Temporal and Spectral Processing Methods for Processing of Degraded Speech: A Review <s> Inverse Filtering <s> When a conversation takes place inside a room, the acoustic speech signal is distorted by wall reflections. The room’s effect on this signal can be characterized by a room impulse response. If the impulse response happens to be minimum phase, it can easily be inverted. Synthetic room impulse responses were generated using a point image method to solve for wall reflections. A Nyquist plot was used to determine whether a given impulse response was minimum phase. Certain synthetic room impulse responses were found to be minimum phase when the initial delay was removed. A minimum phase inverse filter was successfully used to remove the effect of a room impulse response on a speech signal. <s> BIB001 </s> Temporal and Spectral Processing Methods for Processing of Degraded Speech: A Review <s> Inverse Filtering <s> A novel method is proposed for realizing exact inverse filtering of acoustic impulse responses in room. This method is based on the principle called the multiple-input/output inverse theorem (MINT). The inverse is constructed from multiple finite-impulse response (FIR) filters (transversal filters) by adding some extra acoustic signal-transmission channels produced by multiple loudspeakers or microphones. The coefficients of these FIR filters can be computed by the well-known rules of matrix algebra. Inverse filtering in a sound field is investigated experimentally. It is shown that the proposed method is greatly superior to previous methods that use only one acoustic signal-transmission channel. The results prove the possibility of sound reproduction and sound reception without any distortion caused by reflected sounds. > <s> BIB002 </s> Temporal and Spectral Processing Methods for Processing of Degraded Speech: A Review <s> Inverse Filtering <s> This paper presents an efficient algorithm for high-quality speech capture in applications such as hands-free teleconferencing or voice recording by personal computers. We process the microphone signals by a subband adaptive filtering structure using a modulated complex lapped transform (MCLT), in which the subband filters are adapted to maximize the kurtosis of the linear prediction (LP) residual of the reconstructed speech. In this way, we attain good solutions to the problem of blind speech dereverberation. Experimental results with actual data, as well as with artificially difficult reverberant situations, show very good performance, both in terms of a significant reduction of the perceived reverberation, as well as improvement in spectral fidelity. <s> BIB003
Reducing reverberation through inverse filtering is one of the most common approaches. The basic idea is to pass the reverberant signal through a second filter, which inverts the reverberation process and recovers the original signal. This can be written as where g(n) is the inverse filter impulse response and ŝ (n) is the delayed replica of s(n). There are several well known inverse filtering methods to dereverberate the original signal BIB001 BIB002 BIB003 . The challenge in the inverse filtering method is to find the inverse impulse response g(n). The perfect reconstruction of the original signal exists only if the room impulse response function is a minimum phase filter, whose poles and zeros are all inside the unit circle. But in practical case, most room transfer functions are non minimum phase, due to the late energy in the room impulse response and, therefore, inverse filtering based techniques have limited scope in practice BIB001 .