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

[PATCH AUTOSEL 5.6 65/73] ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tablet
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit c8b78f24c1247b7bd0882885c672d9dec5800bc6 ] The MPMAN MPWIN895CL tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. 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/20200405133726.24154-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 6bd9ae813be28..d14d5f7db1689 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -591,6 +591,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* MPMAN MPWIN895CL */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MPMAN"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "MPWIN8900CL"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), -- 2.20.1

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[PATCH AUTOSEL 5.6 63/73] ALSA: hda: Fix potential access overflow in beep helper
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0ad3f0b384d58f3bd1f4fb87d0af5b8f6866f41a ] The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") 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/20200407084402.25589-2-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_beep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index f5fd62ed4df5b..841523f6b88de 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -290,8 +290,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } -- 2.20.1

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[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

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[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

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[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

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[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

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[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

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[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

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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

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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.

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