SND_SOC_DA7219) + +config DA7219_DEFAULT_JACK_INSERTION_POLARITY + bool "Default polarity" + depends on SND_SOC_DA7219 + help + Select this option if your DA7219 codec with default Jack insertion detection polarity (Low). + +config DA7219_INVERTED_JACK_INSERTION_POLARITY + bool "Inverted polarity" + depends on SND_SOC_DA7219 + help + Select this option if your DA7219 codec with inverted Jack insertion detection polarity (High). + +endchoice + +endif # SND_SOC_DA7219 + config SND_SOC_DA732X tristate depends on I2C diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 993a0d00bc48..9d16112cff6b 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -879,6 +879,18 @@ static void da7219_aad_handle_gnd_switch_time(struct snd_soc_component *componen } } +static void da7219_aad_handle_polarity(struct snd_soc_component *component) +{ + snd_soc_component_write(component, 0xF0, 0x8B); + + if (IS_ENABLED(CONFIG_DA7219_DEFAULT_JACK_INSERTION_POLARITY)) + snd_soc_component_write(component, 0x75, 0x80); + else if (IS_ENABLED(CONFIG_DA7219_INVERTED_JACK_INSERTION_POLARITY)) + snd_soc_component_write(component, 0x75, 0x00); + + snd_soc_component_write(component, 0xF0, 0x00); +} + /* * Suspend/Resume */ @@ -955,8 +967,12 @@ int da7219_aad_init(struct snd_soc_component *component) snd_soc_component_update_bits(component, DA7219_ACCDET_CONFIG_1, DA7219_BUTTON_CONFIG_MASK, 0); + /* Handle the default ground switch delay time */ da7219_aad_handle_gnd_switch_time(component); + /* Handle the Jack insertion detection polarity */ + da7219_aad_handle_polarity(component); + da7219_aad->aad_wq = create_singlethread_workqueue("da7219-aad"); if (!da7219_aad->aad_wq) { dev_err(component->dev, "Failed to create aad workqueue\n"); -- 2.17.1

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