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

The vector sum excited linear prediction speech coder is presented. It utilizes a codebook with a structure that allows for a very efficient search procedure. Other advantages of the VSELP codebook structure are discussed, and a detailed description of an 8-kb/s VSELP coder is given. This coder was selected by the Telecommunications Industry Association (TIA) as the standard for use in North American digital cellular telephone systems. The coder uses two VSELP excitation codebooks, a gain quantizer which is robust to channel errors, and a novel adaptive pre/postfilter arrangement. >

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Vector sum excited linear prediction (VSELP) speech coding at 8 kbps

An adaptive postfiltering algorithm for enhancing the perceptual quality of coded speech is presented. The postfilter consists of a long-term postfilter section in cascade with a short-term postfilter section and includes spectral tilt compensation and automatic gain control. The long-term section emphasizes pitch harmonics and attenuates the spectral valleys between pitch harmonics. The short-term section, on the other hand, emphasizes speech formants and attenuates the spectral valleys between formants. Both filter sections have poles and zeros. Unlike earlier postfilters that often introduced a substantial amount of muffling to the output speech, our postfilter significantly reduces this effect by minimizing the spectral tilt in its frequency response. As a result, this postfilter achieves noticeable noise reduction while introducing only minimal distortion in speech. The complexity of the postfilter is quite low. Variations of this postfilter are now being used in several national and international speech coding standards. This paper presents for the first time a complete description of our original postfiltering algorithm and the underlying ideas that motivated its development. >

Adaptive postfiltering for quality enhancement of coded speech

A speech coding system employing an adaptive codebook model of periodicity is augmented with a pitch-predictive filter (PPF). This PPF has a delay equal to the integer component of the pitch-period and a gain which is adaptive based on a measure of periodicity of the speech signal. In accordance with an embodiment of the present invention, speech processing systems which include a first portion comprising an adaptive codebook and corresponding adaptive codebook amplifier and a second portion comprising a fixed codebook coupled to a pitch filter, are adapted to delay the adaptive codebook gain; determine the pitch filter gain based on the delayed adaptive codebook gain, and amplify samples of a signal in the pitch filter based on said determined pitch filter gain. The adaptive codebook gain is delayed for one subframe. The pitch filter gain equals the delayed. adaptive codebook gain, except when the adaptive codebook gain is either less than 0.2 or greater than 0.8., in which cases the pitch filter gain is set equal to 0.2 or 0.8, respectively.

CS-ACELP speech compression system with adaptive pitch prediction filter gain based on a measure of periodicity

This paper describes the 8 kb/s speech coding algorithm G.729 which has been standardized by ITU-T. The algorithm is based on a conjugate-structure algebraic CELP (CS-ACELP) coding technique and uses 10 ms speech frames. The codec delivers toll-quality speech (equivalent to 32 kb/s ADPCM) for most operating conditions. This paper describes the coder structure in detail and discusses the reasons behind certain design choices. A 16-b fixed-point version has been developed as part of Recommendation G.729 and a summary of the subjective test results based on a real-time implementation of this version are presented.

Design and description of CS-ACELP: a toll quality 8 kb/s speech coder

A SELP (stochastically excited linear prediction) algorithm consisting of a two-stage vector quantization using an adaptive codebook and a stochastic codebook is described. The adaptive codebook quantization is similar to a closed-loop long-term prediction filter if the predictor delay is more than one frame length. The performance of the adaptive codebook procedure is improved by extending its codebook to include candidate vectors constructed from past synthetic excitation displaying a high level of periodicity. An algorithm is introduced which through increased symmetry of the error criterion significantly reduces the computational effort required for the search through the adaptive codebook. The algorithm can also be used for stochastic codebooks consisting of overlapping candidate vectors. It is shown that for stochastic codebooks in which neighboring candidates overlap for all but two samples, the quantization performance is as high as for codebooks containing fully independent candidate vectors. >

Improved speech quality and efficient vector quantization in SELP

An efficient stochastically excited linear predictive algorithm for high quality low bit rate transmission of speech

Apparatus for encoding speech using a code excited linear predictive (CELP) encoder using a virtual searching technique during speech transitions such as from unvoiced to voiced regions of speech. The encoder compares candidate excitation vectors stored in a codebook with a target excitation vector representing a frame of speech to determine the candidate vector that best matches the target vector by repeating a first portion of each candidate vector into a second portion of each candidate vector. For increased performance, a stochastically excited linear predictive (SELP) encoder is used in series with the adaptive CELP encoder. The SELP encoder is responsive to the difference between the target vector and the best matched candidate vector to search its own overlapping codebook in a recursive manner to determine a candidate vector that provides the best match. Both of the best matched candidate vectors are used in speech synthesis.

Richard Harry Ketchum

Papers

Improved speech quality and efficient vector quantization in SELP

An efficient stochastically excited linear predictive algorithm for high quality low bit rate transmission of speech

Code excited linear predictive vocoder using virtual searching

Vocoder à prédiction linéaire excité par codes

Linearer Prädiktionsvocoder mit Code-Anregung