S. Morissette

16 kbps wideband speech coding technique based on algebraic CELP

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.<<ETX>>

On reducing computational complexity of codebook search in CELP coder through the use of algebraic codes

A general framework is introduced which allows both fast search and freedom in designing codebooks with good statistical properties. Several previously proposed schemes are compared from this viewpoint. A backward filtering formulation is given to show that sparse algebraic codes (SACs) (i.e., with few nonzero components) offer distinct advantages. It is shown that they reduce the optimal-search computation per codeword. They also allow control of the statistical properties of the codebook in the time and frequency domains. This control can be dynamic in the sense that it can be made to evolve as a function of the linear predictive coding model A(z). The algebraic-code excited linear prediction (ACELP) technology which allows full duplex operation on a single TMS320C25 at rates between 4.8 and 16 kb/s and which is based on SAC-driven dynamic codebooks is described.<<ETX>>

A comparative study of the proposed high quality coding schemes for digital music

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.

Filters for subband coding analytical approach

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.

Generalization of the multipulse coding for low bit rate coding purposes: The generalized decimation

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 effect of a realizable integrator on a data signal

It is often suggested in the literature on how to smooth a staircase approximation data signal with an integrator. This paper shows the limitations of such an approach and presents some ways to correct them.

A new look at broadband square-wave frequency multipliers †

Abstract In a recent publication, Boutin et al. (1979) have presented some circuits ideas on a broadband square-wave frequency multiplier. These circuits need a comparator type A/D converter which is both rare and expensive. I n the present paper, some new circuits are presented in which the A/D converters have been replaced by D/A converters which are cheaper and more accessible.

A new concept for encoding speech amplitude time quantization

For digital modulations applied to the coding of speech signals, a fixed sampling and transmission rate is always chosen. For commercial telephone these rates are respectively, 8 KHz and 64 Kbits/sec, corresponding to a filtered signal bandwidth of 300 to 3300 Hz. A new processing concept (variable sampling and digital quantization) is proposed where a sample is coded with a single binary word. The code word corresponds to an information pair: a variable and adaptive sampling time and a coding angle associated to the signal. The basic principle is conceived around a distribution of the coded samples in amplitude and time (amplitude-time coding) along an adaptive coding curve associated to each coded/decoded sample of the original signal. A variable sampling rate requires a buffer, thus a delay, for transmission at a fixed rate. The transmitted signal so obtained is a transposition of the original speech signal and consequently its characteristics (bandwith, amplitude dynamic range) are modified. Some of the characteristics of the transmitted signal are ultimately used for the digital or even the analog transmission of the signal.