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I'm not sure if my question makes sense, but here it is the problem I have.

I have a bunch of wave (WAV) files that I believe were converted from MP3 or some other format that I have no information about.

These wave files are 44,100Hz but when I listen to them, they're clearly not! They sound very muffed and I could guess that their source was no more than 16,000Hz.

What I wanted to do is run these files through a tool (maybe FFmpeg?) that would decrease the sample rate of these files to the minimum needed to make them sound exactly how they already do, without loosing more quality. It will save disk space as I have hundreds of files and also it will allow me to process them on another tool that requires WAV files as input.

Does it make sense? Is this possible?

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    @Giacomo1968 yes, that's basically what I need. I don't want to reencode them to other formats because I pass them to another tool that expects only raw pcm data... I'll see if those urls help. Thanks!
    – user17952421
    Commented Mar 1 at 2:03
  • If you check on a spectrograph & the files' audio spectrum ends hard at about 8KHz, that's sufficient evidence for a 16KHz sampling frequency.
    – Tetsujin
    Commented Mar 1 at 7:15

1 Answer 1

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When you have an MP3 file, a lower bitrate will cause audio to be muffled, even if that MP3 is in 44khz or higher.

Once an MP3 is converted to WAV, you lose all the information you had in the MP3, and for that reason, it is not possible to convert it to a lower standard that automatically detects the best setting and uses that.

You can batch convert all WAV files to a lower standard still, but you will have to manually target a lower bitrate such as 22khz or lower, and if you set it too low, you will lose more quality.

Do keep in mind that downsampling like this can cause the audio to become more muffled even if you get the bitrate correct due to rounding errors. Ideally you would not want to touch this any more than you have to.

WAV is a raw format, but if you want the optimal quality with the lowest filesize, then convert these WAV files to FLAC, as the conversion to FLAC will automatically detect what the best setting is to create a perfect recreation without any compression.

It will detect that no frequencies above 16khz were used and then rebuild the wavefile accordingly, resulting in a lower filesize.

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  • I just realized I used the word bitrate wrong in this answer. For MP3, the bitrate refers to the encoding bitrate, eg. 320kbps, whereas with wav I mean the 44khz one.
    – LPChip
    Commented Mar 4 at 7:30

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