[ardour-users] sync imported wav files

blindman jones 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.

be well,
mjr

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
> http://www.slinkp.com
> _______________________________________________
> ardour-users mailing list
> ardour-users at lists.ardour.org
> http://lists.ardour.org/listinfo.cgi/ardour-users-ardour.org
>


--
michael jones * erleichda archiving * usa
shivering is the first step towards your destiny.....



More information about the Ardour-Users mailing list