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

[PATCH AUTOSEL 5.17 01/12] ALSA: usb-audio: Don't get sample rate for MCT Trigger 5 USB-to-HDMI
by Sasha Levin 24 May '22

24 May '22

From: Forest Crossman <cyrozap(a)gmail.com> [ Upstream commit d7be213849232a2accb219d537edf056d29186b4 ] This device doesn't support reading the sample rate, so we need to apply this quirk to avoid a 15-second delay waiting for three timeouts. Signed-off-by: Forest Crossman <cyrozap(a)gmail.com> Link: https://lore.kernel.org/r/20220504002444.114011-2-cyrozap@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ab9f3da49941..fbbe59054c3f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1822,6 +1822,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x06f8, 0xd002, /* Hercules DJ Console (Macintosh Edition) */ QUIRK_FLAG_IGNORE_CTL_ERROR), + DEVICE_FLG(0x0711, 0x5800, /* MCT Trigger 5 USB-to-HDMI */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x074d, 0x3553, /* Outlaw RR2150 (Micronas UAC3553B) */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */ -- 2.35.1

1 2

Add missing implementation for topology ignore_suspend flag
by GitHub pull_request - opened 24 May '22

24 May '22

alsa-project/alsa-lib pull request #230 was opened from amadeuszslawinski-intel: ignore_suspend flag was ignored while parsing topology, add missing support for it. This allows use of it on DAI links and DAPM widgets. Request URL : https://github.com/alsa-project/alsa-lib/pull/230 Patch URL : https://github.com/alsa-project/alsa-lib/pull/230.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Fix for arecord recording ghost data
by GitHub pull_request - opened 24 May '22

24 May '22

alsa-project/alsa-utils pull request #150 was opened from amadeuszslawinski-intel: We noticed the issue in our tests, when using arecord would produce files with glitches at recording end. Apparently it was caused by writing full buffer to output file instead of the data read, when stopping recording. Following commit adjust the code to only write the amount of data that was read and making sure that file size matches internal format data. Request URL : https://github.com/alsa-project/alsa-utils/pull/150 Patch URL : https://github.com/alsa-project/alsa-utils/pull/150.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Add new type for ALC245
by Kailang 24 May '22

24 May '22

Hi Takashi, Attach patch will add new codec type for ALC245. BR, Kailang

2 1

[PATCH] ALSA: usb-audio: Configure sync endpoints before data
by Takashi Iwai 24 May '22

24 May '22

From: Craig McLure <craig(a)mclure.net> Devices such as the TC-Helicon GoXLR require the sync endpoint to be configured in advance of the data endpoint in order for sound output to work. This patch simply changes the ordering of EP configuration to resolve this. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215079 Signed-off-by: Craig McLure <craig(a)mclure.net> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 6d699065e81a..b470404a5376 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -439,16 +439,21 @@ static int configure_endpoints(struct snd_usb_audio *chip, /* stop any running stream beforehand */ if (stop_endpoints(subs, false)) sync_pending_stops(subs); + if (subs->sync_endpoint) { + err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); + if (err < 0) + return err; + } err = snd_usb_endpoint_configure(chip, subs->data_endpoint); if (err < 0) return err; snd_usb_set_format_quirk(subs, subs->cur_audiofmt); - } - - if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); - if (err < 0) - return err; + } else { + if (subs->sync_endpoint) { + err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); + if (err < 0) + return err; + } } return 0; -- 2.35.3

2 1

GoXLR: Fixed sampler input configuration
by GitHub pull_request - opened 24 May '22

24 May '22

alsa-project/alsa-ucm-conf pull request #155 was opened from FrostyCoolSlug: Renamed the 'Sample' input channel to 'Sampler' to better reflect it's behaviour, and corrected channel numbers. This fixes sampler capture on the GoXLR. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/155 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/155.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

amixer returns a RC=1 when you use 'amixer -v' while other 'amixer' commands (like 'amixer -h` or `amixer --help') return with a '0'.
by GitHub issues - opened 23 May '22

23 May '22

alsa-project/alsa-utils issue #149 was opened from juggledad: amixer version 1.2.4 Raspberry Pi Zero W v1.1 Raspbian GNU/Linux 11 (bullseye) - desktop version Node-RED v2.2.2 This was discovered using an `exec` node in on a The exec node runs a command and returns its output. When running a test, I wanted to get the amixer version. There are two ways to ask `amixer` for it's version number 1 - use 'amixer -v' 2 - use 'amixer --version` Both returned the version number but had a Return Code of 1. All other 'amixer' commands (like 'amixer -h` or `amixer --help') return with a '0'. Here is a copy of the node-red flow that was used: ` [{"active":true,"complete":"true","console":false,"id":"38409c9414d4fb0b","name":"stdout","statusType":"auto","statusVal":"","targetType":"full","tosidebar":true,"tostatus":false,"type":"debug","wires":[],"x":730,"y":80,"z":"58ad38887591e1b1"},{"active":true,"complete":"true","console":false,"id":"de852fbd07407b7a","name":"stderr","statusType":"auto","statusVal":"","targetType":"full","tosidebar":true,"tostatus":false,"type":"debug","wires":[],"x":730,"y":120,"z":"58ad38887591e1b1"},{"active":true,"complete":"true","console":false,"id":"42faac21ce0190b3","name":"return code","statusType":"auto","statusVal":"","targetType":"full","tosidebar":true,"tostatus":false,"type":"debug","wires":[],"x":750,"y":160,"z":"58ad38887591e1b1"},{"crontab":"","id":"84f238ce5b65227d","name":"","once":false,"onceDelay":0.1,"payload":"amixer -v","payloadType":"str","props":[{"p":"payload"},{"p":"topic","vt":"str"}],"repeat":"","topic":"","type":"inject","wires":[["05c0a4eb46c0c1c8"]],"x":180,"y":60,"z":"58ad38887591e1b1"},{"crontab":"","id":"fabfd7b565392096","name":"","once":false,"onceDelay":0.1,"payload":"amixer -h","payloadType":"str","props":[{"p":"payload"},{"p":"topic","vt":"str"}],"repeat":"","topic":"","type":"inject","wires":[["05c0a4eb46c0c1c8"]],"x":180,"y":180,"z":"58ad38887591e1b1"},{"crontab":"","id":"c54064c5cc3b2484","name":"","once":false,"onceDelay":0.1,"payload":"amixer --version","payloadType":"str","props":[{"p":"payload"},{"p":"topic","vt":"str"}],"repeat":"","topic":"","type":"inject","wires":[["05c0a4eb46c0c1c8"]],"x":200,"y":120,"z":"58ad38887591e1b1"},{"addpay":"payload","append":"","command":"","id":"05c0a4eb46c0c1c8","name":"","oldrc":false,"timer":"","type":"exec","useSpawn":"false","winHide":false,"wires":[["38409c9414d4fb0b"],["de852fbd07407b7a"],["42faac21ce0190b3"]],"x":490,"y":120,"z":"58ad38887591e1b1"}] ` Issue URL : https://github.com/alsa-project/alsa-utils/issues/149 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: fsl_sai: Fix FSL_SAI_xDR/xFR definition
by Shengjiu Wang 23 May '22

23 May '22

There are multiple xDR and xFR registers, the index is from 0 to 7. FSL_SAI_xDR and FSL_SAI_xFR is abandoned, replace them with FSL_SAI_xDR0 and FSL_SAI_xFR0. Fixes: 4f7a0728b530 ("ASoC: fsl_sai: Add support for SAI new version") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index e4965efe6102..1c8f5ca07f9d 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -80,8 +80,8 @@ #define FSL_SAI_xCR3(tx, ofs) (tx ? FSL_SAI_TCR3(ofs) : FSL_SAI_RCR3(ofs)) #define FSL_SAI_xCR4(tx, ofs) (tx ? FSL_SAI_TCR4(ofs) : FSL_SAI_RCR4(ofs)) #define FSL_SAI_xCR5(tx, ofs) (tx ? FSL_SAI_TCR5(ofs) : FSL_SAI_RCR5(ofs)) -#define FSL_SAI_xDR(tx, ofs) (tx ? FSL_SAI_TDR(ofs) : FSL_SAI_RDR(ofs)) -#define FSL_SAI_xFR(tx, ofs) (tx ? FSL_SAI_TFR(ofs) : FSL_SAI_RFR(ofs)) +#define FSL_SAI_xDR0(tx) (tx ? FSL_SAI_TDR0 : FSL_SAI_RDR0) +#define FSL_SAI_xFR0(tx) (tx ? FSL_SAI_TFR0 : FSL_SAI_RFR0) #define FSL_SAI_xMR(tx) (tx ? FSL_SAI_TMR : FSL_SAI_RMR) /* SAI Transmit/Receive Control Register */ -- 2.17.1

2 1

[PATCH] ASoC: soc-pcm: fix BE transition for TRIGGER_START
by Pierre-Louis Bossart 23 May '22

23 May '22

A obvious editing mistake caught with a cppcheck warning sound/soc/soc-pcm.c:2132:8: style: Variable 'ret' is reassigned a value before the old one has been used. [redundantAssignment] ret = soc_pcm_trigger(be_substream, cmd); ^ sound/soc/soc-pcm.c:2126:9: note: ret is assigned ret = soc_pcm_trigger(be_substream, ^ sound/soc/soc-pcm.c:2129:9: note: ret is assigned ret = soc_pcm_trigger(be_substream, ^ Fixes: 374b50e234a3e ('ASoC: soc-pcm: improve BE transition for TRIGGER_START') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/soc-pcm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6f43db35a5c84..a827cc3c158ae 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2128,8 +2128,6 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, else ret = soc_pcm_trigger(be_substream, SNDRV_PCM_TRIGGER_START); - - ret = soc_pcm_trigger(be_substream, cmd); if (ret) { be->dpcm[stream].be_start--; goto next; -- 2.30.2

2 1

[v6] FROMLIST: ASoC: Intel: sof_cs42l42: adding support for ADL configuration and BT offload audio
by Terry Chen 23 May '22

23 May '22

To be able to do driver data for adl_mx98360a_cs4242 which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on ADL platform. Also add the capability to machine driver of creating DAI Link for BT offload. Although BT offload always uses SSP2 port but we reserve the flexibility to assign the port number in macro. Signed-off-by: Terry Chen <terry_chen(a)wistron.corp-partner.google.com> (am from https://patchwork.kernel.org/patch/12845884/) (also found at https://lore.kernel.org/r/20220511075522.1764114-1-terry_chen@wistron.corp-…) --- sound/soc/intel/boards/sof_cs42l42.c | 92 ++++++++++++++++++- .../intel/common/soc-acpi-intel-adl-match.c | 7 ++ 2 files changed, 95 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index ce78c18798876..2efffc7933479 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -41,8 +41,13 @@ #define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) #define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(25) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_BT_OFFLOAD_PRESENT BIT(25) +#define SOF_CS42L42_SSP_BT_SHIFT 26 +#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) +#define SOF_CS42L42_SSP_BT(quirk) \ + (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(29) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(30) enum { LINK_NONE = 0, @@ -50,6 +55,7 @@ enum { LINK_SPK = 2, LINK_DMIC = 3, LINK_HDMI = 4, + LINK_BT = 5, }; /* Default: SSP2 */ @@ -278,6 +284,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int create_spk_amp_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -467,9 +480,56 @@ static int create_hdmi_dai_links(struct device *dev, return -ENOMEM; } +static int create_bt_offload_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_bt) +{ + int ret = 0; + + /* bt offload */ + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) + return 0; + + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ssp_bt); + if (!links[*id].name) { + ret = -ENOMEM; + goto devm_err; + } + + links[*id].id = *id; + links[*id].codecs = dummy_component; + links[*id].num_codecs = ARRAY_SIZE(dummy_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_bt); + if (!links[*id].cpus->dai_name) { + ret = -ENOMEM; + goto devm_err; + } + + (*id)++; + + return 0; + +devm_err: + return ret; +} + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, + int ssp_bt, int dmic_be_num, int hdmi_num) { @@ -522,6 +582,14 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; } break; + case LINK_BT: + ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); + if (ret < 0) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + break; case LINK_NONE: /* caught here if it's not used as terminator in macro */ default: @@ -543,7 +611,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; - int ret, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -568,6 +636,9 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + SOF_CS42L42_SSP_BT_SHIFT; + ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; @@ -578,9 +649,11 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_cs42l42.num_links++; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, - dmic_be_num, hdmi_num); + ssp_bt, dmic_be_num, hdmi_num); if (!dai_links) return -ENOMEM; @@ -621,6 +694,17 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1)) | SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), }, + { + .name = "adl_mx98360a_cs4242", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1) | + SOF_CS42L42_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_SSP_BT(2)) | + SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 7c8cd00457f81..311adeb3afbc4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -384,6 +384,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-cs35l41.tplg", }, + { + .id = "10134242", + .drv_name = "adl_mx98360a_cs4242", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.25.1

3 3