I am trying to use libavformat to write AAC-LC frames (libfdk-aac) to MP4 (or M4A) files. While it mostly works, playing back using standard players results in poor quality and slightly faster playback. One observation from ffprobe is the pkt_duration and nb_samples values are inconsistent for the frame dump. For a 16 kHz sample rate input to the encoder, the pkt_duration indicates 1280 while the nb_samples is 1024. The pts related values are consistent with the pkt_duration values reflecting 80 msec frames. Does this make sense? Would this result in the playback issues I am experiencing? Looking at a good file recorded using iTunes, the pkt_duration and nb_samples are consistent. The first frame contains 960 samples while the remaining ones contain 1024 (44.1 kHz sampling).
Thanks, Bob
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