The application of algebraic code excited linear prediction (ACELP) coding to wideband speech is presented. An algebraic codebook with a 20 bit address can be used without any storage requirements and, more importantly, with a very efficient search procedure which allows for real-time implementation. The authors describe an efficient procedure for searching such a large codebook deploying a focused search strategy, where less than 0.1% of the codebook is searched with performance very close to that of a full search. High-quality speech at a bit rate of 13 kbps was obtained.>

Error-correcting codes are considered as codebooks for high-performance vector quantization (VQ) of the IID Gaussian source at fractional bit rate. A family of good rate-one-half codes is introduced: the stretched Golay codes. The performance of these codes is compared to other good block codes, trellis-coded quantization, and other techniques. The stretched Golay codes are shown to outperform previously published results for block lengths 32, 40, 56, and 64. The good performance, together with fast decoding make these techniques attractive for applications such as low-bit-rate coding of speech.>

Code-Excited Linear Prediction (CELP) produces high quality synthetic speech at low bit rate. However the basic scheme leads to huge computational loads. The paper describes a related scheme, which allows real time implementation on current DSP chips. The very efficient search procedure in the codebook is achieved by means of a new technique called "backward filtering" and the use of algebraic codes. RSB performances are reported for a variety of conditions.

The following paper discusses a vectorial description of the subband decomposition/ reconstruction process and shows its advantages compared to the usual description. The following points are examined:\bulletnecessary and sufficient conditions for exact reconstruction,\bulletreconstruction error measure,\bulletinteresting properties of exact reconstruction processes, and comparison between transform coding and subband coding.

This paper deals with comparisons between various coding schemes for the digital transmission of high quality sound signals typical of what will be transmitted through a Direct Broadcast Satellite System (DBS). Several PCM coding techniques using instantaneous logarithmic coding laws [1] (A-law and µ-law) or near-instantaneous coding law (block coding scheme) such as the NICAM 3 system [2] as well as the Adaptive Delta Modulation (ADM) [3] are compared in this paper. Several pre-emphasis/de-emphasis laws are also proposed to optimally shape the spectrum of the sound program signal before encoding [1]. An investigation of the properties of these laws is carried out and an evaluation, both objective and subjective, of the impact of their utilisation in a codec is presented.

This paper shows a technique of encoding the LPC residual which allows the achievement of speech coding with residual excitation at a bit rate as low as 2400 bps. The method is inspired by the multipulse coding approach introduced by Atal, associated with an irregular downsampl-ing. The real time implementation of a 4800 bps vocoder on a single TMS 320 DSP is discussed.

The paper presents the full description and discusses the performances of a 4800 bit per second residual excited linear prediction vocoder. The LPC analysis is efficiently performed using a type of binary-tree search vector-quantization approach. The technique, which is described in ref (1), uses a set of hyperplane equations to perform a hierarchical pattern classification of the input autocorrelation vector in the autocorrelation space. The end result of the search is the integer i1which is the index of the most appropriate (in the Itakura-distance sense) prediction filter out of a set of N preset filters. The search requires only\Log_{2}Ndot products. In this case vector quantization presents two advantages over the classical approach of the Durbin algorithm followed by scalar quantization. First, a faster algorithm is obtained. Second, the same accuracy in filter representation is possible with less bits per second and consequently more bits can be allocated for representing the residual and gain. The residual is vector quantized in the time domain by blocks of 16 the samples according to the approach of ref (2). The 16 sample block is essentially encoded using the integer I2which is the index of the most appropriate 16-sample waveform out of set of M preset prototype waveforms stored in memory. The paper includes preference testings for comparison with other types of 4800 Kbit/sec vocoders. Some sample recordings will be presented at the conference. Finally, preliminary results in the attempt to implement the vocoder in real time on a MAP 200 array processor are discussed.

The paper presents a new approach to the problem of speech coding in the range of 4 to 8 kbits/sec along with experimental tests performed on a MAP 200 array processor. The basic idea consists of using a dictionary of K indexed waveforms of fixed duration T (where T lies between 1 and 4 ms). The input signal is broken down into blocks of duration T. For each segment the Dictionary is searched for that prototype waveform which is, with respect to some distance measure, the closest to the input block. The digital representation of the signal is thereafter the index of the prototype. The paper addresses the problem of the judicious choice of distance measures and the design of a good dictionary. Experimental results are provided for dictionary coding of the signal using a distance measure combining time and frequency domain features. Experimental results are further presented for the coding of the LPC residual.