May 2023 - Alsa-devel - mailman.alsa-project.org

16-bit rate conversion broken with non-interleaved output
by GitHub issues - opened 14 May '23

14 May '23

alsa-project/alsa-lib issue #320 was opened from ossilator: this setup: ``` Plug PCM: Rate conversion PCM (44100, sformat=S16_LE) Converter: libspeex (external) Protocol version: 10003 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 17832 period_size : 4458 period_time : 92879 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 4458 period_event : 0 start_threshold : 17832 stop_threshold : 17832 silence_threshold: 0 silence_size : 0 boundary : 5019261784704417792 Slave: Hardware PCM card 0 'E-MU 0404b PCI [MAEM8852]' device 3 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_NONINTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 92879 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 0 ``` garbles the output rather impressively, though it's still recognizable as the input, more or less. it happens both when up-sampling and down-sampling. it doesn't happen when not re-sampling. it doesn't happen when the hardware takes 32-bit samples, so i guess it's somewhere in the optimized s16 paths. it doesn't happen with mono. this is not reproducible with the upstream emu10k1 driver, as the multichannel device demands 16 channels, which causes the route conversion plugin to be inserted, which sidesteps the issue. and the regular pcm device accepts any rate. Issue URL : https://github.com/alsa-project/alsa-lib/issues/320 Repository URL: https://github.com/alsa-project/alsa-lib

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Set C-Media USB 7.1 sound card to six_channel for surround40
by GitHub pull_request - opened 14 May '23

14 May '23

alsa-project/alsa-lib pull request #319 was opened from runderwo: This 7.1 analog sound card identifies as the following: 0d8c:0102 C-Media Electronics, Inc. CM106 Like Sound Device (lsusb) 1 [ICUSBAUDIO7D ]: USB-Audio - ICUSBAUDIO7D (/proc/asound/cards) It has the following four-channel analog stream layout: Interface 1 Altset 3 Format: S16_LE Channels: 4 Endpoint: 6 OUT (ADAPTIVE) Rates: 44100, 48000 Bits: 16 Channel map: FL FR FC LFE (/proc/asound/card1/stream0) This layout makes it impossible to play audio to the surround (SL/SR) speakers plugged into the "Surround" jack in 4 channel mode. Instead, set six_channel so that the six-channel layout will be used to access those speakers: Interface 1 Altset 4 Format: S16_LE Channels: 6 Endpoint: 6 OUT (ADAPTIVE) Rates: 44100, 48000 Bits: 16 Channel map: FL FR FC LFE SL SR Tested with speaker-test -Dsurround40:CARD=ICUSBAUDIO7D,DEV=0 -c4 The speaker system that required surround40 in order to downmix LFE into the front/rear mix is a Klipsch ProMedia v.2-400, which has 4 speakers and a subwoofer with an internal crossover, connected via front and rear stereo analog cables. Request URL : https://github.com/alsa-project/alsa-lib/pull/319 Patch URL : https://github.com/alsa-project/alsa-lib/pull/319.patch Repository URL: https://github.com/alsa-project/alsa-lib

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1.2.9: arithmetic exception in pcm/pcm_hw.c:753
by GitHub issues - opened 13 May '23

13 May '23

alsa-project/alsa-lib issue #318 was opened from engineeror: [`period_size`](https://github.com/alsa-project/alsa-lib/blob/904f0f7e7a29244c05e5874bc36486638d1f7e7f/src/pcm/pcm_hw.c#LL753C33-L753C33 ) is zero on USB DACs: https://bugs.archlinux.org/index.php?do=details&task_id=78476 @perexg Issue URL : https://github.com/alsa-project/alsa-lib/issues/318 Repository URL: https://github.com/alsa-project/alsa-lib

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Update USB-Audio.conf for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080
by GitHub pull_request - edited 13 May '23

13 May '23

alsa-project/alsa-ucm-conf pull request #315 was edited from Sunsheep: Added configuration for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Update USB-Audio.conf
by GitHub pull_request - opened 13 May '23

13 May '23

alsa-project/alsa-ucm-conf pull request #315 was opened from Sunsheep: Added configuration for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Newest Update for Fedora 37 (alsa-ucm-1.2.9-1.fc37) causes crackling with device
by GitHub issues - edited 12 May '23

12 May '23

alsa-project/alsa-ucm-conf issue #314 was edited from ClaraCrazy: First bug report here + not skilled with debugging, so please go easy on me :smile: OS: Fedora 37 Alsa-lib: 1.2.9-1 Issue: Audio crackling Audio Device: `Logic3 TX101 5.1 Soundbar` connected via `Scarlett 2i2 3rd Gen audio interface (USB)` Reverted lib back to `alsa-lib.1.2.8-2.fc37` (which in turn also downgraded alsa-ucm and alsa-utils again) and issues are fixed. I have no idea how to debug this, but would love to help in any way possible, sadly there was no issue template that would give me infos on what logs are useful / where to get those so uh.. please someone just point me in the right direction :) Thanks for your time Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/314 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] snd-hda-intel: Apply HP B&O top speaker profile to Pavilion 15
by Ryan C. Underwood 12 May '23

12 May '23

The Pavilion 15 line has B&O top speakers similar to the x360 and applying the same profile produces good sound. Without this, the sound would be tinny and underpowered without either applying model=alc295-hp-x360 or booting another OS first. Signed-off-by: Ryan Underwood <nemesis(a)icequake.net> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e103bb3693c0..7cb232a23a08 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9335,7 +9335,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8077, "HP", ALC256_FIXUP_HP_HEADSET_MIC), SND_PCI_QUIRK(0x103c, 0x8158, "HP", ALC256_FIXUP_HP_HEADSET_MIC), - SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x827f, "HP x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), -- 2.39.2 -- Ryan C. Underwood, <nemesis(a)icequake.net>

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[PATCH] ALSA: firewire-digi00x: prevent potential use after free
by Dan Carpenter 12 May '23

12 May '23

This code was supposed to return an error code if init_stream() failed, but it instead freed dg00x->rx_stream and returned success. This potentially leads to a use after free. Fixes: 9a08067ec318 ("ALSA: firewire-digi00x: support AMDTP domain") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/firewire/digi00x/digi00x-stream.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/firewire/digi00x/digi00x-stream.c b/sound/firewire/digi00x/digi00x-stream.c index a15f55b0dce3..295163bb8abb 100644 --- a/sound/firewire/digi00x/digi00x-stream.c +++ b/sound/firewire/digi00x/digi00x-stream.c @@ -259,8 +259,10 @@ int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x) return err; err = init_stream(dg00x, &dg00x->tx_stream); - if (err < 0) + if (err < 0) { destroy_stream(dg00x, &dg00x->rx_stream); + return err; + } err = amdtp_domain_init(&dg00x->domain); if (err < 0) { -- 2.39.2

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[PATCH] ASoC: SOF: topology: Fix missing error code in sof_route_load()
by Jiapeng Chong 12 May '23

12 May '23

The error code is missing in this code scenario, add the error code '-EINVAL' to the return value 'err'. sound/soc/sof/topology.c:2060 sof_route_load() warn: missing error code 'ret'. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Link: https://bugzilla.openanolis.cn/show_bug.cgi?id=4935 Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/sof/topology.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index d3d536b0a8f5..3ca3a72f1805 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2045,7 +2045,6 @@ static int sof_route_load(struct snd_soc_component *scomp, int index, if (!source_swidget) { dev_err(scomp->dev, "error: source %s not found\n", route->source); - ret = -EINVAL; goto err; } @@ -2064,7 +2063,6 @@ static int sof_route_load(struct snd_soc_component *scomp, int index, if (!sink_swidget) { dev_err(scomp->dev, "error: sink %s not found\n", route->sink); - ret = -EINVAL; goto err; } @@ -2087,6 +2085,8 @@ static int sof_route_load(struct snd_soc_component *scomp, int index, return 0; err: kfree(sroute); + ret = -EINVAL; + return ret; } -- 2.20.1.7.g153144c

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[PATCH 00/14] ALSA: emu10k1: various improvements to the DSP-based mixer code
by Oswald Buddenhagen 12 May '23

12 May '23

Oswald Buddenhagen (14): ALSA: emu10k1: make tone control switch mono ALSA: emu10k1: roll up loops in DSP setup code for Audigy ALSA: emu10k1: fix+optimize E-MU stereo capture DSP code ALSA: emu10k1: simplify snd_emu10k1_audigy_dsp_convert_32_to_2x16() ALSA: emu10k1: apply channel delay hack to all E-MU cards ALSA: emu10k1: simplify tone control switch DSP code ALSA: emu10k1: properly assert DSP init constraints ALSA: emu10k1: polish audigy GPR allocation ALSA: emu10k1: fix non-zero mixer control defaults in highres mode ALSA: emu10k1: validate min/max values of translated controls ALSA: emu10k1: omit non-applicable mixer controls for E-MU cards ALSA: emu10k1: skip mic capture PCM for cards without AC97 codec ALSA: emu10k1: enable bit-exact playback, part 1: DSP attenuation ALSA: emu10k1: enable bit-exact playback, part 2: voice attenuation Documentation/sound/cards/audigy-mixer.rst | 2 +- Documentation/sound/cards/sb-live-mixer.rst | 2 +- include/sound/emu10k1.h | 10 +- include/uapi/sound/emu10k1.h | 8 +- sound/pci/emu10k1/emu10k1.c | 8 +- sound/pci/emu10k1/emufx.c | 763 +++++++++----------- sound/pci/emu10k1/emumixer.c | 15 +- sound/pci/emu10k1/emupcm.c | 4 +- 8 files changed, 369 insertions(+), 443 deletions(-) -- 2.40.0.152.g15d061e6df

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