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. ;-) Moritz _______________________________________________ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
