hi,
while learning to implement audio playback I noticed some formats and files will have different number of samples (frame_size or buffer_size) than my audio subsystem (jack) current setting and after decoding packets it will be annoying to keep track of remaining bits or if my frame_size bigger than the audio file while sample_rate is not an issue. the documentation of AVCodecContext.frame_size says: > decoding: may be set by some decoders to indicate constant frame size I don't get the statement, is it set automatically by Libav when opening file and we should not change it manually? or does it mean we can change it manually to our audio subsystem frame_size for convenience & it will always give us the wanted constant size? what happen to packet with frame < frame_size? does the rest of data zeroed? _______________________________________________ Libav-user mailing list [email protected] https://ffmpeg.org/mailman/listinfo/libav-user To unsubscribe, visit link above, or email [email protected] with subject "unsubscribe".
