Hello, I am currently working on converting a large number of files from wav (Max freq 64000 Hz) to mp3 (max Freq 48000 Hz).
My current call is: ffmpeg -i "INPUTFILENAME.WAV" -af "aresample=resampler=soxr:precision=33" -q:a 2 -ar 48000 -first_pts 0 "OUTPUTFILENAME.mp3" Also without the "first_pts 0" call I was noticing that there was a taper of the waveform amplitude at the start and end of the output files. I want to use the full dynamic range of the file and each file has a varying maximum sound levels. I have investigated ffmpeg-normalize however it seem to cause artifacts in the mp3 files over the last 3 seconds where the file is repetitively clipped in 0.1 second intervals. Normalizing a file to the maximum recorded level seems like a trivial thing to do with ffmpeg... am I missing something obvious. Any ideas or assistance are welcome. _______________________________________________ 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".
