The pseudoanechoic model was proposed recently to simplify the parameter estimation in blind source separation based on frequency-domain independent component analysis. In the method, after separation based in the pseudoanechoic model a time-frequency Wiener postfilter to improve the separation is applied. In this contribution, a deeper analysis of the working principles of the Wiener postfilte...

Estimators for speech enhancement by using wavelet transform is the new technique which is proposed in this paper. Here, we proposed a new set of estimators called magnitude square spectrum estimators beyond the conventional magnitude, power estimators using wavelet transform. Maximum a posteriori(MAP), Minimum Mean Square Error(MMSE) Estimators are derived using hard masking then Soft Masking ...

This paper describes a robustness issue of the transcoder under packet loss channel environments. We briefly introduce the conventional transcoder between AMR and G.729A speech coders, and analyze the performance of transcoder under frame erasure environments. In a tandemmethod even a single packet loss significantly affects to the parametric buffers of the successive frames because it should r...

According to the distribution characteristic of noise and clean speech signal in the frequency domain, a new speech enhancement method based on teager energy operator (TEO) and perceptual wavelet packet decomposition (PWPD) is proposed. Firstly, a modified Mask construction method is made to protect the acoustic cues at the low frequencies. Then a level-dependent parameter is introduced to furt...

Codebook search has high computational load in code excited linear prediction (CELP) speech coders. In this paper, a fuzzy ARTMAP neural network (FAMNN) is used to determine the best index of shape codebook in ITU-T G.728 speech coding algorithm. In this way, the gain value is calculated according to this index and the best index of gain codebook is determined based on the minimum distance to e...

In Compressed Sensing (CS) framework, reconstruction of a signal relies on the knowledge of the sparse basis & measurement matrix used for sensing. Most of the studies so far focus on the application of CS in fields of images, radar, astronomy and Speech. This paper introduce new approach called combined basis that is made by separating voiced and unvoiced parts and applying different basis for...

This paper investigates the performance of speech codec's that uses linear predictive coding (LPC), over different languages. Investigations show that most low-rate (8kbits/s and below) speech coders show bias towards nonaccented English. When the coders are used for heavily accented English or other languages, significant performance degradation is noted. In order to judge the performance of t...

In this paper we propose a new speech compression technique based on the application of a psychoacoustic model combined with a general approach for Filter Bank Design using optimization. This technique is a modified version of the compression technique using a MDCT (Modified Discrete Cosine Transform) filter banks of 32 filters each and a psychoacoustic model. The two techniques are evaluated a...

Single-channel speech enhancement is often formulated in the Short-Time Fourier Transform (STFT) domain. As an alternative, several previous studies have reported advantages of speech processing using pitch-synchronous analysis and filtering in the modulation transform domain. We propose to use the Double Spectrum (DS) obtained by combining pitchsynchronous transform followed by modulation tran...

Existing objective speech-intelligibility measures are suitable for several types of degradation, however, it turns out that they are less appropriate in cases where noisy speech is processed by a time-frequency weighting. To this end, an extensive evaluation is presented of objective measure for intelligibility prediction of noisy speech processed with a technique called ideal time frequency (...