This makes the wav and pcm demuxer demux bigger packets, which is more
efficient.
As a side effect of the bigger packets, audio durations can become less exact
for command lines such as "ffmpeg -i $INPUT -c:a copy -t 1.0 $OUTPUT".
Signed-off-by: Marton Balint <cus@passwd.hu>
The wav demuxer by default tried to demux 4096-byte packets which caused
packets with very few number of samples for files with high channel count.
This caused a significant overhead especially since the latest ffmpeg.c
threading changes.
So let's use a similar approach for selecting audio frame size which is already
used in the PCM demuxer, which is to read 25 times per second but at most 1024
samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
This should be more useful for users since numerical values for channel
layout can be confusing and unintuitive.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Also, make every addition except for sidedata part of version 1 instead of the
new version 2.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Avoids resampling and channel mixing. This only tests the behavior
with respect to input and output audio rather than also testing changes
to the encoder or muxer that do not affect the resulting decoded output.
