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I have a couple of mp3s that are stereo but one of the channels is silent, quite annoying to listen to, especially in earphones. I can't get new recordings (the tracks are bootlegs with unknown origin) so I the second best option I guess would be to convert them to mono replacing the silent track with the other track.

I read this similar question Convert mp3 from mono to stereo using Lame but the solutions suggested does not seem to apply to my situation.

What tool can replace one track with the other in my situation, preferably lossless and preferably using "joint stereo" (= the new channel will in practice be empty, avoiding increasing the size of the files in a significant way)?

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  • Why don't they apply?
    – Ƭᴇcʜιᴇ007
    Commented Nov 16, 2014 at 21:37
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    Wouldnt every device you have play a mono signal on both channels? Given files of that type I would prefer to use the one channel to make a mono file, not to double the data quantity? Converting a stereo to mono without selecting the track itself , would mix the 2 in most programs, again creating unnessisary work, and mixing a dead and possibly noisey track with the one good one, and would/could half the original track and require de-compression re-compression due to mixing cannot be done when compressed.
    – Psycogeek
    Commented Nov 16, 2014 at 22:06
  • You misunderstand. The files are stereo but one of the channels are silent.
    – d-b
    Commented Nov 17, 2014 at 9:46
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    Apparently silent & digitally silent are not necessarily the same thing. Psychogeek has the correct approach, Polyergic's answer has the method to employ. Using a combination of both is your best solution. Your resulting file can easily be mono, which will then play equally from both speakers, without any possibility of adding noise or suffering phase discrepancy.
    – Tetsujin
    Commented Nov 17, 2014 at 11:26

3 Answers 3

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You can do this with ffmpeg. Their Audio Channel Manipulation page has many examples, including this example to split audio channels into separate files:

ffmpeg -i stereo.wav -map_channel 0.0.0 left.wav -map_channel 0.0.1 right.wav

It will do the same with any format it understands, which includes mp3. The example splits one stereo file into two mono files, without any reencoding. I assume all players will (by default) play the one channel to each speaker. You can follow this example to make a stereo file with the same audio copied into both channels. To get joint stereo you have to reencode, which (on mp3) is not lossless; that page does not have an example.

If you're doing this on Windows (and not using Cygwin), it looks like the best source is the Zeranoe FFmpeg builds

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  • I think anything suggested on that page will involve reencoding.
    – mwfearnley
    Commented Aug 25 at 19:31
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So I tried the ffmpeg method and I get the impression that it is reencoding. It takes quite a while to process, uses up a lot of cpu (both way more than MP3directcut); the resulting files show a different codec in Mediainfo's readout; and when comparing spectrograms from original and split files there are marked differences. Just to check I transcoded the original with the same encoding settings (lame 128kpbs cbr stereo mp3 -> lame 128 cbr stereo mp3) and the original was much closer to the transcode than to the split files from ffmpg.

I really don't think this method is lossless. You might as well use a program like Audacity.

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I can't comment ShadSterling's nor Jayden's answers (not enough rep), so here's my 2 cents.

This command does indeed seem to reencode (in my case, left.wav and right.wav were in Vorbis, while stereo.wav was aac), as Jayden hinted.

ffmpeg -i stereo.wav -map_channel 0.0.0 left.wav -map_channel 0.0.1 right.wav

However, this command, taken from the same examples page, does seem (for me the resulting file was in the same codec as the source - AAC - but this not a complete proof that no reencoding happened) to do the job (no reencoding) :

ffmpeg -i stereo.wav -af "pan=mono|c0=c1" mono.m4a
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  • Adding the option -acodec copy says: Filtergraph 'pan=mono|c0=FL' was defined for audio output stream 0:0 but codec copy was selected. Filtering and streamcopy cannot be used together. So I assume the encoding is not lossless.
    – Frank Breitling
    Commented Feb 19, 2023 at 23:12
  • I think it would be easier for ffmpeg to decode and reencode into the same codec, than to losslessly manipulate the command with an -afcommand (which would be written to not depend on any particular codec). Also, I don't think genuine .wav files can contain an AAC codec?
    – mwfearnley
    Commented Aug 25 at 19:43

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