The quality of speech signal over a VoIP system is degraded by various network layer problems which include jamming, jitter, packet loss. Different types of noises also degrade the quality of speech signal such as external noise and quantization noise. This paper improves the quality of VoIP speech signal affected by these noises and network layer problems. The quality of degraded VoIP speech s...

Telephone speech is typically bandlimited to 4 kHz, resulting in a ‘muffled’ quality. Coding speech with bandwidth greater than 4 kHz reduces this distortion, but requires a higher bit rate to avoid other types of distortion. An alternative to coding wider bandwidth speech is to exploit correlation between the 0-4 kHz and 4-8 kHz speech bands to resynthesize wideband speech from narrowband spee...

Digital speech coding is important problem in the area of network speech applications where there is a necessity of transmission of big amount of data through limited bandwidth channels. Investigations in this field aim at finding new methods for signal compression and compatibility with the network transfer protocols. Digital speech coding involves sampling and amplitude quantization of the sp...

We have seen rapid progress in high-quality compression of wideband audio signals. Today’s coding algorithms can achieve substantially better compression than was thought possible only a few years ago. In the case of audio coding with its bandwidth of 20 kHz and more, the concept of perceptual coding has paved the way for significant bit rate reductions. However, multiple coding can reveal orig...

A novel approach to speech coding is introduced in this paper for the coding of speech signals used in vocoders. While conventional methods code the original input speech signal, the proposed algorithm codes the timescale modified signal instead. In this method, the transmitter speeds up original input speech signal by a factor of beta using waveform similarity overlap-addmethod (WSOLA). This s...

This paper presents an analysis-by-synthesis multimode harmonic coder (AbS-MHC) that employs new techniques to improve both the speech model accuracy and the parameter estimation robustness in the low rate harmonic coding framework. To improve the speech model accuracy, an enhanced frequency domain transition model is used in conjunction with the sinusoidal model based harmonic coding of voiced...

Speech coding at very low bitrate is useful for purposes such as voice communication over computer networks. However, speech coding at around 2.0 kbit/s is di cult for CELP coders while maintaining a high quality. In this paper, a speech coding model called `normalized pitch waveform' and its quantization scheme are presented, aiming for effective compression coding of the `voiced' speech. List...

Adaptive linear prediction is commonly used as a key step in digital coding of speech. This paper discusses some of the techniques that have been developed for adapting and coding the predictor coefficients in speech coders. The linear predictors in high quality speech coding often consist of two stages, a short-time span (formant) filter and a long-time span (pitch) filter. The use of such fil...