[MPlayer-users] a few notes on AC3 passthrough in general (was SBLive AC3...)

[Aubin Paul]

On Tue, Feb 19, 2002 at 02:42:53PM +0100, Marcus Blomenkamp wrote:
> The sample format of the data must be 2channel 16bit (little endian 
> IIRC). Samplerates are 48kHz - although my receiver also accepts 
> 44100Hz. I do not know if this is due to an over-compatability of my 
> receiver or if 44100 is also possible in the ac3 specs. For safety's 
> sake lets keep this at 48000Hz. AC3 data chunks are inserted into the 
> stream every 0x1600 bytes (don't bite me on that, look into 
> 'ac3-iec958.c': 'ac3_iec958_build_burst').

I know the AC3 data I'm playing is 48khz/16bit/2channel, and my
decoder doesn't seem to have a problem with that. I can dump the audio
stream and play it back manually with AC3 playing software.

> 1) Some OSS guy invented the format AFMT_AC3. Soundcard drivers 
> implementing this format should therefore adjust it's mixers and 
> switches to produce the desired output. Unfortunately some soundcard 
> drivers do not support this format correctly and most do not even 
> support it at all (including ALSA).

I've verified that mplayer sets the format correctly via the ioctl,
but what is appearing to happen is that the video loses all semblance
of sync.

> I do have two soundcards with digital IO facilities (CMI8738 and 
> Trident4DWaveNX based) plus the mentioned decoder. I'm currently running 
> Linux-2.4.17. Following configurations are happily running here:

I actually considered picking up the C-Media boards, simply for the
fact that they're cheap, and appear to be supported fully. I only
didn't because of the sound.html suggesting that there are problems
with the card; can you tell me how well it worked for you? Any sound
glitches or noise on AC3 or pcm data?

Thanks for your help; I'm hoping someone can shed some light on the
A-V sync issues...

Aubin