Thanks for all the inputs, I have a better understanding of the problem now. The input audio is highly compressed. While applying the filter, ffmpeg is converting the audio based on the fileformat of the output file. If the output file provided is wav format, the resulting file is 1536 kb/s uncompressed cbr, hence the large file size. If the output file provided is of mp3, mka or opus(same as the input file), then conversion of the resulting file is taking longer.
Does ffmpeg always convert the audio file while running the filter? if not, in what cases does it skip the conversion? The things I tried based on the inputs: 1. ffmpeg -i call.wav -af "volume=enable='between(t,0,1)':volume=0" -c:a copy output.mka - does not work, error: filtering and streamcopy cannot be used together. 2. ffmpeg -i call.wav -af "volume=enable='between(t,0,1)':volume=0" -c:a libopus output.mka - In this case conversion is happening. resulting file size is 10mb(under expected range) but the execution time is high due to conversion PS: I am having discussions with the team about hiring a specialist. will keep posted about it. Regards, Shubham _______________________________________________ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
