[MPlayer-dev-eng] [BUG] surround audio plugin broken

[Arpi]

Hi,

> > > Disagree, resampling when the movie is 48000 hz and the soundcard only
> > > supports 44100 should be done by default, just like -vop scale is
> > > auto-inserted when the output device is rgb. Otherwise the output is
> > > just wrong.
> > 
> > yep. but for example automatic resampling of 44100 to 44098 is just dumb.
> > (and slow and causes big delay and whatever)
> 
> Hmm, perhaps the ao drivers should be modified to report the requested
> samperate back to libaf (rather than the actual one) if the two only
> differ by a very small amount. Sound good?

not sounds good (it ins;'t teh job of ao drivers, and it means adding teh
same code to every single ao_*.c), but it seems the only way now.

actually we should extend the libao2 api by query_format()-like function,
and the same for libaf too, and then codecs-filters-ao could handshake teh
best data exchange format, just like it's done for video now in libmpcodecs.
unfortunatelly(?) libaf was designed to be mplayer-independent, so it isn't
easy to add now.

the current method is a bit broken/lazy:
- codecs has single output dataformat (maybe depends on input file's
  headers, maybe constant per codec)
- codecs tries to produce N channel output where N is set by opt. -channels
  (afaik only liba52 supports it, where the codec can do the downsampling)
- codecs always produce integer (or compressed, for ac3 passthru) output,
  signed or unsigned, 8 or 16 bits. if teh codec's internal format is float,
  the codec (or ad_*.c wrapper) converts it to integer (see vorbis or faad).
  (liba52 is similar, but there are optimized float->int helped by the
   decoding process so it's better than decoding-to-float->float-to-int)
- now mplayer goes to libao2's config() and ask it for the
  format,channels,rate got from the codec
- the selected ao driver tries to setup output device to the requested
  parameters, if fails then it tries to get something similar config, and
  updates the parameters in ao_data struct.
- now mplayer asks libaf to convert from codec's output fmt+chans+rate to
  the one returned by libao2 in ao_data. it will auto-insert filters.

maybe we should change all codecs to output in their native formats only,
and left all kind of downsampling, float2int etc to libaf.
this way we lose conversions helped by the codecs (examples: liba52's
float2int, channels downsampling, mp3lib's half or quarter freq decoding,
mono decoding of stereo sterams, stereo decoding of mono streams).
but this way we could ask the soundcard to accept the codec's native format,
and do the conversion only if it cannot be done. if both codecs & ao
supports float, then prefer float in filters, otherwise if both codec & ao
are integer, then don't convert it to & back from float, it just eats cpu.
if the user wants hi-fi quality, he can still force float filters on
commandline.

the best would be if codecs could support multiple output formats, rates,
channels, and libaf could handle the handshaking between the selected (by
user) filters, codec and ao. then auto-insert converters where it's needed.
but it requires ig api changes and will result in many problems, see -vop :)


A'rpi / Astral & ESP-team

--
Developer of MPlayer, the Movie Player for Linux - http://www.MPlayerHQ.hu