Wheezy and alsa: [ac3] Specified sample_fmt is not supported

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Wheezy and alsa: [ac3] Specified sample_fmt is not supported

Postby soeiro » 2011-09-18 19:53

Folks

I had a working configuration and during the last few weeks something has broken. I use an alsa filter to convert a 5.1 audio stream to AC3 so that it can be sent to my hometheater system (the only other option would be by HDMI, but the FOSS driver doesn't support more then 2 audio channels yet).

My .asoundrc configuration is:

Code: Select all
pcm.a52encode {
        type a52
        format S16_LE
        channels 6
        rate 48000
        bitrate 448
        card "Intel"
}

pcm.hometheater {
        type plug
        slave.pcm "a52encode"
        slave.channels 6
}


It used to work, but now whenever I try to send audio to the "hometheater" device this happens:
Code: Select all
$ speaker-test -Dhometheater -c6 -twav
speaker-test 1.0.24.2

Playback device is hometheater
Stream parameters are 48000Hz, S16_LE, 6 channels
WAV file(s)
Rate set to 48000Hz (requested 48000Hz)
Buffer size range from 3072 to 15360
Period size range from 1536 to 1536
Using max buffer size 15360
Periods = 4
[ac3 @ 0x1c14000] Specified sample_fmt is not supported.
Unable to set hw params for playback: invalid argument                                                                             
Setting of hwparams failed: invalid argument


By using google I've found found out the something in the libavcodec has changed and now it only expects float values. However, I can't seem to do anything about it. So, does anyone know how to create a .asoundrc filter that would convert from S16_LE format to whatever float that is needed by libavcodec? Or any other solutions to send a 5.1 audio stream to a receiver using either spdif or hdmi?

Thanks,
Luis
soeiro
 
Posts: 17
Joined: 2010-08-23 22:33

Re: Wheezy and alsa: [ac3] Specified sample_fmt is not suppo

Postby soeiro » 2011-09-20 19:23

Just adding more informaton.

I've found the problem also reported here:

http://www.ffmpeg.org/trac/ffmpeg/ticket/363

It seems that the libasound module wasn't sending the sample format to libavcodec. The article says that the problem should be fixed, but I didn't find a new libasound in debian yet. Maybe I misunderstoodsomething...Does anybody have any clue?

Thanks
soeiro
 
Posts: 17
Joined: 2010-08-23 22:33


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