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Re: on-board sound on ASUS TRX-40
by apt-ghetto 04 Nov '20

04 Nov '20

Hi, With the mainboard Asus Prime TRX40-Pro I do not have any on-board sound. I tried to fix it myself after reading here on the mailing list (thanks to Dan Aloni and Takashi Iwai) and with the patch below, I was able to partially fix it. The front headphone still does not work and I did not test the microphone(s) input. But at least I can hear sound when using the C/SUB, REAR and LINE OUT at the rear. As mentioned, the patch does not solve all problems, but it is a little pass towards a system with working sound. The patch below was tested by me on Gentoo with 5.4.72 and on Arch Linux with 5.9.3. I hope, it will land in the kernel sources somehow. Please feel free to modify and use it. Because the real name is required, I cannot submit the patch myself. Thank you for reading this and eventually considering to submit this to the kernel sources. ----------- Begin tested patch ----------- diff -uprN -X linux-5.4.74-vanilla/Documentation/dontdiff linux-5.4.74-vanilla/sound/usb/mixer_maps.c linux-5.4.74/sound/usb/mixer_maps.c --- linux-5.4.74-vanilla/sound/usb/mixer_maps.c 2020-11-01 12:01:07.000000000 +0100 +++ linux-5.4.74/sound/usb/mixer_maps.c 2020-11-03 11:03:15.444424941 +0100 @@ -538,6 +538,11 @@ static struct usbmix_ctl_map usbmix_ctl_ .id = USB_ID(0x0b05, 0x1917), .map = asus_rog_map, }, + { /* ASUS Prime TRX40-Pro */ + .id = USB_ID(0x0b05, 0x1918), + .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, + }, { /* MSI TRX40 Creator */ .id = USB_ID(0x0db0, 0x0d64), .map = trx40_mobo_map, diff -uprN -X linux-5.4.74-vanilla/Documentation/dontdiff linux-5.4.74-vanilla/sound/usb/quirks-table.h linux-5.4.74/sound/usb/quirks-table.h --- linux-5.4.74-vanilla/sound/usb/quirks-table.h 2020-11-03 10:53:24.000396963 +0100 +++ linux-5.4.74/sound/usb/quirks-table.h 2020-11-03 10:59:02.699412985 +0100 @@ -3667,6 +3667,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge } \ } ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */ +ALC1220_VB_DESKTOP(0x0b05, 0x1918), /* ASUS Prime TRX40-Pro */ ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */ ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ ----------- End tested patch ----------- Here is the version for the master branch, but totally untested: ----------- Begin untested patch from master branch ----------- diff --git a/sound/usb/card.c b/sound/usb/card.c index fa764b61fe9c..b6c1ffb65f69 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -393,6 +393,10 @@ static const struct usb_audio_device_name usb_audio_names[] = { DEVICE_NAME(0x0d8c, 0x0102, NULL, "ICUSBAUDIO7D"), DEVICE_NAME(0x0d8c, 0x0103, NULL, "Audio Advantage MicroII"), + /* ASUS Prime TRX40-Pro */ + PROFILE_NAME(0x0b05, 0x1918, + "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"), + /* MSI TRX40 Creator */ PROFILE_NAME(0x0db0, 0x0d64, "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"), diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index c369c81e74c4..6323faf49d67 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -563,6 +563,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x0b05, 0x1917), .map = asus_rog_map, }, + { /* ASUS Prime TRX40-Pro */ + .id = USB_ID(0x0b05, 0x1918), + .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, + }, { /* MSI TRX40 Creator */ .id = USB_ID(0x0db0, 0x0d64), .map = trx40_mobo_map, ----------- End untested patch from master branch ----------- Sent with ProtonMail Secure Email.

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[PATCH v3 0/9] ASoC: Mediatek: Add support for MT8192 SoC
by Jiaxin Yu 04 Nov '20

04 Nov '20

This series of patches adds support for Mediatek AFE for MT8192 SoC. At the same time, the calibration function of MT6359 is completed with real machine driver. The patch is based on broonie tree "for-next" branch. Change since v2: - split the dai driver files as a seperate patch - fix dt-bindings to GPL-2.0-only License - remove unnecessary preperty descriptions such as 'maxItems' Change since v1: - fixed some typos related to dt-bindings in [v1,3/5] and [v1,5/5] - add vendor prefix to the properties, such as "mediatek,apmixedsys" - add a dependency description to indicate the required header files Jiaxin Yu (9): ASoC: mediatek: mt6359: add the calibration functions ASoC: mediatek: mt8192: add platform driver ASoC: mediatek: mt8192: support i2s in platform driver ASoC: mediatek: mt8192: support add in platform driver ASoC: mediatek: mt8192: support pcm in platform driver ASoC: mediatek: mt8192: support tdm in platform driver dt-bindings: mediatek: mt8192: add audio afe document ASoC: mediatek: mt8192: add machine driver with mt6359, rt1015 and rt5682 dt-bindings: mediatek: mt8192: add mt8192-mt6358-rt1015-rt5682 document .../bindings/sound/mt8192-afe-pcm.yaml | 100 + .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 42 + sound/soc/codecs/mt6359.c | 110 + sound/soc/codecs/mt6359.h | 7 + sound/soc/mediatek/Kconfig | 23 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 13 +- sound/soc/mediatek/common/mtk-base-afe.h | 1 + sound/soc/mediatek/mt8192/Makefile | 16 + sound/soc/mediatek/mt8192/mt8192-afe-clk.c | 669 ++++ sound/soc/mediatek/mt8192/mt8192-afe-clk.h | 244 ++ sound/soc/mediatek/mt8192/mt8192-afe-common.h | 170 + .../soc/mediatek/mt8192/mt8192-afe-control.c | 163 + sound/soc/mediatek/mt8192/mt8192-afe-gpio.c | 306 ++ sound/soc/mediatek/mt8192/mt8192-afe-gpio.h | 19 + sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 2389 +++++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 1489 ++++++++ sound/soc/mediatek/mt8192/mt8192-dai-i2s.c | 2139 +++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-pcm.c | 409 +++ sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 778 ++++ .../mediatek/mt8192/mt8192-interconnection.h | 65 + .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 1058 ++++++ sound/soc/mediatek/mt8192/mt8192-reg.h | 3131 +++++++++++++++++ 23 files changed, 13338 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8192/Makefile create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-clk.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-clk.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-common.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-control.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-gpio.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-gpio.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-adda.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-interconnection.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-reg.h -- 2.18.0

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ASUS TM420 headphone enable
by Kailang 04 Nov '20

04 Nov '20

Hi Takashi, ASUS TM420 need to use COEF bit to turn on HP. Attach was enable patch. BR, Kailang

2 2

[PATCH v2 1/1] ALSA: compress: allow pause and resume during draining
by Gyeongtaek Lee 04 Nov '20

04 Nov '20

With a stream with low bitrate, user can't pause or resume the stream near the end of the stream because current ALSA doesn't allow it. If the stream has very low bitrate enough to store whole stream into the buffer, user can't do anything except stop the stream and then restart it from the first because most of applications call draining after sending last frame to the kernel. If pause, resume are allowed during draining, user experience can be enhanced. To prevent malfunction in HW drivers which don't support pause during draining, pause during draining will only work if HW driver enable this feature explicitly by calling snd_compr_use_pause_in_draining(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- include/sound/compress_driver.h | 17 +++++++++++++ sound/core/compress_offload.c | 44 +++++++++++++++++++++++++++------ 2 files changed, 53 insertions(+), 8 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 70cbc5095e72..5a0d6737de5e 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -67,6 +67,7 @@ struct snd_compr_runtime { * @metadata_set: metadata set flag, true when set * @next_track: has userspace signal next track transition, true when set * @partial_drain: undergoing partial_drain for stream, true when set + * @pause_in_draining: paused during draining state, true when set * @private_data: pointer to DSP private data * @dma_buffer: allocated buffer if any */ @@ -80,6 +81,7 @@ struct snd_compr_stream { bool metadata_set; bool next_track; bool partial_drain; + bool pause_in_draining; void *private_data; struct snd_dma_buffer dma_buffer; }; @@ -142,6 +144,7 @@ struct snd_compr_ops { * @direction: Playback or capture direction * @lock: device lock * @device: device id + * @use_pause_in_draining: allow pause in draining, true when set */ struct snd_compr { const char *name; @@ -152,6 +155,7 @@ struct snd_compr { unsigned int direction; struct mutex lock; int device; + bool use_pause_in_draining; #ifdef CONFIG_SND_VERBOSE_PROCFS /* private: */ char id[64]; @@ -166,6 +170,19 @@ int snd_compress_deregister(struct snd_compr *device); int snd_compress_new(struct snd_card *card, int device, int type, const char *id, struct snd_compr *compr); +/** + * snd_compr_use_pause_in_draining - Allow pause and resume in draining state + * @substream: compress substream to set + * + * Allow pause and resume in draining state. + * Only HW driver supports this transition can call this API. + */ +static inline void snd_compr_use_pause_in_draining( + struct snd_compr_stream *substream) +{ + substream->device->use_pause_in_draining = true; +} + /* dsp driver callback apis * For playback: driver should call snd_compress_fragment_elapsed() to let the * framework know that a fragment has been consumed from the ring buffer diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index c1fec932c49d..3f4e14cdd15a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -709,11 +709,24 @@ static int snd_compr_pause(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_RUNNING: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->device->use_pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->pause_in_draining = true; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + } return retval; } @@ -721,11 +734,25 @@ static int snd_compr_resume(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_PAUSED: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->device->use_pause_in_draining || + !stream->pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->pause_in_draining = false; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + } return retval; } @@ -768,6 +795,7 @@ static int snd_compr_stop(struct snd_compr_stream *stream) /* clear flags and stop any drain wait */ stream->partial_drain = false; stream->metadata_set = false; + stream->pause_in_draining = false; snd_compr_drain_notify(stream); stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; -- 2.21.0

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[PATCH v2 0/1] allow pause and resume during draining
by Gyeongtaek Lee 04 Nov '20

04 Nov '20

On Mon, 26 Oct 2020 18:01:33 +0100, Takashi Iwai wrote: >On Mon, 26 Oct 2020 10:18:38 +0100, >Gyeongtaek Lee wrote: >> >> With a stream with low bitrate, user can't pause or resume the stream >> near the end of the stream because current ALSA doesn't allow it. >> If the stream has very low bitrate enough to store whole stream into >> the buffer, user can't do anything except stop the stream and then >> restart it from the first because most of applications call draining >> after sending last frame to the kernel. >> If pause, resume are allowed during draining, user experience can be >> enhanced. >> To prevent malfunction in HW drivers which don't support pause >> during draining, pause during draining will only work if HW driver >> enable this feature explicitly by calling >> snd_compr_use_pause_in_draining(). >> >> Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> >> Cc: stable(a)vger.kernel.org > >Could you restart the new thread? It's been hanging too deeply and >hard to read through. > >Since it's the revised patch, please give the revision number (v2 or >such) and show what's different from the previous patches. > Ok. I'll send the patch again with [PATCH v2] prefix. >About the changes: > >> +/** >> + * snd_compr_use_pause_in_draining - Allow pause and resume in draining state >> + * @substream: compress substream to set >> + * >> + * Allow pause and resume in draining state. >> + * Only HW driver supports this transition can call this API. >> + */ >> +static inline void snd_compr_use_pause_in_draining( >> + struct snd_compr_stream *substream) >> +{ >> + substream->device->use_pause_in_draining = true; >> +} > >How to set the flag is an open question. A natural way would be to >set it somehow at creating the component object, but currently there >seems no way to pass any flags. Could you explain more about what is your concerning? For me, calling snd_compr_use_pause_in_draining() in open() callback of snd_compr_ops was good enough. I've tested it and it worked well on linux 5.4. > > >thanks, > >Takashi > Gyeongtaek Lee (1): ALSA: compress: allow pause and resume during draining include/sound/compress_driver.h | 17 +++++++++++++ sound/core/compress_offload.c | 44 +++++++++++++++++++++++++++------ 2 files changed, 53 insertions(+), 8 deletions(-) base-commit: 3cea11cd5e3b00d91caf0b4730194039b45c5891 -- 2.21.0

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(Regression) Broken pipe when streaming music from bandcamp with mpv
by GitHub issues - edited 04 Nov '20

04 Nov '20

alsa-project/alsa-lib issue #96 was edited from orbea: OS: `Slackware64-current` alsa-lib: `1.2.4` mpv: https://github.com/mpv-player/mpv/commit/172146e9f7a231b5de21921d883612d18b… yt-dlc: `2020.10.31` (https://github.com/blackjack4494/yt-dlc - bandcamp support is broken with youtube-dl currently) When streaming music from bandcamp.com with alsa it will fail with a broken pipe when automatically changing tracks at the end of a track. This will not happen if the track is skipped. ``` $ mpv https://actnoir1.bandcamp.com/album/shape-a-new-start Playing: https://actnoir1.bandcamp.com/track/shatterproof-beauty (+) Audio --aid=1 (mp3 2ch 44100Hz) AO: [alsa] 44100Hz stereo 2ch float A: 00:05:20 / 00:05:21 (100%) Cache: 0.0s Playing: https://actnoir1.bandcamp.com/track/set-fire A: 00:05:20 / 00:05:21 (100%) Cache: 0.0s [ao/alsa] snd_pcm_status: Broken pipe A: 00:05:20 / 00:05:21 (100%) Cache: 0.0s (+) Audio --aid=1 (mp3 2ch 44100Hz) [ao/alsa] snd_pcm_status: Broken pipe [ao/alsa] snd_pcm_status: Broken pipe [ao/alsa] snd_pcm_status: Broken pipe mpv: ../audio/out/buffer.c:585: _Bool ao_play_data(struct ao *): Assertion `space >= 0' failed. The process was killed by SIGABRT: Aborted ``` mpv log: [mpv.log](https://github.com/alsa-project/alsa-lib/files/5485135/mpv.log) alsa-info log: [alsa-info.log](https://github.com/alsa-project/alsa-lib/files/5485141/alsa-… I bisected this and found the first bad commit. ``` $ git bisect good 4f90392f07e8822d1984ed990f622ad36022a4a3 is the first bad commit commit 4f90392f07e8822d1984ed990f622ad36022a4a3 Author: Jaroslav Kysela <perex(a)perex.cz> Date: Fri Oct 9 19:57:57 2020 +0200 pcm: fix the snd_pcm_plugin_status() avail and delay fields The avail and delay fields in the returned status structure does not reflect the actual hw_ptr/appl_ptr. This change correct this. TODO: Unfortunately, the delay might contain also information about extra hardware / buffering delay which is hidden with this change. Link: https://lore.kernel.org/alsa-devel/d9c1f37e-5c8d-f289-270e-c6cda7a56ce3@axi… Reported-by: Jonas Holmberg <jonashg(a)axis.com> Tested-by: Jonas Holmberg <jonashg(a)axis.com> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> src/pcm/pcm_plugin.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) ``` 4f90392f07e8822d1984ed990f622ad36022a4a3 Issue URL : https://github.com/alsa-project/alsa-lib/issues/96 Repository URL: https://github.com/alsa-project/alsa-lib

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(Regression) Broken pipe when streaming music from bandcamp with mpv
by GitHub issues - opened 04 Nov '20

04 Nov '20

alsa-project/alsa-lib issue #96 was opened from orbea: OS: `Slackware64-current` alsa-lib: `1.2.4` mpv: https://github.com/mpv-player/mpv/commit/172146e9f7a231b5de21921d883612d18b… yt-dlc: `2020.10.31` When streaming music from bandcamp.com with alsa it will fail with a broken pipe when automatically changing tracks at the end of a track. This will not happen if the track is skipped. ``` $ mpv https://actnoir1.bandcamp.com/album/shape-a-new-start Playing: https://actnoir1.bandcamp.com/track/shatterproof-beauty (+) Audio --aid=1 (mp3 2ch 44100Hz) AO: [alsa] 44100Hz stereo 2ch float A: 00:05:20 / 00:05:21 (100%) Cache: 0.0s Playing: https://actnoir1.bandcamp.com/track/set-fire A: 00:05:20 / 00:05:21 (100%) Cache: 0.0s [ao/alsa] snd_pcm_status: Broken pipe A: 00:05:20 / 00:05:21 (100%) Cache: 0.0s (+) Audio --aid=1 (mp3 2ch 44100Hz) [ao/alsa] snd_pcm_status: Broken pipe [ao/alsa] snd_pcm_status: Broken pipe [ao/alsa] snd_pcm_status: Broken pipe mpv: ../audio/out/buffer.c:585: _Bool ao_play_data(struct ao *): Assertion `space >= 0' failed. The process was killed by SIGABRT: Aborted ``` mpv log: [mpv.log](https://github.com/alsa-project/alsa-lib/files/5485135/mpv.log) alsa-info log: [alsa-info.log](https://github.com/alsa-project/alsa-lib/files/5485141/alsa-… I bisected this and found the first bad commit. ``` $ git bisect good 4f90392f07e8822d1984ed990f622ad36022a4a3 is the first bad commit commit 4f90392f07e8822d1984ed990f622ad36022a4a3 Author: Jaroslav Kysela <perex(a)perex.cz> Date: Fri Oct 9 19:57:57 2020 +0200 pcm: fix the snd_pcm_plugin_status() avail and delay fields The avail and delay fields in the returned status structure does not reflect the actual hw_ptr/appl_ptr. This change correct this. TODO: Unfortunately, the delay might contain also information about extra hardware / buffering delay which is hidden with this change. Link: https://lore.kernel.org/alsa-devel/d9c1f37e-5c8d-f289-270e-c6cda7a56ce3@axi… Reported-by: Jonas Holmberg <jonashg(a)axis.com> Tested-by: Jonas Holmberg <jonashg(a)axis.com> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> src/pcm/pcm_plugin.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) ``` 4f90392f07e8822d1984ed990f622ad36022a4a3 Issue URL : https://github.com/alsa-project/alsa-lib/issues/96 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v2] ASoC: rt1015: support TDM slot configuration
by shumingf＠realtek.com 04 Nov '20

04 Nov '20

From: Shuming Fan <shumingf(a)realtek.com> Add TDM slot callback function to support TDM configuration Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1015.c | 107 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1015.h | 42 +++++++++++++++ 2 files changed, 149 insertions(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 25fe2ddedd54..b067bd175287 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -944,6 +944,112 @@ static int rt1015_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) return 0; } +static int rt1015_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int val = 0, rx_slotnum, tx_slotnum; + int ret = 0, first_bit; + + switch (slots) { + case 2: + val |= RT1015_I2S_TX_2CH; + break; + case 4: + val |= RT1015_I2S_TX_4CH; + break; + case 6: + val |= RT1015_I2S_TX_6CH; + break; + case 8: + val |= RT1015_I2S_TX_8CH; + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + switch (slot_width) { + case 16: + val |= RT1015_I2S_CH_TX_LEN_16B; + break; + case 20: + val |= RT1015_I2S_CH_TX_LEN_20B; + break; + case 24: + val |= RT1015_I2S_CH_TX_LEN_24B; + break; + case 32: + val |= RT1015_I2S_CH_TX_LEN_32B; + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Rx slot configuration */ + rx_slotnum = hweight_long(rx_mask); + if (rx_slotnum != 1) { + ret = -EINVAL; + dev_err(component->dev, "too many rx slots or zero slot\n"); + goto _set_tdm_err_; + } + + /* This is an assumption that the system sends stereo audio to the amplifier typically. + * And the stereo audio is placed in slot 0/2/4/6 as the starting slot. + * According to the slot number, we also set the mono LR to select the right slot. + */ + first_bit = __ffs(rx_mask); + switch (first_bit) { + case 0: + case 2: + case 4: + case 6: + snd_soc_component_update_bits(component, + RT1015_PAD_DRV2, RT1015_MONO_LR_SEL_MASK, + RT1015_MONO_L_CHANNEL); + snd_soc_component_update_bits(component, + RT1015_TDM1_4, + RT1015_TDM_I2S_TX_L_DAC1_1_MASK | + RT1015_TDM_I2S_TX_R_DAC1_1_MASK, + (first_bit << RT1015_TDM_I2S_TX_L_DAC1_1_SFT) | + ((first_bit+1) << RT1015_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + case 1: + case 3: + case 5: + case 7: + snd_soc_component_update_bits(component, + RT1015_PAD_DRV2, RT1015_MONO_LR_SEL_MASK, + RT1015_MONO_R_CHANNEL); + snd_soc_component_update_bits(component, + RT1015_TDM1_4, + RT1015_TDM_I2S_TX_L_DAC1_1_MASK | + RT1015_TDM_I2S_TX_R_DAC1_1_MASK, + ((first_bit-1) << RT1015_TDM_I2S_TX_L_DAC1_1_SFT) | + (first_bit << RT1015_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Tx slot configuration */ + tx_slotnum = hweight_long(tx_mask); + if (tx_slotnum) { + ret = -EINVAL; + dev_err(component->dev, "doesn't need to support tx slots\n"); + goto _set_tdm_err_; + } + + snd_soc_component_update_bits(component, RT1015_TDM1_1, + RT1015_I2S_CH_TX_MASK | RT1015_I2S_CH_RX_MASK | + RT1015_I2S_CH_TX_LEN_MASK | RT1015_I2S_CH_RX_LEN_MASK, val); + +_set_tdm_err_: + return ret; +} + static int rt1015_probe(struct snd_soc_component *component) { struct rt1015_priv *rt1015 = @@ -975,6 +1081,7 @@ static struct snd_soc_dai_ops rt1015_aif_dai_ops = { .hw_params = rt1015_hw_params, .set_fmt = rt1015_set_dai_fmt, .set_bclk_ratio = rt1015_set_bclk_ratio, + .set_tdm_slot = rt1015_set_tdm_slot, }; static struct snd_soc_dai_driver rt1015_dai[] = { diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index d3fdd30aca6d..ad8274c80990 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -213,6 +213,12 @@ #define RT1015_ID_VERA 0x0 #define RT1015_ID_VERB 0x1 +/* 0x00f2 */ +#define RT1015_MONO_LR_SEL_MASK (0x3 << 4) +#define RT1015_MONO_L_CHANNEL (0x0 << 4) +#define RT1015_MONO_R_CHANNEL (0x1 << 4) +#define RT1015_MONO_LR_MIX_CHANNEL (0x2 << 4) + /* 0x0102 */ #define RT1015_DAC_VOL_MASK (0x7f << 9) #define RT1015_DAC_VOL_SFT 9 @@ -275,6 +281,42 @@ #define RT1015_TDM_INV_BCLK_MASK (0x1 << 15) #define RT1015_TDM_INV_BCLK_SFT 15 #define RT1015_TDM_INV_BCLK (0x1 << 15) +#define RT1015_I2S_CH_TX_MASK (0x3 << 10) +#define RT1015_I2S_CH_TX_SFT 10 +#define RT1015_I2S_TX_2CH (0x0 << 10) +#define RT1015_I2S_TX_4CH (0x1 << 10) +#define RT1015_I2S_TX_6CH (0x2 << 10) +#define RT1015_I2S_TX_8CH (0x3 << 10) +#define RT1015_I2S_CH_RX_MASK (0x3 << 8) +#define RT1015_I2S_CH_RX_SFT 8 +#define RT1015_I2S_RX_2CH (0x0 << 8) +#define RT1015_I2S_RX_4CH (0x1 << 8) +#define RT1015_I2S_RX_6CH (0x2 << 8) +#define RT1015_I2S_RX_8CH (0x3 << 8) +#define RT1015_I2S_LR_CH_SEL_MASK (0x1 << 7) +#define RT1015_I2S_LR_CH_SEL_SFT 7 +#define RT1015_I2S_LEFT_CH_SEL (0x0 << 7) +#define RT1015_I2S_RIGHT_CH_SEL (0x1 << 7) +#define RT1015_I2S_CH_TX_LEN_MASK (0x7 << 4) +#define RT1015_I2S_CH_TX_LEN_SFT 4 +#define RT1015_I2S_CH_TX_LEN_16B (0x0 << 4) +#define RT1015_I2S_CH_TX_LEN_20B (0x1 << 4) +#define RT1015_I2S_CH_TX_LEN_24B (0x2 << 4) +#define RT1015_I2S_CH_TX_LEN_32B (0x3 << 4) +#define RT1015_I2S_CH_TX_LEN_8B (0x4 << 4) +#define RT1015_I2S_CH_RX_LEN_MASK (0x7 << 0) +#define RT1015_I2S_CH_RX_LEN_SFT 0 +#define RT1015_I2S_CH_RX_LEN_16B (0x0 << 0) +#define RT1015_I2S_CH_RX_LEN_20B (0x1 << 0) +#define RT1015_I2S_CH_RX_LEN_24B (0x2 << 0) +#define RT1015_I2S_CH_RX_LEN_32B (0x3 << 0) +#define RT1015_I2S_CH_RX_LEN_8B (0x4 << 0) + +/* TDM1 Setting-4 (0x011a) */ +#define RT1015_TDM_I2S_TX_L_DAC1_1_MASK (0x7 << 12) +#define RT1015_TDM_I2S_TX_R_DAC1_1_MASK (0x7 << 8) +#define RT1015_TDM_I2S_TX_L_DAC1_1_SFT 12 +#define RT1015_TDM_I2S_TX_R_DAC1_1_SFT 8 /* 0x0330 */ #define RT1015_ABST_AUTO_EN_MASK (0x1 << 13) -- 2.28.0

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[PATCH] sound/core/seq: remove dead code
by Yu Hao 03 Nov '20

03 Nov '20

The function snd_seq_queue_client_termination() is only called from function seq_free_client1(). The seq_free_client1() calls function snd_seq_queue_client_leave(), which deletes all objects whose owner equals to client->number in global array queue_list, before the function snd_seq_queue_client_termination(), which checks whether there are any objects in global array queue_list whose owner equals to client->number, with the same argument client->number. So the checking code in function snd_seq_queue_client_termination() is dead code. Remove those dead code. Signed-off-by: Yu Hao <yuhaobehappy(a)gmail.com> --- sound/core/seq/seq_queue.c | 11 ----------- 1 file changed, 11 deletions(-) diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index 71a6ea62c3be..91b3f3295d0b 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -545,21 +545,10 @@ void snd_seq_queue_client_termination(int client) unsigned long flags; int i; struct snd_seq_queue *q; - bool matched; for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) { if ((q = queueptr(i)) == NULL) continue; - spin_lock_irqsave(&q->owner_lock, flags); - matched = (q->owner == client); - if (matched) - q->klocked = 1; - spin_unlock_irqrestore(&q->owner_lock, flags); - if (matched) { - if (q->timer->running) - snd_seq_timer_stop(q->timer); - snd_seq_timer_reset(q->timer); - } queuefree(q); } } -- 2.17.1

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[PATCH] ASoC: rt715:add Mic Mute LED control support
by Perry Yuan 03 Nov '20

03 Nov '20

From: perry_yuan <perry_yuan(a)dell.com> Some new Dell system is going to support audio internal micphone privacy setting from hardware level with micmute led state changing This patch allow to change micmute led state through this micphone led control interface like hda_generic provided. Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> Signed-off-by: Limonciello Mario <mario_limonciello(a)dell.com> --- sound/soc/codecs/rt715.c | 43 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt715.h | 1 + 2 files changed, 44 insertions(+) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 099c8bd20006..2df2895d0092 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -26,6 +26,7 @@ #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/of_device.h> +#include <linux/leds.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -213,6 +214,45 @@ static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +static const char *rt715_micmute_led_mode[] = { + "Off", "On" +}; + +static const struct soc_enum rt715_micmute_led_mode_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(rt715_micmute_led_mode), + rt715_micmute_led_mode); + +static int rt715_mic_mute_led_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + int led_mode = rt715->micmute_led; + + ucontrol->value.integer.value[0] = led_mode; +#if IS_ENABLED(CONFIG_LEDS_TRIGGER_AUDIO) + ledtrig_audio_set(LED_AUDIO_MICMUTE, + rt715->micmute_led ? LED_ON : LED_OFF); +#endif + return 0; +} + +static int rt715_micmute_led_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + int led_mode = ucontrol->value.integer.value[0]; + + rt715->micmute_led = led_mode; +#if IS_ENABLED(CONFIG_LEDS_TRIGGER_AUDIO) + ledtrig_audio_set(LED_AUDIO_MICMUTE, + rt715->micmute_led ? LED_ON : LED_OFF); +#endif + return 0; +} + + static const struct snd_kcontrol_new rt715_snd_controls[] = { /* Capture switch */ SOC_DOUBLE_R_EXT("ADC 07 Capture Switch", RT715_SET_GAIN_MIC_ADC_H, @@ -277,6 +317,9 @@ static const struct snd_kcontrol_new rt715_snd_controls[] = { RT715_SET_GAIN_LINE2_L, RT715_DIR_IN_SFT, 3, 0, rt715_set_amp_gain_get, rt715_set_amp_gain_put, mic_vol_tlv), + /*Micmute Led Control*/ + SOC_ENUM_EXT("Micmute Led Mode", rt715_micmute_led_mode_enum, + rt715_mic_mute_led_mode_get, rt715_micmute_led_mode_put), }; static int rt715_mux_get(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/rt715.h b/sound/soc/codecs/rt715.h index df0f24f9bc0c..32917b7846b4 100644 --- a/sound/soc/codecs/rt715.h +++ b/sound/soc/codecs/rt715.h @@ -22,6 +22,7 @@ struct rt715_priv { struct sdw_bus_params params; bool hw_init; bool first_hw_init; + int micmute_led; }; struct sdw_stream_data { -- 2.25.1

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