Speech data compression

The analysis-by-synthesis method is the most useful application for the parametric representation. The necessary components for the model are derived from signal analysis procedures while the output speech waveform is obtained from the synthetic procedure. This method, such as the Codebook Exited Coder (CELP) [1], is first implemented in the time domain. The basic approach is to model the correlation among the speech samples by using a linear time-varying filter. An excitation model can then be obtained by removing the correlation. Since the filter will not ignore the noise, the parametric representation does have problems with the noisy speech data. An alternative procedure is to implement the technique in the frequency domain. This leads to a flexible method for lower bit rate procedure transmission. Furthermore, it provides a suitable way to model the filter in a noisy environment. Methods such as the harmonic vocoder and Multiband Excitation Coder (MBE) [4] are all frequency domain techniques. Since the speech data is recovered from the parametric model, the output depends on the model parameters, which may greatly effect the quality of the speech.

The objective of this thesis is to develop efficient algorithms for implementing the harmonic vocoder in the frequency domain. A reUable method is developed to realize the analysis procedure and to achieve the correct fundamental elements of speech signal. An efficient method is proposed to synthesize output speech signal and to improve speech quality. Also, the techniques of model refinement and enhancement will be described in this thesis. In practice, the analogue speech signal is sampled at 8000Hz and this rate is used throughout this research. The research is concentrated on the method for speech data compression and speech quality improvement rather than coding schemes.