interpolate spurious zero amplitude samples


I need to edit the soundtrack of a captured digital video recording (raw dv format, i.e. audio is pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s).

For some reason I don’t understand some audio samples have spurious zero amplitude. Imagine a half-period [0:pi] of a sine wave which is represented by about 50 audio samples. In my recording, samples 3, 4, 8, 9, 13, 14, etc are zero instead of having the correct non-zero value.

I’m puzzled as to what could have created these, but regardless I would like to remove these zeros and interpolate the true values from the neighbouring samples.

Is this possible from within audacity, or otherwise how could one solve this?

That’s weird indeed…

You could try using a low-pass filter…

What was the sound quality of the video like before conversion?
Does the audio now sound worse?
How did you convert from raw dv format?

Thanks for your comments.

I’ve tried a low-pass filter but even if I choose a cut-off frequency as low as 500 Hz the track sounds much like aliens in a 1950’s movie :wink:

The track was extracted from the captured video file using

ffmpeg -i video.dv -acodec copy out.wav

so nothing special here. This always worked fine before. I will try to re-capture the scene from the camera. Perhaps the camera electronics are dying?