Multipulse-excitation has greatly improved the speech quality achievable from linear predictive coders which previously required speech to be classified as voiced or unvoiced for excitation purposes. Multipulse removes the need for voicing classification, improving speech quality by enhancing the excitation and offsetting errors in the vocal tract filter. An investigation of multipulse-excitation applied to a channel vocoder and a formant synthesiser was conducted. The prime objective was to improve the performance of these algorithms and achieve multipulse linear prediction speech quality, our target quality. This dissertation outlines and restates the idea of multipulse-excitation applied to a linear predictive vocoder. We then examine a high quality channel vocoder and formant synthesiser, and the use of multipulse-excitation to improve their performances. In each case time and frequency domain multipulsecalgorithms were used. Various modifications were made to these algorithms in order to accommodate multipulse-excitation and improve the overall speech quality. In the case of the channel vocoder this involved a novel technique, which sacrificed the inherent waveform preserving properties of the multipulse algorithm. Only by increasing both the pulse rate and the number of channels could the multipulse-excited channel vocoder achieve our target quality. With the formant synthesiser it was possible, by variation of the pulse rate alone, to achieve our target quality. Comparisons are drawn between the three multipulse algorithms and reasons given for their differing performance; this is substantiated by experimental results. These results suggested interesting improvements to the multipulse-excited formant synthesiser; and also hinted at a new and novel technique for formant tracking, using multipulse-excitation applied to a formant synthesiser.

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Speech coding is a very important area of research in digital signal processing. It is a fundamental element of digital communications and has progressed at a fast pace in parallel to the increase of demands in telecommunication services and capabilities. Most of the speech coders reported in the literature are based on linear prediction (LP) analysis. Code Excited Linear Predictive (CELP) coder is a typical and popular example of this class of coders. This coder performs LP analysis of speech for extracting LP coefficients and employs an analysis-by-synthesis procedure to search a stochastic codebook to compute the excitation signal. The method used for performing LP analysis plays an important role in the design of a CELP coder. The autocorrelation method is conventionally used for LP analysis. Though this works reasonably well for noise-free (clean) speech, its performance goes down when signal is corrupted by noise. Spectral analysis of speech signals in noisy environments is an aspect of speech coding that deserves more attention. This dissertation studies the application of recently proposed robust LP analysis methods for estimating the power spectrum envelope of speech signals. These methods are the moving average, moving maximum and average threshold methods. The proposed methods will be compared to the more commonly used methods of LP analysis, such as the conventional autocorrelation method and the Spectral Envelope Estimation Vocoder (SEEVOC) method. The Linear Predictive Coding (LPC) spectrum calculated from these proposed methods are shown to be more robust. These methods work as well as the conventional methods when the speech signal is clean or has high signal-to-noise ratio. Also, these robust methods give less quantisation distortion than the conventional methods. The application of these robust methods for speech compression using the CELP coder provides better speech quality when compared to the conventional LP analysis methods.

Contemporary and future speech telecommunication systems now utilise low bit rate (LBR) speech coding techniques in efforts to eliminate bandwidth expansion as a disadvantage of digital coding and transmission. These speech coders employ model-based approaches in compressing human speech into a number of parameters, using a well-known process known as linear predictive coding (LPC). However, a major side-effect observed in these coders is that errors in the model parameters have noticeable and undesirable consequences on the synthesised speech quality, and unless they are protected from such corruptions, the level of service quality will deteriorate rapidly. Traditionally, forward error correction (FEC) coding is used to remove these errors, but these require substantial redundancy. Therefore, a different perspective of the error control problems and solutions is necessary. In this thesis, emphasis is constantly placed on exploiting the constraints and residual redundancies present in the model parameters. It is also shown that with such source criteria in the LBR speech coders, varying degrees of error protection from channel corruptions are feasible. From these observations, error control requirements and methodologies, using both block- and parameter-orientated aspects, are analysed, devised and implemented. It is evident, that under the unusual circumstances which LBR speech coders have to operate in, the importance and significance of source reliant error control will continue to attract research and commercial interests. The work detailed in this thesis is focused on two LPC-based speech coders. One of the ideas developed for these two coders is an advanced zero redundancy scheme for the LPC parameters which is designed to operate at high channel error rates. Another concept proposed here is the use of source criteria to enhance the decoding capabilities of FEC codes to exceed that of maximum likelihood decoding performance. Lastly, for practical operation of LBR speech coders, lost frame recovery strategies are viewed to be an indispensable part of error control. This topic is scrutinised in this thesis by investigating the behaviour of a specific speech coder under irrecoverable error conditions. In all of the ideas pursued above, the effectiveness of the algorithms formulated here are quantified using both objective and subjective tests. Consequently, the capabilities of the techniques devised in this thesis can be demonstrated, examples of which are: (1) higher speech quality produced under noisy channels, using an improved zero-redundancy algorithm for the LPC filter coefficients; (2) as much as 50% improvement in the residual BER and decoding failures of FEC schemes, through the utilisation of source criteria in LBR speech coders; and (3) acceptable speech quality produced under high frame loss rates (14%), after formulating effective strategies for recovery of speech coder parameters. It is hoped that the material described here provide concepts which can help achieve the ideals of maximum efficiency and quality in LBR speech telecommunications.

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