Rephrasing the requirement.
We have an audio pipeline which redacts the audios using the set of start
time and end time provided from our ML team. To achieve this, we execute
the ffmpeg command with filter
"volume=enable='between(t,starttime,endtime)':volume=0".
For some cases, we are receiving audio files with codecs which are causing
the outputs to have large file sizes. One such example is below,
ffprobe and ffmpeg output of the file is attached to the mail.
*command:* ffmpeg -i call.wav -af "volume=enable='between(t,0,1)':volume=0"
output.wav
*behaviour: *executed within a second, input file size 2.1 mb, output file
size 176 mb
As we can see here, the output file size is 88 times the original. The
objective here is to achieve the output file size of less than 2-3 times
the original without reducing the sound quality.
I went through the documentation but couldn't find any suitable way to fix
it. Let me know if anyone knows the issue in my command. Please point me to
some resource containing the details of audio codecs and filtering.
Regards,
Shubham
The input file is a 19 kb/s highly compressed audio file (opus codec
lossy format variable bitrate)
The generated wav output is in pcm 1536 kb/s uncompressed cbr.
so 1536 / 19 = 80.84... times bigger output (at least) so your 88 times
bigger is expected going from highly compressed variable bitrate audio
to uncompressed constant bitrate audio.
You should not convert the audio, just use a universal container like
matroska that can contain almost any existing codec.
Try this command:
ffmpeg -i call.wav -af "volume=enable='between(t,0,1)':volume=0"
-c:a copy output.mka
Learn about codecs, lossy and lossless encoding and differnt containers
to know what is going on.
maybe this can help you learn:
http://ce.sharif.edu/courses/91-92/2/ce873-1/resources/root/Class%20Notes/MMN-lec3-Audio-911121.pdf
Greetings
Ferdi
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