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

[PATCH AUTOSEL 5.6 62/73] ALSA: ice1724: Fix invalid access for enumerated ctl items
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c47914c00be346bc5b48c48de7b0da5c2d1a296c ] The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 91f83cef0e56c..9aa12a67d3705 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -536,7 +536,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -550,7 +550,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; -- 2.20.1

1 0

[PATCH AUTOSEL 5.6 58/73] ALSA: hda/realtek - Add quirk for MSI GL63
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 1d3aa4a5516d2e4933fe3cca11d3349ef63bc547 ] MSI GL63 laptop requires the similar quirk like other MSI models, ALC1220_FIXUP_CLEVO_P950. The board BIOS doesn't provide a PCI SSID for the device, hence we need to take the codec SSID (1462:1275) instead. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207157 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408135645.21896-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 63e1a56f705b7..5cced684cf41f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2449,6 +2449,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), -- 2.20.1

1 0

[PATCH AUTOSEL 5.6 57/73] ALSA: hda: Add driver blacklist
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 3c6fd1f07ed03a04debbb9a9d782205f1ef5e2ab ] The recent AMD platform exposes an HD-audio bus but without any actual codecs, which is internally tied with a USB-audio device, supposedly. It results in "no codecs" error of HD-audio bus driver, and it's nothing but a waste of resources. This patch introduces a static blacklist table for skipping such a known bogus PCI SSID entry. As of writing this patch, the known SSIDs are: * 1043:874f - ASUS ROG Zenith II / Strix * 1462:cb59 - MSI TRX40 Creator * 1462:cb60 - MSI TRX40 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408140449.22319-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 92a042e34d3e5..bd093593f8fbd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2076,6 +2076,17 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, #endif } +/* Blacklist for skipping the whole probe: + * some HD-audio PCI entries are exposed without any codecs, and such devices + * should be ignored from the beginning. + */ +static const struct snd_pci_quirk driver_blacklist[] = { + SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0), + SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), + SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), + {} +}; + static const struct hda_controller_ops pci_hda_ops = { .disable_msi_reset_irq = disable_msi_reset_irq, .pcm_mmap_prepare = pcm_mmap_prepare, @@ -2092,6 +2103,11 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; + if (snd_pci_quirk_lookup(pci, driver_blacklist)) { + dev_info(&pci->dev, "Skipping the blacklisted device\n"); + return -ENODEV; + } + if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { -- 2.20.1

1 0

[PATCH AUTOSEL 5.6 22/73] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 81630dc042af998b9f58cd8e2c29dab9777ea176 ] sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index baef461a99f19..2c3798034b1de 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -966,7 +966,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.20.1

1 0

[PATCH AUTOSEL 5.6 19/73] ASoC: dpcm: allow start or stop during pause for backend
by Sasha Levin 18 Apr '20

18 Apr '20

From: 이경택 <gt82.lee(a)samsung.com> [ Upstream commit 21fca8bdbb64df1297e8c65a746c4c9f4a689751 ] soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c59b3688ca01..8f6f0ad502880 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2236,7 +2236,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2266,7 +2267,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.20.1

1 0

[PATCH AUTOSEL 5.6 18/73] ASoC: qcom: q6asm-dai: Add SNDRV_PCM_INFO_BATCH flag
by Sasha Levin 18 Apr '20

18 Apr '20

From: Stephan Gerhold <stephan(a)gerhold.net> [ Upstream commit 7f2430cda819a9ecb1df5a0f3ef4f1c20db3f811 ] At the moment, playing audio with PulseAudio with the qdsp6 driver results in distorted sound. It seems like its timer-based scheduling does not work properly with qdsp6 since setting tsched=0 in the PulseAudio configuration avoids the issue. Apparently this happens when the pointer() callback is not accurate enough. There is a SNDRV_PCM_INFO_BATCH flag that can be used to stop PulseAudio from using timer-based scheduling by default. According to https://www.alsa-project.org/pipermail/alsa-devel/2014-March/073816.html: The flag is being used in the sense explained in the previous audio meeting -- the data transfer granularity isn't fine enough but aligned to the period size (or less). q6asm-dai reports the position as multiple of prtd->pcm_count = snd_pcm_lib_period_bytes(substream) so it indeed just a multiple of the period size. Therefore adding the flag here seems appropriate and makes audio work out of the box. Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20200330175210.47518-1-stephan@gerhold.net Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c0d422d0ab94f..d7dc80ede8927 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -73,7 +73,7 @@ struct q6asm_dai_data { }; static const struct snd_pcm_hardware q6asm_dai_hardware_capture = { - .info = (SNDRV_PCM_INFO_MMAP | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | @@ -95,7 +95,7 @@ static const struct snd_pcm_hardware q6asm_dai_hardware_capture = { }; static struct snd_pcm_hardware q6asm_dai_hardware_playback = { - .info = (SNDRV_PCM_INFO_MMAP | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | -- 2.20.1

1 0

State of hda-analyser.py
by Moosa Mahsoom 18 Apr '20

18 Apr '20

Hi, I am a final year student from BITS Pilani, India. I have been facing some issues with my laptop audio and hence wanted to use the hda_analyser script. However, I noticed that the python script doesn't seem to compatible with the current pygtk libraries. If possible,I would like to contribute towards bringing the scripts to parity. Yours Sincerely, Moosa Mahsoom

1 0

Bug WD19 dockstation
by Gregorio Rodrigo 17 Apr '20

17 Apr '20

Hi everyone, I am not sure if this is the proper way to submit a bug/ticket/question. I posted a message here https://github.com/edrose/dell-dock-audio-fix/issues/2#issuecomment-6152907… and got the suggestion to write to the alsa mail list. I have a WD19 docking station and the lineout is not recognized. It seems it is recognized as an input. However it works if I run the speaker-test command. Anyone knows if I can apply those UCM settings or how could I do it? I copy here the message. I have a WD19 docking station and i am facing the same problem. Debian GNU/Linux bullseye/sid \n \l Linux xps 5.5.0-1-amd64 #1 <https://github.com/edrose/dell-dock-audio-fix/issues/1> SMP Debian 5.5.13-2 (2020-03-30) x86_64 GNU/Linux pactl list cards Card #1 <https://github.com/edrose/dell-dock-audio-fix/issues/1> Name: alsa_card.usb-Generic_USB_Audio_200901010001-00 Driver: module-alsa-card.c Owner Module: 25 Properties: alsa.card = "2" alsa.card_name = "WD19 Dock" alsa.long_card_name = "Dell-WD15-Dock" alsa.driver_name = "snd_usb_audio" device.bus_path = "pci-0000:00:14.0-usb-0:9.3.4:1.0" sysfs.path = "/devices/pci0000:00/0000:00:14.0/usb3/3-9/3-9.3/3-9.3.4/3-9.3.4:1.0/sound/card2" udev.id = "usb-Generic_USB_Audio_200901010001-00" device.bus = "usb" device.vendor.id = "0bda" device.vendor.name = "Realtek Semiconductor Corp." device.product.id = "402e" device.product.name = "USB Audio" device.serial = "Generic_USB_Audio_200901010001" device.string = "2" device.description = "USB Audio" module-udev-detect.discovered = "1" device.icon_name = "audio-card-usb" Profiles: input:analog-stereo: Analog Stereo Input (sinks: 0, sources: 1, priority: 65, available: yes) output:analog-stereo: Analog Stereo Output (sinks: 1, sources: 0, priority: 6500, available: yes) output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (sinks: 1, sources: 1, priority: 6565, available: yes) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: output:analog-stereo+input:analog-stereo Ports: analog-input-mic: Microphone (priority: 8700, latency offset: 0 usec) Properties: device.icon_name = "audio-input-microphone" Part of profile(s): input:analog-stereo, output:analog-stereo+input:analog-stereo analog-output-headphones: Headphones (priority: 9900, latency offset: 0 usec) Properties: device.icon_name = "audio-headphones" Part of profile(s): output:analog-stereo, output:analog-stereo+input:analog-stereo http://alsa-project.org/db/?f=616054544b8748431c354f1affbfd705c88fad37 If i run manualy speaker-test and select the proper subdevice, i get sound over the lineoutput. I cannot set it as default (or i dont know how to do it). It seems that if i am able to set the UCM2 configuration from WD15 should work. Any advise? Thank you in advance.

1 0

[PATCH] ASoC: stm32: sai: fix sai probe
by Olivier Moysan 17 Apr '20

17 Apr '20

pcm config must be set before snd_dmaengine_pcm_register() call. Fixes: 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue") Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 34a7c3d6fb91..41f01c3e639e 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1547,6 +1547,9 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) + conf = &stm32_sai_pcm_config_spdif; + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); if (ret) { if (ret != -EPROBE_DEFER) @@ -1556,15 +1559,10 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) ret = snd_soc_register_component(&pdev->dev, &stm32_component, &sai->cpu_dai_drv, 1); - if (ret) { + if (ret) snd_dmaengine_pcm_unregister(&pdev->dev); - return ret; - } - - if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) - conf = &stm32_sai_pcm_config_spdif; - return 0; + return ret; } static int stm32_sai_sub_remove(struct platform_device *pdev) -- 2.17.1

2 1

[PATCH] ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell
by Pierre-Louis Bossart 17 Apr '20

17 Apr '20

Major regressions were detected by SOF CI on CherryTrail and Broadwell: [ 25.705750] SSP2-Codec: ASoC: no backend playback stream [ 27.923378] SSP2-Codec: ASoC: no users playback at close - state This is root-caused to the introduction of the DAI capability checks with snd_soc_dai_stream_valid(). Its use in soc-pcm.c makes it a requirement for all DAIs to report at least a non-zero min_channels field. For some reason the SSP structures used for SKL+ did provide this information but legacy platforms didn't. Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/bdw.c | 16 +++++++++++++ sound/soc/sof/intel/byt.c | 48 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 64 insertions(+) diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 6c23c5769330..a32a3ef78ec5 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -567,9 +567,25 @@ static void bdw_set_mach_params(const struct snd_soc_acpi_mach *mach, static struct snd_soc_dai_driver bdw_dai[] = { { .name = "ssp0-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp1-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, }; diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index f84391294f12..29fd1d86156c 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -459,21 +459,69 @@ static void byt_set_mach_params(const struct snd_soc_acpi_mach *mach, static struct snd_soc_dai_driver byt_dai[] = { { .name = "ssp0-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp1-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp2-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + } }, { .name = "ssp3-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp4-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp5-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, }; -- 2.20.1

2 1