This keeps the streams sychronized. The packets must be interleaved per-DTS.

Add some parameters to existing function documentation.
Remove some unneeded documentation.
Convert some static function documentation to non-doxygen style.

This fixes a bug where the whole buffer was returned as decoded audio due to
*data_size not being set to zero and the return value being >= 0.

we only support decoding of mono imc

ask for a sample if not mono

Streams from RTSP or SDP that do not match an allowed type will
be skipped entirely, which allows video-only or audio-only
streaming from servers that provide both.
Signed-off-by: NMartin Storsjö <martin@martin.st>

also return AVERROR(EINVAL) instead of -1

Put the new raw GSM demuxer in its own file.
Fixes raw GSM demuxing.

calculate it from c->code_size instead.

This is an easy alternative to setting bit_rate. This patch also selects the
closest bit_rate to the requested one rather than requiring an exact value.

This should never happen, but the check avoids a divide-by-zero.

This requires some workarounds in the WAV muxer and demuxer. We need to write
the correct bits_per_coded_sample and block_align in the muxer. In the
demuxer, we cannot rely on the bits_per_coded_sample value, so we use the bit
rate and sample rate to determine the value.

This avoids having the decoder rely on AVCodecContext.bit_rate, which is not
required to be set by the user for decoding according to our API.

This also allows for not having a decoder close function.