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

Copy hw out to loop problem when recording started before playing
by Anton Nikolaev 28 Jul '21

28 Jul '21

Hello everyone. Please help. Do not understand what is wrong Chains is : Plug asym1 - Out= Plug with multi (Copy hardware dmix output to dmix loopout) IN= Dsnoop with hardware dev Plug asym2 - Out= plug dmix loop IN= plug dsnoop loop Work good if playing to plug asym1 started before recording from asym2. Not working play to asym1 if recording from asym2 started before playing to asym1. /Etc/asound.conf is: pcm.hwcard0 { type hw; card 0; device 0; } pcm.!default { type plug; slave.pcm { type asym; playback.pcm "devhwcopy"; capture.pcm "devhwin"; } } pcm.devhwcopy { type plug; route_policy "duplicate"; slave.pcm { type multi; slaves { a { channels 2; pcm "devhwout"; } b { channels 2; pcm "devloopout"; } } bindings { 0 { slave a channel 0 } 1 { slave a channel 1 } 2 { slave b channel 0 } 3 { slave b channel 1 } } } ttable [ [ 1 0 1 0 ] [ 0 1 0 1 ] ] } pcm.devhwout { type dmix; ipc_key 1024001; ipc_perm 0666; hw_ptr_alignment roundup; slave { pcm "hwcard0"; channels 2; period_size 1024; buffer_size 8192; } bindings { 0 0 1 1 } } pcm.devhwin { type dsnoop; ipc_key 1024002; ipc_perm 0666; slave { pcm "hwcard0"; channels 2; rate 44100; period_size 1024; buffer_size 8192; } bindings { 0 0 1 1 } } pcm.devloopout { type dmix; ipc_key 1024011; ipc_perm 0666; slave { pcm "hw:LOOPCARD,0,0"; channels 2; period_size 1024; buffer_size 8192; } bindings { 0 0 1 1 } } pcm.devloopin { type dsnoop; ipc_key 1024012; ipc_perm 0666; slave { pcm "hw:LOOPCARD,1,0"; channels 2; rate 44100; period_size 1024; buffer_size 8192; } bindings { 0 0 1 1 } } pcm.devloop { type plug; slave.pcm { type asym; playback.pcm "devloopout"; capture.pcm "devloopin"; } } (English is not native for me, sorry)

1 0

[PATCH 0/5] ASoC: soc-pcm: cleanup cppcheck warning
by Kuninori Morimoto 28 Jul '21

28 Jul '21

Hi Mark Now I'm posting audio-graph-card2 patch-set, but it seems it needs longer discussion. Thus I want to post more easy patch first, and reduce my local patches. These are cppcheck warning cleanup patches for soc-pcm. Kuninori Morimoto (5): ASoC: soc-pcm: cleanup cppcheck warning at soc_pcm_apply_msb() ASoC: soc-pcm: cleanup cppcheck warning at soc_pcm_components_close() ASoC: soc-pcm: cleanup cppcheck warning at soc_get_playback_capture() ASoC: soc-pcm: cleanup cppcheck warning at dpcm_be_is_active() ASoC: soc-pcm: cleanup cppcheck warning at dpcm_runtime_setup_be_chan() sound/soc/soc-pcm.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) -- 2.25.1 Thank you for your help !! Best regards --- Kuninori Morimoto

3 8

Pulseaudio dropping GPU audio
by Liam Hupfer 27 Jul '21

27 Jul '21

Hello, I’ve set up a Linux VM where I’m passing through an RX 580. I’m using the GPU audio and my monitor’s audio jack. For some reason the Linux VM keeps losing track of the audio output. It always shows up under `lspci' under ID `06:00.0' using the `snd_hda_intel' driver, but `pulseaudio' (I also tried `pipewire', same issue) keep losing it. Most times I can restart `pulseaudio' and it will find the device again; occasionally, it takes multiple restarts. I checked the Arch wiki and I found something[¹] on audio over HDMI (I’m using DisplayPort, if that matters). I checked the device and it did report `Enable+'. Even so, I added the kernel parameter `snd_hda_intel.enable_msi=1', but nothing changed. Here are some `journalctl --user -u pulseaudio' logs: ┌──── │ 20:37:31 host pulseaudio[2373]: ALSA woke us up to write new data to the device, but there was actually nothing to write. │ 20:37:31 host pulseaudio[2373]: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. │ 20:37:31 host pulseaudio[2373]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. └──── Anyone have any suggestions for troubleshooting? `pulseaudio'’s logs suggested it might have to do with ALSA, so I asked for help here. Thank you! —Liam [¹] <https://wiki.archlinux.org/title/PCI_passthrough_via_OVMF#Slowed_down_audio…>

2 1

alsaloop: Support "Capture Pitch 1000000" rate shift
by GitHub pull_request - opened 27 Jul '21

27 Jul '21

alsa-project/alsa-utils pull request #105 was opened from t123yh: In Linux 5.14, a new feedback mechanism for USB Audio Gadget is introduced (commit e89bb428), along with a new control element "Capture Pitch 1000000". This patch adds support for this feature. Request URL : https://github.com/alsa-project/alsa-utils/pull/105 Patch URL : https://github.com/alsa-project/alsa-utils/pull/105.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: SOF: Intel: hda: enforce exclusion between HDaudio and SoundWire
by Pierre-Louis Bossart 27 Jul '21

27 Jul '21

On some platforms with an external HDaudio codec, the DSDT reports the presence of SoundWire devices. Pin-mux restrictions and board reworks usually prevent coexistence between the two types of links, let's prevent unnecessary operations from starting. In the case of a single iDISP codec being detected, we still start the links even if no SoundWire machine configuration was detected, so that we can double-check what the hardware is and add the missing configuration if applicable. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> --- sound/soc/sof/intel/hda.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index e1e368ff2b12..891e6e1b9121 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -187,12 +187,16 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev) int hda_sdw_startup(struct snd_sof_dev *sdev) { struct sof_intel_hda_dev *hdev; + struct snd_sof_pdata *pdata = sdev->pdata; hdev = sdev->pdata->hw_pdata; if (!hdev->sdw) return 0; + if (pdata->machine && !pdata->machine->mach_params.link_mask) + return 0; + return sdw_intel_startup(hdev->sdw); } @@ -1002,6 +1006,14 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev) hda_mach->mach_params.dmic_num = dmic_num; pdata->machine = hda_mach; pdata->tplg_filename = tplg_filename; + + if (codec_num == 2) { + /* + * Prevent SoundWire links from starting when an external + * HDaudio codec is used + */ + hda_mach->mach_params.link_mask = 0; + } } } -- 2.25.1

2 1

[PATCH] ASoC: topology: Select SND_DYNAMIC_MINORS
by Pierre-Louis Bossart 27 Jul '21

27 Jul '21

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> The indexes of the devices are described within the topology file, it is a possibility that the topology encodes invalid indexes when DYNAMIC_MINORS is not enabled in kernel: #define SNDRV_MINOR_COMPRESS 2 /* 2 - 3 */ #define SNDRV_MINOR_HWDEP 4 /* 4 - 7 */ #define SNDRV_MINOR_RAWMIDI 8 /* 8 - 15 */ #define SNDRV_MINOR_PCM_PLAYBACK 16 /* 16 - 23 */ #define SNDRV_MINOR_PCM_CAPTURE 24 /* 24 - 31 */ If the topology assigns an index greater than 7 for PLAYBACK/CAPTURE PCM then there will be minor number collision. As an example: card0 creates a capture PCM with index 10 -> minor = 34 card1 creates compress device with index 0 -> minor = 34 Card1 will fail to instantiate because the minor for the compress stream is already taken. To avoid seemingly mysterious issues with card creation, select the DYNAMIC_MINORS when the topology is enabled. The other option would be to try to do out of bound index checks in case of DYNAMIC_MINOR is not enabled and do not even attempt to create the device with failing the topology load. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 8a13462e1a63..5dcf77af07af 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_COMPRESS config SND_SOC_TOPOLOGY bool + select SND_DYNAMIC_MINORS config SND_SOC_TOPOLOGY_KUNIT_TEST tristate "KUnit tests for SoC topology" -- 2.25.1

2 1

[PATCH] ASoC: amd: fix an IS_ERR() vs NULL bug in probe
by Dan Carpenter 27 Jul '21

27 Jul '21

The devm_ioremap() function returns NULL on error, it doesn't return error pointers. Fixes: e550339ee652 ("ASoC: amd: add vangogh i2s controller driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/amd/vangogh/acp5x-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-i2s.c b/sound/soc/amd/vangogh/acp5x-i2s.c index 86b37c93c3d2..2705e57eb713 100644 --- a/sound/soc/amd/vangogh/acp5x-i2s.c +++ b/sound/soc/amd/vangogh/acp5x-i2s.c @@ -399,8 +399,8 @@ static int acp5x_dai_probe(struct platform_device *pdev) } adata->acp5x_base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); - if (IS_ERR(adata->acp5x_base)) - return PTR_ERR(adata->acp5x_base); + if (!adata->acp5x_base) + return -ENOMEM; adata->master_mode = I2S_MASTER_MODE_ENABLE; dev_set_drvdata(&pdev->dev, adata); -- 2.20.1

2 1

[PATCH] ASoC: rt5682: enable SAR ADC power saving mode during suspend
by shumingf＠realtek.com 27 Jul '21

27 Jul '21

From: Shuming Fan <shumingf(a)realtek.com> The SAR ADC power saving mode could reduce power consumption on MICVDD rail. Therefore, this patch saves power consumption during suspend state if the headset was connected. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 44 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 51ecaa2abcd1..7dc01ae6bb66 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2916,10 +2916,46 @@ static void rt5682_remove(struct snd_soc_component *component) static int rt5682_suspend(struct snd_soc_component *component) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int val; if (rt5682->is_sdw) return 0; + cancel_delayed_work_sync(&rt5682->jack_detect_work); + cancel_delayed_work_sync(&rt5682->jd_check_work); + if (rt5682->hs_jack && rt5682->jack_type == SND_JACK_HEADSET) { + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_MB1_PATH_MASK | RT5682_MB2_PATH_MASK, + RT5682_CTRL_MB1_REG | RT5682_CTRL_MB2_REG); + val = snd_soc_component_read(component, + RT5682_CBJ_CTRL_2) & RT5682_JACK_TYPE_MASK; + + switch (val) { + case 0x1: + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_SEL_MB1_MASK | RT5682_SAR_SEL_MB2_MASK, + RT5682_SAR_SEL_MB1_NOSEL | RT5682_SAR_SEL_MB2_SEL); + break; + case 0x2: + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_SEL_MB1_MASK | RT5682_SAR_SEL_MB2_MASK, + RT5682_SAR_SEL_MB1_SEL | RT5682_SAR_SEL_MB2_NOSEL); + break; + default: + break; + } + + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ, 0); + + /* enter SAR ADC power saving mode */ + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK | RT5682_SAR_BUTDET_RST_MASK, 0); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK | RT5682_SAR_BUTDET_RST_MASK, + RT5682_SAR_BUTT_DET_EN | RT5682_SAR_BUTDET_POW_SAV | RT5682_SAR_BUTDET_RST_NORMAL); + } + regcache_cache_only(rt5682->regmap, true); regcache_mark_dirty(rt5682->regmap); return 0; @@ -2935,6 +2971,14 @@ static int rt5682_resume(struct snd_soc_component *component) regcache_cache_only(rt5682->regmap, false); regcache_sync(rt5682->regmap); + if (rt5682->hs_jack && rt5682->jack_type == SND_JACK_HEADSET) { + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_MB1_PATH_MASK | RT5682_MB2_PATH_MASK, + RT5682_CTRL_MB1_FSM | RT5682_CTRL_MB2_FSM); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ, RT5682_PWR_CBJ); + } + mod_delayed_work(system_power_efficient_wq, &rt5682->jack_detect_work, msecs_to_jiffies(250)); -- 2.31.1

2 1

[PATCH AUTOSEL 4.19 2/6] ASoC: tlv320aic31xx: fix reversed bclk/wclk master bits
by Sasha Levin 27 Jul '21

27 Jul '21

From: Kyle Russell <bkylerussell(a)gmail.com> [ Upstream commit 9cf76a72af6ab81030dea6481b1d7bdd814fbdaf ] These are backwards from Table 7-71 of the TLV320AIC3100 spec [1]. This was broken in 12eb4d66ba2e when BCLK_MASTER and WCLK_MASTER were converted from 0x08 and 0x04 to BIT(2) and BIT(3), respectively. -#define AIC31XX_BCLK_MASTER 0x08 -#define AIC31XX_WCLK_MASTER 0x04 +#define AIC31XX_BCLK_MASTER BIT(2) +#define AIC31XX_WCLK_MASTER BIT(3) Probably just a typo since the defines were not listed in bit order. [1] https://www.ti.com/lit/gpn/tlv320aic3100 Signed-off-by: Kyle Russell <bkylerussell(a)gmail.com> Link: https://lore.kernel.org/r/20210622010941.241386-1-bkylerussell@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 0b587585b38b..6071de5fca57 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -147,8 +147,8 @@ struct aic31xx_pdata { #define AIC31XX_WORD_LEN_24BITS 0x02 #define AIC31XX_WORD_LEN_32BITS 0x03 #define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2) -#define AIC31XX_BCLK_MASTER BIT(2) -#define AIC31XX_WCLK_MASTER BIT(3) +#define AIC31XX_BCLK_MASTER BIT(3) +#define AIC31XX_WCLK_MASTER BIT(2) /* AIC31XX_DATA_OFFSET */ #define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0) -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 3/9] ASoC: tlv320aic31xx: fix reversed bclk/wclk master bits
by Sasha Levin 27 Jul '21

27 Jul '21

From: Kyle Russell <bkylerussell(a)gmail.com> [ Upstream commit 9cf76a72af6ab81030dea6481b1d7bdd814fbdaf ] These are backwards from Table 7-71 of the TLV320AIC3100 spec [1]. This was broken in 12eb4d66ba2e when BCLK_MASTER and WCLK_MASTER were converted from 0x08 and 0x04 to BIT(2) and BIT(3), respectively. -#define AIC31XX_BCLK_MASTER 0x08 -#define AIC31XX_WCLK_MASTER 0x04 +#define AIC31XX_BCLK_MASTER BIT(2) +#define AIC31XX_WCLK_MASTER BIT(3) Probably just a typo since the defines were not listed in bit order. [1] https://www.ti.com/lit/gpn/tlv320aic3100 Signed-off-by: Kyle Russell <bkylerussell(a)gmail.com> Link: https://lore.kernel.org/r/20210622010941.241386-1-bkylerussell@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index cb024955c978..73c5f6c8ed69 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -151,8 +151,8 @@ struct aic31xx_pdata { #define AIC31XX_WORD_LEN_24BITS 0x02 #define AIC31XX_WORD_LEN_32BITS 0x03 #define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2) -#define AIC31XX_BCLK_MASTER BIT(2) -#define AIC31XX_WCLK_MASTER BIT(3) +#define AIC31XX_BCLK_MASTER BIT(3) +#define AIC31XX_WCLK_MASTER BIT(2) /* AIC31XX_DATA_OFFSET */ #define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0) -- 2.30.2

1 0