interpolate spurious zero amplitude samples

Special Interest Groups Audio Processing
1

Hi,

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?
spuriouszeros.png

2

That’s weird indeed…

You could try using a low-pass filter…

3

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?

4

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?