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. _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
