Sharif Digital Repository

In this paper, the problem of single microphone source separation via Nonnegative Matrix Factorization (NMF) by exploiting video information is addressed. Respective audio and video modalities coming from a single human speech usually have similar time changes. It means that changes in one of them usually corresponds to changes in the other one. So it is expected that activation coefficient matrices of their NMF decomposition are similar. Based on this similarity, in this paper the activation coefficient matrix of the video modality is used as an initialization for audio source separation via NMF. In addition, the mentioned similarity is used for post-processing and for clustering the rows... 

In this paper, a fast phoneme recognition system is introduced based on sparse representation. In this approach, the phoneme recognition is fulfilled by Viterbi decoding on support vector machines (SVM) output probability estimates. The candidate classes for classification are adaptively pruned by a k-dimensional (KD) tree search followed by a sparse representation (SR) based class selector with adaptive number of classes. We applied the proposed approach to introduce a phoneme recognition system and compared it with some well-known phoneme recognition systems according to accuracy and complexity issues. By this approach, we obtain competitive phoneme error rate with promising computational... 

This paper focuses on localizing acoustic source using fuzzy logic. For this purpose, an array of microphones are used as sensors. The Time Delay Estimation (TDE) is performed by correlation between received signals from each pair of microphones. In this paper, instead of assuming a crisp value as a TDOA, we introduce fuzzy number of TDOA in which the absolute value of the cross correlation function is normalized and then assumed as TDOA membership function for each pair of microphones. Truth value of presenting acoustic source is computed in region of interest based on achieved TDOA membership functions. Computing this truth value is represented in this paper based on fuzzy logic. The... 

In this analytical study, it has been desired to develop a practical and simple control mechanism to control, at a given point and its neighborhood, the sound arriving from a distant source, assuming that a primary pure-tone sound pressure is propagated from a relatively far distance. The control model consists of a microphone as a sensor for measuring the sound pressure and a loud speaker for applying the control force. Corresponding equations have been developed to determine an optimum control force, and afterwards a parametric study on the factors affecting the control results has been performed. The results show that the control system can significantly reduce low frequency sound... 

Reverberated speech signals in noisy acoustical environments cause some problems such as reducing speech intelligibility, distinguishing speakers, locating source, quality for hands-free telephony, hearing aid, etc. Adaptive filters can be applied to suppress the interfering signals and reduce the reverberation effects or to dereverberate the received speech signals at microphone. In this paper, Bayesian State-Space Kalman and Wiener filters have been employed to reduce the effect of noise on received speech signal and their results are compared. Also, a dereverberation method is proposed by applying an inverse filter to the received speech signals to reduce the effect of reverberation on... 

Adaptive Noise Cancellers (ANCs) do not provide sufficient noise reduction in the diffuse noise fields. In this paper, a new hybrid structure is proposed as a solution to this problem. The proposed system is a combination of two subsystems, an ANC and a new multistage post-filter. The post-filter is based on linear prediction (LP) and attempts to extract speech component by using intermediate ANC signals. The system is implemented on an over-sampled DFT filterbank with different analysis and synthesis prototype filters. The experimental results using various quality measures show that the proposed system is superior to both the subband ANC and subband LP based speech enhancement systems.1 ©... 

A new voice activity detection (VAD) algorithm with soft decision output in Mel-frequency domain is developed based on hidden Markov model (HMM) and is incorporated in an HMM-based speech enhancement system. The proposed VAD uses a two-state ergodic HMM representing speech presence and speech absence. The states are constructed from noisy speech and noise HMMs used in the speech enhancement system. This composite model provides a robust detection of speech segments in the presence of noise and obviates the need for extra modeling in HMM-based speech enhancement applications. As the main purpose of the proposed VAD is to detect speech segments accurately, a hang-over mechanism is proposed and... 

Adaptive Noise Cancellation (ANC) is a well-known technique for background noise reduction in automobile and vehicular environments. The noise fields in automobile and other vehicle interior obey the diffuse noise field model closely. On the other hand, the ANC does not provide sufficient noise reduction in the diffuse noise fields. In this paper, a new multistage post-filter is designed for ANC as a solution to diffuse noise conditions. The designed post-filter is a single channel Linear Prediction (LP) based speech enhancement system. The LP is performed by an adaptive lattice filter and attempts to extract speech components by using intermediate ANC signals. The post-filter has no...