Showing papers on "Enhanced Variable Rate Codec published in 1989"

Wear-toll quality 4.8 kbps speech codec

Abstract: A speech codec operating at low data rates uses an iterative method to jointly optimize pitch and gain parameter sets. A 26-bit spectrum filter coding scheme may be used, involving successive subtractions and quantizations. The codec may preferably use a decomposed multipulse excitation model, wherein the multipulse vectors used as the excitation signal are decomposed into position and amplitude codewords. Multipulse vectors are coded by comparing each vector to a reference multipulse vector and quantizing the resulting difference vector. An expanded multipulse excitation codebook and associated fast search method, optionally with a dynamically-weighted distortion measure, allow selection of the best excitation vector without memory or computational overload. In a dynamic bit allocation technique, the number of bits allocated to the pitch and excitation signals depend on whether the signals are "significant" or "insignificant". Silence/speech detection is based on an average signal energy over an interval and a minimum average energy over a predetermined number of intervals. Adaptive post-filter and the automatic gain control schemes are also provided. Interpolation is used for spectrum filter smoothing, and an algorithm is provided for ensuring stability of the spectrum filter. Specially designed scalar quantizers are provided for the pitch gain and excitation gain.

Speech codec for the European mobile radio system

Abstract: The speech coding scheme which will be used as the standard for the European mobile radio system has been selected by the CEPT Groupe Special-Mobile (GSM) as a result of formal subjective listening tests. It is based on the regular-pulse excitation linear predictive coding technique (RPE-LPC) combined with long-term prediction (LTP). The solution is called the RPE-LTP codec. The codec algorithm and the error protection scheme are presented. The net bit rate is 13.0 kb/s, and the gross bit rate, including error protection, is 22.8 kb/s. The experimental implementation based on VLSI signal processors is described. The speech quality obtained with the technique considered is far superior to that obtainable with present-day analog mobile radio systems. A duplex speech codec including error protection can be implemented with two VLSI sign processors with external data memories of about 1 K*16 b. >

A Multiprocessor Based Low Bitrate Video Codec Architecture and a Well-Adapted Two Phase Software Coding Strategy

Abstract: The first part of the paper describes a specialized multiprocessor environment for hybrid coding of visual communications signals in the range from ISDN basic access to primary rate transmission channels. Most important is a proprietary 4-wide SIMD parallel video processor with 80 MIPS. The second part deals with the software philosophy of the codec. It uses preanalysis and prebuffering in the first phase of coding a frame. In the second phase, limited processing power and available channel bits are distributed optimally over time and over changed areas of one frame. Codec delay is halved with respect to conventional codecs.

Testing and Evaluation of Low Data Rate Voice CODEC Equipment, Phase 2

Abstract: : The FAA is currently evaluating low rate voice digitizing coder/ decoder (CODEC's) for use with the Aeronautical Mobile Satellite Service (AMSS). Phase II of this evaluation consisted of air traffic control (ATC) personnel participating in an objective intelligibility test of several CODEC's under operational conditions. The results of the testing show that the intelligibility of the low rate 4.8 kilobits per second (kbps) CoDEC's is essentially equivalent to the intelligibility of the 9.6 kbps CODEC. The results also show that the 4.8 kbps CODEC's can operate with high intelligibility under conditions of high bit error rates and operational background noise. Keywords: Air traffic controllers; Digital voice communications; Voice coding.