On 20-08-2018 03:27 AM, Marvin Scholz wrote:
+@example +$ ffmpeg -i input -map 0:a -af loudnorm=I=-23:TP=-1:print_format=summary -f null - + +[...] +Input Integrated: -9.0 LUFS +Input True Peak: +1.5 dBTP +Input LRA: 9.4 LU +Input Threshold: -19.5 LUFS +@end example + +Then pass the input measurements to the next run that produces the result: + +@example +ffmpeg -i input -af loudnorm=I=-23:TP=-1:measured_I=-9.0:measured_TP=1.5:measured_LRA=9.4:measured_thresh=-19.5:print_format=summary output
a) Your first pass command will run the filter for all audio streams in the input, but your 2nd pass command won't. Suggest you remove the mapping.
b) Except for a particular set of conditions*, loudnorm will resample the audio to 192 kHz. ffmpeg may then resample the result to the highest supported rate by the encoder. For native AAC, this will be 96 kHz; for PCM, it will remain to 192kHz. Some (many ?) players, especially web ones, don't support these sampling rates. It's advised to insert aresample afterwards e.g. aresample=48000
*if all of these conditions are met: linear is true and enabled; target LRA is equal or greater than input LRA; target TP is equal or greater than input LRA after adjusting for integrated loudness change.
Gyan _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
