A week or two ago, calamari posted a great little utility to improve the audio sync when converting .avi files called 'AVIFix v1.20'. Using this utility, one would load the .avi, and the audio .wav (which had been saved out uncompressed via VirtualDub). calamari's utility would then compare the length of the audio and video streams and allow you to 'sync them up' by rewriting the AVI headers. This seems to help immensely in getting the audio to not fall out of sync after encoding to vcd/svcd.
However, I noticed something odd. It appears that resampling a .wav file from 48Khz to 44.1Khz (using ssrc.exe) changes the length of the audio file! So if I used AVIFix with the 48KHz .wav, I'm actually breaking the sync (possibly even more than it was!) because the audio length is NOT the same after down-sampling!
Is this known/expected? Should downsampling a file have any effect on its length?
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Of course. Think about it - if the time doesn't change, there are 48000 samples per second or 44100 samples per second. This is a different amount of data, so you would expect your file size to shrink by 8.8% (48/44.1 - 100)
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Sure, I have no problem with a difference in file size. Ahh... I was unclear on what I meant by length. I didn't mean filesize (of course it would be smaller!), I meant length as in playtime.
For example, the original 48Khz .wav was 1:13:54.62, while the converted 44.1Khz .wav was 1:13:54.21. These numbers are made up, but reflect the type of changes I'm seeing. Granted, the difference here is minor, but if compounded by the use of AVIFix, you can see a noticble desync of a/v.
I would expect both .wav files to be _exactly_ the same play-length: 1:13:54.62 -
Whenever I have used ssrc the length (in time) of the audio file remains unchanged - which is what I would expect when resampling.
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