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

[PATCH 0/3] Tegra194 HW Fixes
by Sameer Pujar 04 May '20

04 May '20

This series proposes SW workarounds for Tegra194 HDA HW bugs. Following are the two issues seen: 1. GCAP register does not reflect true capability. The actual number of SDO lines is "4", where as it reflects "2". 2. With 4 SDO line configuration playback fails for, 44.1K/48K, 2-channel, 16-bps audio stream. After fixing [1], issue [2] is uncovered. As per recommendation by Tegra HW team the workarounds are pushed. Testing done ============ * Verify GCAP register after registering HDA sound card * Verify audio playback for 44.1K/48K, 2-channel, 16-bps. Sameer Pujar (3): ALSA: hda/tegra: correct number of SDO lines for Tegra194 ALSA: hda: add member to store ratio for stripe control ALSA: hda/tegra: workaround playback failure on Tegra194 include/sound/hdaudio.h | 3 +++ sound/hda/hdac_controller.c | 11 ++++++++++ sound/hda/hdac_stream.c | 2 +- sound/pci/hda/hda_tegra.c | 50 +++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 65 insertions(+), 1 deletion(-) -- 2.7.4

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[RFC PATCH] drm: dw-hdmi-i2s: Workaround for interchannel sample offset
by Matthias Blankertz 04 May '20

04 May '20

Add a second reset of the I2S block after the audio parameters are set. Without this, it was observed on the R-Car Gen3 platform that in ~50% of audio stream starts, the samples on each odd-numbered channel were delayed by one sample relative to the even-numbered channels: A stereo test stream was sent with a walking ones pattern, that is the data pattern was: Sample Left Right 1 0x0001 0x0001 2 0x0002 0x0002 3 0x0004 0x0004 4 0x0008 0x0008 5 0x0010 0x0010 etc. When the error occurred, the receiving device got the following data: Left Right ... 0x0004 0x0002 0x0008 0x0004 0x0010 0x0008 ... With the additional reset, the samples were always properly aligned on the receiver. Maybe this relates to the comment of commit 46cecde310bb ("drm/bridge: dw-hdmi-i2s: reset audio fifo before applying new params"), but without the datasheet it is only speculation. Signed-off-by: Matthias Blankertz <matthias.blankertz(a)cetitec.com> --- Can someone with knowledge of the hardware, or access to the datasheet give some insight if this is a "proper" fix, an ugly workaround, or the problem is actually elsewhere and this only changes behaviour due to e.g. timing changes? Thanks, Matthias drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c index d7e65c869415..0cb526f6b8d9 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -110,6 +110,14 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + /* Re-reset I2S in an attempt to fix inter-channel sample offset + * Without this second reset of the I2S interface, it was observed on + * the R-Car Gen3 platform that in ~50% of audio stream starts, the + * samples on each odd-numbered channel were delayed by one sample + * relative to the even-numbered channels. + */ + audio->write(audio->hdmi, (u8)~HDMI_MC_SWRSTZ_I2SSWRST_REQ, + HDMI_MC_SWRSTZ); return 0; } base-commit: 7111951b8d4973bda27ff663f2cf18b663d15b48 -- 2.26.2

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[PATCH] ALSA: usb-audio: add mapping for ASRock TRX40 Creator
by Andrew Oakley 04 May '20

04 May '20

This is another TRX40 based motherboard with ALC1220-VB USB-audio that requires a static mapping table. This motherboard also has a PCI device which advertises no codecs. The PCI ID is 1022:1487 and PCI SSID is 1022:d102. As this is using the AMD vendor ID, don't blacklist for now in case other boards have a working audio device with the same ssid. alsa-info.sh report for this board: http://alsa-project.org/db/?f=0a742f89066527497b77ce16bca486daccf8a70c Signed-off-by: Andrew Oakley <andrew(a)adoakley.name> --- sound/usb/mixer_maps.c | 5 +++++ sound/usb/quirks-table.h | 1 + 2 files changed, 6 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 0260c750e156..bfdc6ad52785 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -549,6 +549,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = trx40_mobo_map, .connector_map = trx40_mobo_connector_map, }, + { /* Asrock TRX40 Creator */ + .id = USB_ID(0x26ce, 0x0a01), + .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index a1df4c5b4f8c..6313c30f5c85 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3563,6 +3563,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */ ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */ ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ +ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ #undef ALC1220_VB_DESKTOP #undef USB_DEVICE_VENDOR_SPEC -- 2.24.1

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Add initial support for Realtek ALC1220 TRX40 motherboards
by GitHub pull_request - opened 03 May '20

03 May '20

alsa-project/alsa-ucm-conf pull request #25 was opened from maximlevitsky: Tested with TRX40 Designare but should work on most TRX40 motherboards Tested with pulseaudio 13.99.1 (from fedora 31 package) Tested with kernel 5.7.0-rc3 The following are the notes of the testing and implementation * Back speaker works in 5.1 mode, however pulseaudio doesn't know how to handle mixer controls with 8 channels, thus 'PlaybackMixerElem' is for now commented out This profile at least makes pulseaudio to set volume on this mixer element to 100% when it boots 7.1 mode doesn't work due to lack of jack retasking in the kernel Also pulseaudio refuses to set any postive volume on subwoofer on unknown grounds, even though on the kernel level it works. * SPDIF is not tested physically yet. The SPDIF mute mixer control is recognised by pulseaudio correctly * PlaybackPrioriry/CapturePriority seems to have no effect to pulseaudio, it doesn't autoswitch inputs when jacks are insterted. You need to switch manually * When all capture inputs are disconnected, pulseaudio thinks that the UCM2 profile is invalid and doesn't use it by default. It can be forced to use it by selecting it in the pavucontrol. Or you can for now comment out one of 'JackControl' statements, to make it think that one of the jacks is always plugged in. * Front mic is alway working in mic mode due to lack of jack retasting as well. * PulseAudio doesn't take advantage of EnableSequence/DisableSequence fully. It just enables all the devices regardless of the jack status, and it will disable them only when you swith the device profile to Off * The default volume on capture devices is set to 59% which roughtly repesents 0 dB amplification according to alsamixer Signed-off-by: Maxim Levitsky <maximlevitsky(a)gmail.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/25 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/25.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 1/2] Revert "ALSA: hda/realtek: Fix pop noise on ALC225"
by Kai-Heng Feng 03 May '20

03 May '20

This reverts commit 3b36b13d5e69d6f51ff1c55d1b404a74646c9757. Enable power save node breaks some systems with ACL225. Revert the patch and use a platform specific quirk for the original issue isntead. Fixes: 3b36b13d5e69 ("ALSA: hda/realtek: Fix pop noise on ALC225") BugLink: https://bugs.launchpad.net/bugs/1875916 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b26f0acfd86e..dedf0c17f198 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8246,8 +8246,6 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; break; case 0x10ec0225: - codec->power_save_node = 1; - /* fall through */ case 0x10ec0295: case 0x10ec0299: spec->codec_variant = ALC269_TYPE_ALC225; -- 2.17.1

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helem (MIXER, 'Master Playback Volume', 0, 1, 0) appears twice or more cannot load mixer controls: Invalid argument
by GitHub issues - edited 03 May '20

03 May '20

alsa-project/alsa-lib issue #47 was edited from igrekster: I have a PCIE Asus Sound Card (Asus Xonar AE) that uses USB interface: ``` Bus 003 Device 002: ID 0b05:180f ASUSTek Computer, Inc. ``` It's been working fine on earlier versions of kernel/alsa-lib but at some point the volume control stopped working (the playback still works fine): ``` $ alsamixer -c 1 ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.2-r1/work/alsa-lib-1.2.2/src/mixer/simple_none.c:1544:(simple_add1) helem (MIXER,'Master Playback Volume',0,1,0) appears twice or more cannot load mixer controls: Invalid argument ``` This is on kernel 5.4.37/alsa 1.2.2. Unfortunately I cannot downgrade to the point when it was working to pinpoint the issue. I tried kernels 5.3.11 and alsa 1.1.8. I've tried building the latest alsa-lib, and it is still showing the error. I'm happy to try patches to alsa or kernel modules to help debug this. ``` $ amixer -c 1 controls numid=7,iface=CARD,name='Clock Source 18 Validity' numid=10,iface=CARD,name='Clock Source 19 Validity' numid=17,iface=CARD,name='Clock Source 22 Validity' numid=11,iface=CARD,name='IEC958 In - Output Jack' numid=18,iface=CARD,name='Keep Interface' numid=14,iface=CARD,name='Line - Input Jack' numid=12,iface=CARD,name='Mic - Input Jack' numid=8,iface=CARD,name='Speaker - Output Jack' numid=20,iface=MIXER,name='Master Playback Volume' numid=19,iface=MIXER,name='Master Playback Volume',device=1 numid=5,iface=MIXER,name='PCM Playback Switch' numid=9,iface=MIXER,name='PCM Playback Switch',index=1 numid=16,iface=MIXER,name='PCM Capture Source' numid=15,iface=MIXER,name='Line Capture Switch' numid=13,iface=MIXER,name='Mic Capture Switch' numid=6,iface=MIXER,name='Input Gain Pad Control' numid=4,iface=PCM,name='Capture Channel Map' numid=1,iface=PCM,name='Playback Channel Map' numid=2,iface=PCM,name='Playback Channel Map',device=1 numid=3,iface=PCM,name='Playback Channel Map',device=2 ``` `amixer -c scontrols` fails withe the same error as `alsamixer`. A few more details: [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/4569764/alsa-… [proc-asound.txt](https://github.com/alsa-project/alsa-lib/files/4569766/pro… Issue URL : https://github.com/alsa-project/alsa-lib/issues/47 Repository URL: https://github.com/alsa-project/alsa-lib

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helem (MIXER, 'Master Playback Volume', 0, 1, 0) appears twice or more cannot load mixer controls: Invalid argument
by GitHub issues - opened 03 May '20

03 May '20

alsa-project/alsa-lib issue #47 was opened from igrekster: I have a PCIE Asus Sound Card (Asus Xonar AE) that uses USB interface: ``` Bus 003 Device 002: ID 0b05:180f ASUSTek Computer, Inc. ``` It's been working fine on earlier versions of kernel/alsa-lib but at some point the volume control stopped working (the playback still works fine): ``` $ alsamixer -c 1 ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.2-r1/work/alsa-lib-1.2.2/src/mixer/simple_none.c:1544:(simple_add1) helem (MIXER,'Master Playback Volume',0,1,0) appears twice or more cannot load mixer controls: Invalid argument ``` This is on kernel 5.4.37/alsa 1.2.2. Unfortunately I cannot downgrade to the point when it was working to pinpoint the issue. I tried kernels 5.3.11 and alsa 1.1.8. I've tried building the latest alsa-lib, and it is still showing the error. I'm happy to try patches to alsa or kernel modules to help debug this. A few more details: [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/4569764/alsa-… [proc-asound.txt](https://github.com/alsa-project/alsa-lib/files/4569766/pro… Issue URL : https://github.com/alsa-project/alsa-lib/issues/47 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ALSA: firewire-lib: fix 'function sizeof not defined' error of tracepoints format
by Takashi Sakamoto 03 May '20

03 May '20

The snd-firewire-lib.ko has 'amdtp-packet' event of tracepoints. Current printk format for the event includes 'sizeof(u8)' macro expected to be extended in compilation time. However, this is not done. As a result, perf tools cannot parse the event for printing: $ mount -l -t debugfs debugfs on /sys/kernel/debug type debugfs (rw,nosuid,nodev,noexec,relatime) $ cat /sys/kernel/debug/tracing/events/snd_firewire_lib/amdtp_packet/format ... print fmt: "%02u %04u %04x %04x %02d %03u %02u %03u %02u %01u %02u %s", REC->second, REC->cycle, REC->src, REC->dest, REC->channel, REC->payload_quadlets, REC->data_blocks, REC->data_block_counter, REC->packet_index, REC->irq, REC->index, __print_array(__get_dynamic_array(cip_header), __get_dynamic_array_len(cip_header), sizeof(u8)) $ sudo perf record -e snd_firewire_lib:amdtp_packet [snd_firewire_lib:amdtp_packet] function sizeof not defined Error: expected type 5 but read 0 This commit fixes it by obsoleting the macro with actual size. Cc: <stable(a)vger.kernel.org> Fixes: bde2bbdb307a: ("ALSA: firewire-lib: use dynamic array for CIP header of tracing events") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream-trace.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h index 16c7f6605511..26e7cb555d3c 100644 --- a/sound/firewire/amdtp-stream-trace.h +++ b/sound/firewire/amdtp-stream-trace.h @@ -66,8 +66,7 @@ TRACE_EVENT(amdtp_packet, __entry->irq, __entry->index, __print_array(__get_dynamic_array(cip_header), - __get_dynamic_array_len(cip_header), - sizeof(u8))) + __get_dynamic_array_len(cip_header), 1)) ); #endif -- 2.25.1

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missing snd_pcm_drain() in alsa-lib/test/pcm_min.c
by GitHub issues - opened 03 May '20

03 May '20

alsa-project/alsa-lib issue #46 was opened from robert-rozee: hi, the example pcm_min.c located in alsa-lib/test/ is missing a "snd_pcm_drain(handle);" before the handle is closed. while the example works as is, the lack of calling ...drain before ...close means that the end of the sample being played is truncated. when expanding the example to play actual content (rather than random noise) this becomes an issue. i took this example and used it as the basis of my own routine that played a sine wave tone (with the volume feathered at the start and end). many hours were spent trying to work around this problem of truncated output, before one of the developers pointed out that i needed to drain before closing. as i am a beginner with ALSA and working from scant information, i didn't know that snd_pcm_drain() even existed. cheers, rob :-) Issue URL : https://github.com/alsa-project/alsa-lib/issues/46 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 0/2] ALSA: line6: hwdep: add support for poll and non-blocking read
by Vasily Khoruzhick 02 May '20

02 May '20

This series adds support for polling and non-blocking read for hwdep interface. This allows apps to listen to HW events without using busy loop. Example of app that uses hwdep interface for POD HD500 can be found here: https://github.com/anarsoul/line6_hwdep_test Vasily Khoruzhick (2): ALSA: line6: hwdep: add support for O_NONBLOCK opening mode ALSA: line6: Add poll callback for hwdep sound/usb/line6/driver.c | 20 ++++++++++++++++++++ sound/usb/line6/driver.h | 1 + 2 files changed, 21 insertions(+) -- 2.26.2

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