Declipper not finding clipped samples

Question

I have a heavily clipped piece of music I wish to declip.  If I scan the audio with Audition (CS6) “Amplitude Statistics”, it reports the number of clipped samples in their thousands.
If I scan within declipper, it reports none.  In fact even if I deliberately increase the amplitude on the audio by +10db or more, the declipper still reports no clipped samples.
The declipper tool is simply not working, and no matter what I try, it never reports clipped samples.
Any ideas anyone?

Answer 1

SteveD333 wrote:
You could have done the same thing at -20dB; the waveform would have looked the same, just lower in amplitude.  I wouldn’t call that clipping, I’d call it an example of what happens when a digital representation of the real waveform is inprecise. I would see your example as more accurately describing sampling errors, which due to the algorithm that seeks to recreate the true waveform, does the best it can, and hence you see the result.  It will work, it just isn’t right.
Clipping's not quite as straightforward as that, I'm afraid. If the samples are in the correct relative positions, then there has been no overload situation in the digital sense - but if you try to output the waveforms I created (which could occur quite legitimately), you'd almost certainly find that the D-A converter itself would clip, as you would have exceeded its analog output capability. But the samples, as I said, could be legitimate... and to be fair, most manufacturers these days recognise this situation, and allow sufficient headroom for the worst case output scenario.
Clipping is only a form of distortion - in other words, the arrangement of the samples does not represent the rates of change of the input signal. So any sample that's correctly in place at 0dB isn't clipped. Real-world clipping, when it occurs, has to occur either in the A-D or D-A conversion. So let's be very clear - clipping isn't a digital thing at all, and correcting it (since you have no original reference) is at best a statistical process. On a rising waveform, what you have to analyse is whether the rate of change increase that suddenly arises when a sample can't go as high as the previous several samples might indicate is correct or not. In the case of a string of 0dB samples, then there's an extremely good chance that they are incorrect (unless this is a high amplitude square wave...), but you still have no information, other than a spectral analysis guess from either side, as to what the correct positions should be.
And it's this that the Amplitude statistics can't work out, because there simply isn't time. And it's also what causes the clipping analysis sometimes to come up with results that don't appear to make sense. It really is a case of analysing where a waveform is rising, and where it falls again, and trying to correlate the samples in between to represent where it thinks the peak should be. Obviously the closer you get to the Nyquist limit, the easier this is, because in the limiting case, you can guarantee that a sample that appears to be too low (apparent increase in rate of change) actually is. The lower the rate of change though, the harder to analyse, and more imprecise, that process gets.