Low bit rate speech coding methods and a new interframe differential coding scheme for line spectrum pairs

Ankara : Department of Electrical and Electronics Engineering and the Institute of Engineering and Sciences of Bilkent University, 1992.Thesis (Master's) -- Bilkent University, 1992.Includes bibliographical references leaves 30-32.Low bit rate speech coding techniques and a new coding scheme for vocal tract parameters are presented. Linear prediction based voice coding techniques (linear predictive coding and code excited linear predictive coding) are examined and implemented. A new interframe differential coding scheme for line spectrum pairs is developed. The new scheme reduces the spectral distortion of the linear predictive filter while maintaining a high compression ratio.Erzin, EnginM.S

Comparison of CELP speech coder with a wavelet method

This thesis compares the speech quality of Code Excited Linear Predictor (CELP, Federal Standard 1016) speech coder with a new wavelet method to compress speech. The performances of both are compared by performing subjective listening tests. The test signals used are clean signals (i.e. with no background noise), speech signals with room noise and speech signals with artificial noise added. Results indicate that for clean signals and signals with predominantly voiced components the CELP standard performs better than the wavelet method but for signals with room noise the wavelet method performs much better than the CELP. For signals with artificial noise added, the results are mixed depending on the level of artificial noise added with CELP performing better for low level noise added signals and the wavelet method performing better for higher noise levels

CELP ALGORITHMAND IMPLEMENTATION FORSPEECHCOMPRESSION

ABSTRACT This paper describes a fast algorithm and implementation of code excited linear predictive (CELP) speech coding. It presents principles of the algorithm, including (i) fast conversion of line spectrum pair parameters to linear predictive coding parameters, and (ii) fast searches of the parameters of adaptive and stochastic codebooks. The algorithm can be readily used for speech compression applications, such as on (i) high quality low-bit rate speech transmission in pointto-point or store-and-forward (network based) mode, and (ii) efficient speech storage in speech recording or multimedia databases. The implementation performs in real-time and near real-time on various platforms, including an IBM-PC AT equipped with a TMS32OC30 module, an IBM PC 486, a SUN Sparcstation 2, a SUN Sparcstation 5, and an IBM Power PC (Power 590). l. INTRODUCTION Why is CELP Useful ? Obtaining efficient representation of speech at low bit rates for communication or storage has been a problem of considerable importance, because of technical as well as economical requirements. Telephone-quality digital speech in a pulse code modulation (PCM) form requires a 64 kbits/s rate which cannot be transmitted in real time through 6 kHz and 30 kHz channel capacities of HF and VHF bands, respectively. Voice mail and multimedia employ speech storage, demanding efficient ways of storing speech, since one minute of PCM speech already requires 480 kbytes of storage space. Even if the channel can accommodate real-time speech, speech compression allows more communication connections to share the precious channel. Similarly, speech compression allows more speech messages to be stored in the storage of the same size. This paper describes a speech compression technique for those purposes, called code-excited linear predictive (CELP) coding [Atal86] [JlaJS93], which obtains bit rates of as low as 4.8 kbits/s, giving a compression ratio of up to 13: 1 The importance of CELP goes beyond its quality vs. bit-rate performance, as it *provides a generic structure for future generation of' perceptual speech coders If further compression iIs still required, the coder minimizes the error perceptibility by exploiting masking properties of human speech perception. To certain extent, the speech energy itself perceptually masks the distortion. Thus the same energy levels of distortion have different perceptual effect if applied to speech signals with different energy levels. This approach promises a new level of highier quality and lower bit rate speech compression One novelty of CELP is in incorporating the masking property in a working, practical scheme. Such incorporation is non trivial blecause perceptual distortion measures lack tractable means that have often been available in the traditional distortion energy measure. 9

An investigation into glottal waveform based speech coding

Coding of voiced speech by extraction of the glottal waveform has shown promise in improving the efficiency of speech coding systems. This thesis describes an investigation into the performance of such a system. The effect of reverberation on the radiation impedance at the lips is shown to be negligible under normal conditions. Also, the accuracy of the Image Method for adding artificial reverberation to anechoic speech recordings is established. A new algorithm, Pre-emphasised Maximum Likelihood Epoch Detection (PMLED), for Glottal Closure Instant detection is proposed. The algorithm is tested on natural speech and is shown to be both accurate and robust. Two techniques for giottai waveform estimation, Closed Phase Inverse Filtering (CPIF) and Iterative Adaptive Inverse Filtering (IAIF), are compared. In tandem with an LF model fitting procedure, both techniques display a high degree of accuracy However, IAIF is found to be slightly more robust. Based on these results, a Glottal Excited Linear Predictive (GELP) coding system for voiced speech is proposed and tested. Using a differential LF parameter quantisation scheme, the system achieves speech quality similar to that of U S Federal Standard 1016 CELP at a lower mean bit rate while incurring no extra delay

The development of speech coding and the first standard coder for public mobile telephony

This thesis describes in its core chapter (Chapter 4) the original algorithmic and design features of the ??rst coder for public mobile telephony, the GSM full-rate speech coder, as standardized in 1988. It has never been described in so much detail as presented here. The coder is put in a historical perspective by two preceding chapters on the history of speech production models and the development of speech coding techniques until the mid 1980s, respectively. In the epilogue a brief review is given of later developments in speech coding. The introductory Chapter 1 starts with some preliminaries. It is de- ??ned what speech coding is and the reader is introduced to speech coding standards and the standardization institutes which set them. Then, the attributes of a speech coder playing a role in standardization are explained. Subsequently, several applications of speech coders - including mobile telephony - will be discussed and the state of the art in speech coding will be illustrated on the basis of some worldwide recognized standards. Chapter 2 starts with a summary of the features of speech signals and their source, the human speech organ. Then, historical models of speech production which form the basis of di??erent kinds of modern speech coders are discussed. Starting with a review of ancient mechanical models, we will arrive at the electrical source-??lter model of the 1930s. Subsequently, the acoustic-tube models as they arose in the 1950s and 1960s are discussed. Finally the 1970s are reviewed which brought the discrete-time ??lter model on the basis of linear prediction. In a unique way the logical sequencing of these models is exposed, and the links are discussed. Whereas the historical models are discussed in a narrative style, the acoustic tube models and the linear prediction tech nique as applied to speech, are subject to more mathematical analysis in order to create a sound basis for the treatise of Chapter 4. This trend continues in Chapter 3, whenever instrumental in completing that basis. In Chapter 3 the reader is taken by the hand on a guided tour through time during which successive speech coding methods pass in review. In an original way special attention is paid to the evolutionary aspect. Speci??cally, for each newly proposed method it is discussed what it added to the known techniques of the time. After presenting the relevant predecessors starting with Pulse Code Modulation (PCM) and the early vocoders of the 1930s, we will arrive at Residual-Excited Linear Predictive (RELP) coders, Analysis-by-Synthesis systems and Regular- Pulse Excitation in 1984. The latter forms the basis of the GSM full-rate coder. In Chapter 4, which constitutes the core of this thesis, explicit forms of Multi-Pulse Excited (MPE) and Regular-Pulse Excited (RPE) analysis-by-synthesis coding systems are developed. Starting from current pulse-amplitude computation methods in 1984, which included solving sets of equations (typically of order 10-16) two hundred times a second, several explicit-form designs are considered by which solving sets of equations in real time is avoided. Then, the design of a speci??c explicitform RPE coder and an associated eÆcient architecture are described. The explicit forms and the resulting architectural features have never been published in so much detail as presented here. Implementation of such a codec enabled real-time operation on a state-of-the-art singlechip digital signal processor of the time. This coder, at a bit rate of 13 kbit/s, has been selected as the Full-Rate GSM standard in 1988. Its performance is recapitulated. Chapter 5 is an epilogue brie y reviewing the major developments in speech coding technology after 1988. Many speech coding standards have been set, for mobile telephony as well as for other applications, since then. The chapter is concluded by an outlook