The ADM(adaptive delta modulation) speech coder is generally used with a time‐invariant low‐pass filter at the decoder output. The purpose of this low‐pass filter is to reject coder noise at frequencies above the fixed speech passband. The speech spectrum, however, tends to occupy only the lower frequencies within the passband during voiced speech, and is somewhat “high pass” during unvoiced speech. In this paper, we show how the quality of ADM may be significantly improved by adoptively filtering the coder output such as to follow the natural bandwidth of the speech. This was found to reduce drastically the perceived noise in the output of the ADM coding system at low bit rates. The use of an adaptive low‐pass filter realizes almost all of this quality gain. (An adaptive high‐pass filter seems to reject less audible noise components and seems more prone to introducing objectionable artifacts.) We also discuss a method for reducing the bit rate with little or no sacrifice in qualify (relative to normal ADM) by adapting the sampling rate along with the time‐varying low‐pass filter.
|Original language||English (US)|
|Number of pages||19|
|Journal||Bell System Technical Journal|
|State||Published - Jan 1 1981|
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