Quantization, the process of approximating continuous-amplitude signals by digital (discrete-amplitude) signals, is an important aspect of data compression or coding, the field concerned with the reduction of the number of bits necessary to transmit or store analog data, subject to a distortion or fidelity criterion. The independent quantization of each signal value or parameter is termed scalar quantization, while the joint quantization of a block of parameters is termed block or vector quantization. This tutorial review presents the basic concepts employed in vector quantization and gives a realistic assessment of its benefits and costs when compared to scalar quantization. Vector quantization is presented as a process of redundancy removal that makes effective use of four interrelated properties of vector parameters: linear dependency (correlation), nonlinear dependency, shape of the probability density function (pdf), and vector dimensionality itself. In contrast, scalar quantization can utilize effectively only linear dependency and pdf shape. The basic concepts are illustrated by means of simple examples and the theoretical limits of vector quantizer performance are reviewed, based on results from rate-distortion theory. Practical issues relating to quantizer design, implementation, and performance in actual applications are explored. While many of the methods presented are quite general and can be used for the coding of arbitrary signals, this paper focuses primarily on the coding of speech signals and parameters.

Vector quantization in speech coding

The past decade has witnessed substantial progress towards the application of low-rate speech coders to civilian and military communications as well as computer-related voice applications. Central to this progress has been the development of new speech coders capable of producing high-quality speech at low data rates. Most of these coders incorporate mechanisms to: represent the spectral properties of speech, provide for speech waveform matching, and "optimize" the coder's performance for the human ear. A number of these coders have already been adopted in national and international cellular telephony standards. The objective of this paper is to provide a tutorial overview of speech coding methodologies with emphasis on those algorithms that are part of the recent low-rate standards for cellular communications. Although the emphasis is on the new low-rate coders, we attempt to provide a comprehensive survey by covering some of the traditional methodologies as well. We feel that this approach will not only point out key references but will also provide valuable background to the beginner. The paper starts with a historical perspective and continues with a brief discussion on the speech properties and performance measures. We then proceed with descriptions of waveform coders, sinusoidal transform coders, linear predictive vocoders, and analysis-by-synthesis linear predictive coders. Finally, we present concluding remarks followed by a discussion of opportunities for future research. >

Speech coding: a tutorial review

Since even before the time of Alexander Graham Bell's revolutionary invention, engineers and scientists have studied the phenomenon of speech communication with an eye on creating more efficient and effective systems of human-to-human and human-to-machine communication. Starting in the 1960s, digital signal processing (DSP), assumed a central role in speech studies, and today DSP is the key to realizing the fruits of the knowledge that has been gained through decades of research. Concomitant advances in integrated circuit technology and computer architecture have aligned to create a technological environment with virtually limitless opportunities for innovation in speech communication applications. In this text, we highlight the central role of DSP techniques in modern speech communication research and applications. We present a comprehensive overview of digital speech processing that ranges from the basic nature of the speech signal, through a variety of methods of representing speech in digital form, to applications in voice communication and automatic synthesis and recognition of speech. The breadth of this subject does not allow us to discuss any aspect of speech processing to great depth; hence our goal is to provide a useful introduction to the wide range of important concepts that comprise the field of digital speech processing. A more comprehensive treatment will appear in the forthcoming book, Theory and Application of Digital Speech Processing [101].

https://research.iaun.ac.ir/pd/mahmoodian/pdfs/UploadFile_2643.pdf

An Introduction to Digital Speech Processing

This chapter provides an overview of the various methods and techniques used for assessment of speech quality. A summary is given of some of the most commonly used listening tests designed to obtain reliable ratings of the quality of processed speech from human listeners. Considerations for conducting successful subjective listening tests are given along with cautions that need to be exercised. While the listening tests are considered the gold standard in terms of assessment of speech quality, they can be costly and time consuming. For that reason, much research effort has been placed on devising objective measures that correlate highly with subjective rating scores. An overview of some of the most commonly used objective measures is provided along with a discussion on how well they correlate with subjective listening tests.

/pdf/speech-quality-assessment-3p312m0ru5.pdf

Speech Quality Assessment

This paper analyzes the stability and performance of pitch filters in speech coding when pitch prediction is combined with formant prediction. A computationally simple stability test based on a sufficient condition is formulated for pitch synthesis filters. For typical orders of pitch filters, this sufficient test is very tight. Based on the test, a simple stabilization technique that minimizes the loss in prediction gain of the pitch predictor is employed to generate stable synthesis filters. Finally, it is observed that the quality of decoded speech improves significantly when stable synthesis filters are employed.

/pdf/stability-and-performance-analysis-of-pitch-filters-in-18rmxwouxw.pdf

Stability and performance analysis of pitch filters in speech coders

This paper considers distance measures for determining the deviation between two smoothed short-time speech spectra. Since such distance measures are employed in speech processing applications that either involve or relate to human perceptual judgment, the effectiveness of these measures will be enhanced if they provide results consistent with human speech perception. As a first step, we suggest Flanagan's results on difference limens for formant frequencies as one basis for checking the perceptual consistency of a measure. A general necessary condition for perceptual consistency is derived for a class of spectral distance measures. A class of perceptually consistent measures obtained through experimental investigations is then described, and results obtained using one such measure under Flanagan's test conditions are presented.

Towards perceptually consistent measures of spectral distance

This paper considers the optimization of speech quality of adaptive predictive coding (APC) systems for transmission over a synchronous 16 kb/s channel. Among the important issues included in this on-going optimization study are: comparative evaluation of several methods for adaptive coding of the APC residual; comparative testing of several methods for adaptive shaping of the spectrum of the quantization noise; and optimization of various parameter values and their bit allocation. In addition to reporting the results from this study, we report on the occurrence of "limit cycles" or regions of excessive quantization noise build-up, offer an explanation for their cause in terms of the feedback gain (or "loop gain") of the APC transmitter, and present experimental results obtained using several remedial means for this problem. Also, we report on the relative properties of different APC coder configurations. Based on the results of our optimization study to date, we present several optimized APC systems.

Speech-quality optimization of 16 kb/s adaptive predictive coders

http://ieeexplore.ieee.org/iel6/8365/26347/01170369.pdf

The application of a functional perceptual model of speech to variable-rate LPC systems

Several methods are presented for the objective speech quality evaluation of narrowband LPC vocoders, based on a framework that we proposed at the 1976 ICASSP conference. In each method, the error in short-term spectral behavior between vocoded speech and the original is computed once every 10 ms. These errors are appropriately weighted and averaged over an utterance to produce a single objective score. Several short-term error measures, and time-weighting and averaging techniques are investigated. We evaluate the objective methods by correlating the resulting objective scores with formal subjective speech quality judgments. High correlations obtained indicate the usefulness of these methods.

Objective speech quality evaluation of narrowband LPC vocoders

While there exist methods in the literature for objectively evaluating the intelligibility of speech in the presence of stationary noise, little has been done regarding the objective evaluation of either the intelligibility or the quality of vocoded speech. We present a framework within which we have begun a step-by-step program to develop objective measures for vocoded speech quality that are consistent with results from subjective tests.

R. Viswanathan

Papers

Towards perceptually consistent measures of spectral distance

Speech-quality optimization of 16 kb/s adaptive predictive coders

The application of a functional perceptual model of speech to variable-rate LPC systems

Objective speech quality evaluation of narrowband LPC vocoders

A framework for the objective evaluation of vocoder speech quality