On 2/12/16, Moritz Barsnick <barsn...@gmx.net> wrote: > On Fri, Feb 12, 2016 at 11:02:32 +0100, Paul B Mahol wrote: >> ffmpeg -f lavfi -i >> anoisesrc=sample_rate=16000:duration=35:nb_samples=16000 -map 0:a -c:a >> pcm_s16le -ac 1 -f segment -segment_time 5 -segment_format s16le >> out.%03d.raw > > Well, anoisesrc was just an for creating arbitrary input. The original > poster may not be able to influence the sample rate and number of > samples in his input. Therefore I sought after other flags to influence > those. > >> You can't resample after, you will get different number of samples >> per packet. > > So, in the real world: Would you recommend inserting the aresample > filter? (And possibly something for downmixing to mono.) > > My head is spinning. Samples, packets, frames. How do they relate in > audio? Especially regarding this codec. > > I would have thought that, in the path from codec to muxer, the concept > of frames would be somewhat broken down, and (for this particular > codec) there would be a frame per sample. Or is it a packet per sample? > Or neither of the two? > > Anyway, I would have thought that this codec, being "key frame only", > could and would be cut at exact points, assuming the sample rate allows > an exact number of samples to fit into the given time interval. > > Carl Eugen's response was better than both of mine: Just try to solve > the problem, don't try to understand all the mechanisms. ;-)
Use aresample=16000,asetnsamples=16000 and it should work _______________________________________________ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
