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Performance of an ADPCM/TASI system for PCM speech compression

1981; Institute of Electrical and Electronics Engineers; Volume: 29; Issue: 2 Linguagem: Inglês

10.1109/tassp.1981.1163525

0096-3518

S. Townes, K. Kou, J.P. Agrawal, J. O'Neal, D. Mobedjina,

Advanced Adaptive Filtering Techniques

This paper contains a summary of performance measurements for a digital speech interpolation (DSI) system called adaptive differential PCM with time assignment speech interpolation (ADPCM/ TASI). This system is used to compress the output of two or more T1 PCM terminals into a bit rate normally requited by one such terminal. The performance measurements presented are as follows. 1) Signal-to-noise power ratios (SNR) for a 1004 Hz tone signal measured by a transmission impairment measuring test set (TIMS). 2) Idle channel noise performance measured by a TIMS. 3) SNR for speech measured by actual computation. These measurements were made for a variety of traffic loading conditions, and a number of different input power levels. The ADPCM/TASI system was constructed by using an AMD 2900 bit slice microprocessor architecture, but was designed to operate on only one speech channel. The presence of other active channels was simulated programatically. The performance of the system when voice band data signals occupy some of the T1 channels is discussed and is reflected in the measurements. An audio tape was made to demonstrate the subjective performance of the system with and without errors in the transmission medium. The objective performance of the system is compared to the D3 channel bank specifications.