SIGNAL ESTIMATION FROM MODIFIED SHORT-TIME FOURIER-TRANSFORM

SIGNAL ESTIMATION FROM MODIFIED SHORT-TIME FOURIER-TRANSFORM
复制标题

DOI:
10.1109/tassp.1984.1164317
复制
发表时间:
1984-01-01
期刊:
IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING
影响因子:
--
通讯作者:
LIM, JS
LIM, JS
中科院分区:
其他
文献类型:
--
作者:
GRIFFIN, DW;LIM, JS

文献摘要

被引文献

相似文献

在本文中,我们提出了一种算法来估计从其修改的短时傅里叶变换(STFT)的信号。该算法计算简单,通过最小化估计信号的STFT和修改的STFT之间的均方误差来获得。使用该算法,我们还开发了一种迭代算法来估计信号的修改STFT幅度。示出的迭代算法减少,在每次迭代中,估计信号的STFT幅度和修改的STFT幅度之间的均方误差。迭代算法中涉及的主要计算是离散傅里叶变换(DFT)计算,并且该算法似乎是用当前硬件技术可实时实现的。本文提出的算法已应用于语音的时间尺度修正。由此产生的系统生成非常高质量的语音,并且似乎比任何现有的方法在性能上更好。
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.