1984/04/01 by D. Griffin, Daniel W. Griffin, Jae Lim +1 · 9 citations
Computer Science · Engineering · #Advanced Adaptive Filtering Techniques #Speech Recognition and Synthesis #Speech and Audio Processing
paper · doi:10.1109/tassp.1984.1164317
openalex publication_date 1984/04/01 · openalex created_date 2025/10/10 · openalex updated_date 2026/08/01
In this paper, we present an algorithm to estimate a signal from its modified short-time Fourier transform (STFT). This algorithm is computationally simple and is obtained by minimizing the mean squared error between the STFT of the estimated signal and the modified STFT. Using this algorithm, we also develop an iterative algorithm to estimate a signal from its modified STFT magnitude. The iterative algorithm is shown to decrease, in each iteration, the mean squared error between the STFT magnitude of the estimated signal and the modified STFT magnitude. The major computation involved in the iterative algorithm is the discrete Fourier transform (DFT) computation, and the algorithm appears to be real-time implementable with current hardware technology. The algorithm developed in this paper has been applied to the time-scale modification of speech. The resulting system generates very high-quality speech, and appears to be better in performance than any existing method.