Abstract: A low-rate encoding/digital speech interpolation (LRE/DSI) system architecture is organized to eliminate the LRE mistracking problem normally associated with DSI-induced speech clipping, to reduce the number of LRE codes required and to provide a mechanism for gradually introducing overload.

Abstract: An apparatus for detecting the modulation rate of voice-band data signals. The detector determines the level occupancy L.sub.p, the partial entropy E.sub.p, and the zero-crossing count N.sub.zc for the input signal and compares these values with threshold values to determine the data rate.

Abstract: An adaptive gain control is provided at the input of a Variable Rate Nearly Instantaneous Companding (VRNIC) processor to adjust the gain of low level speech signals so that the transmission bit rate and SNR will remain substantially the same for low signal levels as for high signal levels.

Abstract: A variable slope delta coding processor using adaptive prediction in which a digitized signal is transmitted with approximately half the channel bandwidth required by previous techniques without any substantial sacrifice of received signal quality. An estimate of a digital sample is subtracted from a digital sample of an input signal to produce an error signal. The sign bit of the error signal is transmitted as a single bit output signal. A predetermined number of immediately adjacent single bit output signals are utilized to calculate an adaptive step value using a compression technique. A set of reconstructed signal values is built up, one for each of the input signal, with each reconstructed signal value being calculated as the sum of the adaptive step size value and the estimate of the sample of the input signal. For each new sample of the input signal, an oldest reconstructed signal value is discarded and replaced by a newly calculated value.

Abstract: A nearly instantaneous companding processor uses variable precision block quantization for reducing the digital transmission rate of speech signals while at the same time maintaining a high quality signal in a digital communication system. The variable number of bits which are used to encode each sample is controlled as a function of the maximum sample magnitude in a block of N samples. This technique takes advantage of the amplitude probability density function of a speech signal to thereby reduce the quantizing noise without increasing the transmission rate.

Abstract: An echo canceller having a digital transversal filter with an adaptive control loop for achieving minimum echo is disclosed. Minimum echo is attained by subtracting a synthesized echo from the real echo, the synthesized echo being formed in a digital transversal filter by multiplying a stored replica of the impulse response times the incoming signal. The stored replica is updated using the steepest descent technique by adjusting each of the stages of the replica memory a given amount. The incoming signal is differential pulse code modulation encoded to reduce the process complexity and the size of the memories which store the replica and the incoming signals or, alternatively, increase the performance of the canceller.