This paper examines enhancement to ITU-T Recommendation G.711.1 PCM wideband extension speech coder. To further improve the core lower-band coding performance the use of vector quantization and delayed decision coding is studied. A particular case of delayed decision coding, tree encoding, is implemented in the above standard. The bitstream is compatible with both the legacy G.711 and the G.711...

Speech is the most innate and fastest means of communication between humans. There has been a significant amount of research in the field of speech signal processing, and numerous models have been designed. The evolutionary origins of speech remain obscure. An algorithm is designed to modify various speech parameters with different frame size using linear, quadratic, and cubic fitting method. V...

This paper presents a concealment method based on forward error correction (FEC) to improve speech quality deterioration caused by packet losses for CELP based coders. We applied our scheme to the standard ITU-T G729 standard speech coder to evaluate the proposed method. The perceptual evaluation of speech quality (PESQ) and enhanced modified bark spectral distortion (EMBSD) tests under various...

OFDM scrambling is one of the most popular techniques for secure communication. This paper proposes a new scrambling technique based on random permutation with the pseudo random binary generator to improve the performance of OFDM scrambler. To measure the intelligibility of speech, speech transmission index (STI) and common intelligibility scale (CIS) are used. The Bit Error Rate (BER) and Sign...

We present a method for learning a low-dimensional manifold for speech from clean speech samples in high-dimensional space. Using this manifold, we perform speech denoising by projecting noisy speech onto the manifold to remove nonspeech components. This method of denoising classifies our algorithm as a signal subspace denoising method, where highdimensional noisy data is projected onto the sig...

Compared with thresholding methods based on the traditional wavelet transform (WT), empirical (EWT) has been demonstrated to outperform in terms of noise elimination by constructing an adaptive filter bank. However, as state-of-the-art version EWT, enhanced EWT (EEWT) requires that number components superposed signal prior knowledge is known, which impractical reality and limits application thi...

We propose a spectro-temporal regularization approach for NMF that accounts for a source’s spectral variability over time. The regularization terms allow NMF to adapt the spectral basis matrices optimally to reduce mismatch between the spectral characteristics of sources observed during training and encountered during separation. We first tested our algorithm on a simulated source separation ta...

VoIP speech coders have PLC mechanisms to compensate for a packet loss. In this paper, various kinds of packet losses are designed. With these losses the PLC algorithms of speech coders are evaluated and analyzed by PESQ value. Based on these outcomes, reduction of loss rate is crucial, especially for VoIP. Thus, we suggest an enhanced congestion avoidance RED mechanism to enable PLC work bette...

This paper proposes a time-domain method to improve speech intelligibility in noisy scenarios. In the proposed approach, series of Gammatone filters are adopted detect harmonic components speech. The outputs amplified emphasize first harmonics, reducing masking effects acoustic noises. GTFF0 solution and two baseline techniques examined considering four background noises with different non-stat...