[FFmpeg-devel] problem with block level audio decoding on a that is not streaming

[Brett Tiplitz]

I've tried to update my code from avcodec_decode_audio4 to the new
avcodec_send_packet.  I've got a block of data to decode, so this is not a
streaming design.

1.  When I pass a big block of data to the interface, the time required to
process the buffer goes from 1.5s to 30s.
2.  When I try and pass data to the system block by block, (or bunches of
blocks), the block size has to be a multiple of the underlying packet
size.  Otherwise, the decoder discards the incomplete packet.
3.  I also tried to detect when the buffer was getting low and add data to
the buffer, but it appears that this additional data was ignored.

I tried specifically the instructions on https://blogs.gentoo.org/lu_
zero/2016/03/29/new-avcodec-api/ but this exhibits the same behavior, where
an odd block size is discarded so I loose the blocks using this exact
approach unless the block size sent is a multiple of the frame size.

I looks to be like the new method works well for streaming, but not so well
for non streaming decoding.