On Tue, 13 Feb 2018, Philipp M. Scholl wrote:
The blocksize of the PCM decoder is hard-coded. This creates unnecessary delay when reading low-rate (<100Hz) streams. This creates issues when multiplexing multiple streams, since other inputs are only opened/read after a low-rate input block was completely read. This patch decreases the blocksize for low-rate inputs, so approximately a block is read every 10ms. This decreases the startup delay when multiplexing inputs with different rates.
Maybe it's just me, but I'd shoot for at least 40 ms instead of 10 ms, too small audio frames are ineffective, and some areas of the code limit the number of frames in buffers (e.g.: avfilter), or the number of packets in queues (e.g.: muxing queue), so having bigger audio frames by default makes it less probable for the user to reach these limits and get strange errors.
If you really need smaller frames than 40 ms, then I suggest add a tunable parameter for the demuxer.
Regards, Marton _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
