[ardour-users] sync imported wav files

Petter Sundlöf petter.sundlof at findus.dhs.org
Tue Dec 6 10:44:01 PST 2005

But that won't work/will be a lot of work if there are small hickups 
inside the regions, that are not caused by clock differences.

Paul Winkler wrote:
> On Tue, Dec 06, 2005 at 04:24:19PM +0100, Petter Sundl??f wrote:
>>I've done this exact thing, lined up a Microphone->MD recording with a 
>>mixer desk>MD recording. Fortunately I had to do very little fixing. I 
>>just zoomed in very close, and aligned them at the start. Then look for 
>>where I could start to hear the out-of-sync-ness, and aligned it there.
> another thing you could do - this is a bit of work, but it should
> avoid audible phase issues *and* save you some manual realignment
> work if the recording is long. I've done things like this before - let
> me see if I can remember the general procedure:
> - Import both files
> - Line up as precisely as you can at the start
>   (try to find a good strong peak that's obviously the same thing
>   in both files).
> - Go to the end, zoom in very close, find some strong peak in both
>   tracks (it will be out of sync of course).
> - Calculate the number of frames between the beginning peak and
>   the ending peak in the desk recording. Call this "d".
> - calculate the number of frames between the beginning peak and
>   the ending peak in the audience recording; call this "a".
> - Calculate the ratio of the two lengths: ratio = d / a
> - use sndfile-resample (comes with libsamplerate) on your
>   audience recording, like so:
>   sndfile-resample -by <RATIO GOES HERE> audience.wav audience2.wav
>   Now you have a copy of audience2.wav that has the right number
>   of samples between the beginning and ending. Unfortunately it has a
>   funny sampling rate.
> - What we want to do now is change the sampling rate in the header
>   WITHOUT changing the audio data at all.  One way to fix that 
>   is to use sox like so:
>   sox audience2.wav -t raw audience2.raw
>   Now you have the same audio data with no header.
>   You can use sox again to create a new header.
>   To do this, you need to know enough about the audio format
>   to tell sox what to do. For "cd quality" wav files, that
>   would be like so:
>   sox -r 44100 -c 2 -s -w audience2.raw audience_final.wav
> The end result is that you have hopefully compensated for the
> two different clock frequencies in your two recorders.
> Now import audience_final.wav into your ardour session,
> align the start with the desk recording as in step 1, and
> hopefully you'll be in pretty good shape  - check to
> see if the endings are still in sync.
> Let me know if it works :-)
> if it's even further out of sync, that means I got the ratio
> backwards :-)  Invert and start again from the sndfile-resample step. 
> (e.g. if you were using a ratio of 0.99,  use instead 1/0.99 = 1.010101...)

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