Variable rate encoding using adaptive delta modulation
Abstract
In this paper a variable sampling frequency adaptive delta modulation (ADM) system intended for audio signal encoding is presented. The objective is to reduce the channel bit rate required by other delta modulation systems, while maintaining the same quality of signal reproduction. The ADM rate is selected by first measuring the slope during a time segment of the audio signal. The slope measure is mapped to an ADM rate via a set of signal-to-noise (SNR) curves and a specified performance level. A probabilistic analysis of the slope and its comparison with simulation results is presented. Using a composite test signal to simulate a voice waveform, the system performance has been studied as a function of the observation time segment. SNR curves are presented and it is shown that a bandwidth compression of at least 30% can be achieved over a conventional ADM encoder.
- Publication:
-
16th Annual Allerton Conference on Communication, Control and Computing
- Pub Date:
- 1979
- Bibcode:
- 1979ccc..proc..294P
- Keywords:
-
- Adaptive Control;
- Delta Modulation;
- Signal Encoding;
- Signal To Noise Ratios;
- Voice Communication;
- Data Compression;
- Data Sampling;
- Digital Simulation;
- Performance Prediction;
- Probability Theory;
- System Effectiveness;
- Transmission Efficiency;
- Communications and Radar