Greetings Carl, > Sent: Wednesday, May 13, 2015 at 11:46 AM > From: "Carl Eugen Hoyos" <[email protected]> > To: [email protected] > Subject: Re: [Libav-user] setting raw alsa pkt size > > 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.) > well I think it is doable as I've found a project named tp-screencasting-teaching-system which allows broadcasting a lesson over the net.
> > 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. > 939MB per AVFrame. > > 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?) > not sure, that is why I've asked. > > 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. > I've assumed that but I'm failing to see how it is possible that a single raw audio frame will require so much space. _______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
