Frequency-based interpolation of sampled signals with applications in audio restoration

An interpolation technique has been developed for replacing missing samples in a sampled waveform drawn from a stationary stochastic process, given the power spectrum for the process. The method works with a finite block of data and is based on the assumption that components of the block discrete Fourier transform (DFT) are Gaussian zero-mean independent random variables with variance proportional to the power spectrum at each frequency value. These assumptions make the interpolator particularly suitable for signals with a sharply defined harmonic structure, such as audio waveforms recorded from music or voiced speech. Some results are presented and comparisons are made with previous techniques.<>