[ardour-users] sync imported wav files
erleichda at gmail.com
Wed Dec 7 20:40:39 PST 2005
I can not believe how perfect this worked.
Thank you very much for taking the time to share this.
On 12/6/05, Paul Winkler <pw_lists at slinkp.com> 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...)
> Paul Winkler
> ardour-users mailing list
> ardour-users at lists.ardour.org
michael jones * erleichda archiving * usa
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