Hi,
I'm continuing to investigate that issue and did some tests with aplay, using the 5.1 wav file that was used to create the AAC file with I had trouble with. It turns out that aplay also mute or lower some channels and do not play 5.1 files well. I check the available PCM with aplay and got the following result :
[code]bash-4.2$ aplay -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=SB HDA ATI SB, ALC889A Analog Default Audio Device sysdefault:CARD=SB HDA ATI SB, ALC889A Analog Default Audio Device front:CARD=SB,DEV=0 HDA ATI SB, ALC889A Analog Front speakers surround40:CARD=SB,DEV=0 HDA ATI SB, ALC889A Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=SB,DEV=0 HDA ATI SB, ALC889A Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=SB,DEV=0 HDA ATI SB, ALC889A Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=SB,DEV=0 HDA ATI SB, ALC889A Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=SB,DEV=0 HDA ATI SB, ALC889A Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=SB,DEV=0 HDA ATI SB, ALC889A Digital IEC958 (S/PDIF) Digital Audio Output hdmi:CARD=HDMI,DEV=0 HDA ATI HDMI, HDMI 0 HDMI Audio Output[/code]
Using the default, I get sound but not properly downmixed :
bash-4.2$ aplay -v -D default ndj_5_1_MIX.wav Lecture WAVE 'ndj_5_1_MIX.wav' : Signed 24 bit Little Endian in 3bytes, Fréquence 48000 Hz, 6 Canaux Plug PCM: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0 1 <- 1 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S24_3LE subformat : STD channels : 6 rate : 48000 exact rate : 48000 (48000/1) msbits : 24 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 Slave: Soft volume PCM Control: PCM Playback Volume min_dB: -51 max_dB: 0 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 Hardware PCM card 0 'HDA ATI SB' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 4611686018427387904 silence_threshold: 0 silence_size : 4611686018427387904 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 0
So I thought maybe I should force stereo by using the "front" PCM. But I got this :
bash-4.2$ aplay -v -D front ndj_5_1_MIX.wav Lecture WAVE 'ndj_5_1_MIX.wav' : Signed 24 bit Little Endian in 3bytes, Fréquence 48000 Hz, 6 Canaux aplay: set_params:1233: Ce format d'échantillonage n'est pas disponible Available formats: - S16_LE - S32_LE
And the same goes for all non-Default PCM. So the Default PCM does not downmix properly and the others do not work. Is there any way around that?
Best regards,
Richard Van Den Boom
Hi,
I'm running slackware-current on my main desktop, which uses Alsa directly for sound, there is no sound server like PulseAudio or Jack. I use VLC as a principal multimedia player and until about 2 months ago, everything worked fine : Slackware was running alsa 1.0.26 and I could use whatever version of VLC, even git, with no issue. In may, Slackware-current upgraded to 1.0.27 alsa and then upgraded to the bugfix .1 then .2 releases. Since then, any newer version of VLC that I try (self compiled from git) exhibit the same wrong behaviour : every time I watch a film with 5.1 surround sound, only two channels are played, and I often miss the voices, for instance.
I raised the issue on the VLC forum and they pointed me to the following logs in the VLC output :
Hardware PCM card 0 'HDA ATI SB' device 0 subdevice 0
followed by :
Hardware PCM card 0 'HDA ATI SB' device 0 subdevice 0
The previous output from working VLC showed a downmixing line :
BUFFER_BYTES: [1024 655360000]
According to the VLC developers, channel mapping support, introduced by alsa 1.0.27, was added to VLC in december 2012, and this must have been kicking in since Slackware added alsa 1.0.27. Their analysis is that, although the hardware seem only to support two channels, the softvol plugin is reporting 5.1 support and advertising it. Thus VLC automatically choose the 5.1 output and do not downmix it. However, although the chip (Realtek ALC889A) indeed support 5.1 sound, and although I indeed have several audio port on my mainboard, I only plugged stereo speakers and thus got a problem. I tried setting up force downmix in /etc/asound.conf like this :
pcm.!surround51 { type vdownmix slave.pcm "default" } pcm.!surround40 { type vdownmix slave.pcm "default" }
But to no avail. Is this a bug I should report or is there a proper configuration that must be used? I can provide a alsa-info output if necessary.
Best regards,
Richard Van Den Boom
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