Bonjour,
I want to add an audio track to a film. I have a film in english running at 23.976fps and the same film in french running at 25fps.
I extracted the french audio track from the french film and using mkvtoolnix, I added this track to the english film. For audio tracks, there is field "stretch by" where I put 1.0427 which converts the audio from 25fps to 23.976fps.
This does the job and the audio is correctly synchronised with the video *but* the sound has a lot of micro-cuts which are unpleasant to hear.
Is there a way to avoid this phenomenon?
Another question: at some places the audio is desynchronised from the video, I think because some images are missing due to a difference between the English and French version, is there a software to use to correct these problems (similar to those softwares to synchronise the subtitles like aegisub for instance)?
I am working under linux.
Thank you for attention.
F.P.
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This is expected since mkvmerge doesn't change the audio, so you get overflow/underflow on playback (unless the player is exceptionally smart). Alternative is to use an audio tool to resample the audio, or better, change the video frame rate to 25fps.
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Thank you for your answer... So I will know that I can't use mkvtoolnix to modify the audio tracks, in spite of the fact that it pretends to be able to modify the speed!
I found another solution : audacity effects--->change speed. Changing the speed from 25fps to 23.976fps is harmless on tempo and pitch. Maybe this cannot be used for greater differences...
F.P. -
Chances are even that slowing down will get you the original pitch back. It would depend on the native recording speed of course. It's not uncommon for 25fps video to have been sped-up from 24 fps with pitch altered audio.
Of course, if the source has been recorded at 25 fps, the option Luke M suggested is the way to go.
Another thing maybe is output format. Your source's audio-codec might not be available (for free). Resampling will take place in (decoded) pcm. This process is not a strict lossless one. To avoid further "quality loss" introduced by a lossy encoder, you could consider outputting as uncompressed pcm or lossless flac.
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