daggs <daggs@...> writes: > as a design decision I've decided to split the grab > and encode to two different threads. each input > (e.g. video and audio) has it's own set of threads.
Which means that A/V sync is impossible to keep, or do I miss something? (Note that it isn't easy with alsa and xcb in any case.) > but for audio I get a whopping 939834256 which > translates into 939 mb which is unacceptable for me. 939MB per hour? Or per day? This is the maximum alsa buffer size on your system. FFmpeg uses the maximum possible alsa buffer size to make sure no audio gets lost. There is a TODO in libavdevice/alsa.c on implementing custom buffer size, consider sending a patch. > looking at the code I see that the default codec > format is AV_CODEC_ID_PCM_S16LE. Which is the codec with the second smallest bandwidth amount. (But changing the codec probably wouldn't change the buffer size, or does it?) > I was wondering if here is a way to reduce raw > capture audio frame size to more reasonable amount? If you reduce the buffer too much, you will probably miss audio from time to time. Carl Eugen _______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
