Adaptive filter and filtering method for low bit rate coding

A filtering method for enhancing digitally processed speech or audio signals comprising the steps of:

• buffering said speech or audio signals into frames of vectors, each vector having K successive samples;
• performing analysis of said buffered frames of speech or audio signals in predetermined blocks to compute linear predictive coefficients and signal power in the current frame;
• generating a signal probability estimator value sig-prob based on comparison of the signal power in the current frame to a long term estimate of the noise power;
• first filtering each vector by a delay controlled by said linear predictive coefficient and said signal probability estimator value, wherein filtering is accomplished by using a transfer function of the form ##EQU6## where 1-P is the LPC coefficient, z is the inverse of the unit delay operator used in the transform representation of the transfer functions, .alpha. and .beta. are scaling factors * sig-prob;
• and
• second filtering by the transfer function of the form 1-.mu.z.sup.-1 * sig-prob, where .mu.=scaling factor.

A low bit rate speech communication system for transmitting speech signals comprising:

• means for buffering said speech signals into frames of vectors, each vector having successive samples;
• means for performing analysis of said buffered frames of speech or audio signals in predetermined blocks to compute encoded speech including linear predictive coefficients and power in the current frame;
• means for transmitting said encoded speech over a transmission channel,
• a synthesizer coupled to said means for transmitting and responsive to said encoded speech for decoding said speech into digital signals;
• a digital to analog converter means responsive to said digital signals from said synthesizer for providing speech signals,
• said synthesizer comprising means for enhancing digitally processed speech comprising:
• means for generating a signal probability estimator value sig-prob based on comparison of the power in the current frame to a long term estimate of the noise power;
• first filter means for filtering each vector by a delay controlled by said linear predictive coefficient and said signal probability estimator value, wherein filtering is accomplished by using a transfer function of the form ##EQU7## where 1-P is the LPC coefficients, z is the inverse of the unit delay operator used in the transform representation of the transfer functions, .alpha. and .beta. are scaling factors;
• and
• second filter means for filtering by the transfer function of the form 1-.mu.z.sup.-1 * sig-prob, where .mu.=scaling factor.

A filter for improving digitally processed speech signals comprising:

• means for generating a signal probability estimator value based on a comparison of signal power of said signals in a current frame to a long term estimate of noise power;
• a first filter for filtering said signals controlled by linear predictive coefficients and said signal probability value;
• and
• a second filter having the transfer function of the form 1-.mu.z.sup.-1 * signal probability value where .mu. is a scaling factor, and z.sup.-1 is a unit delay factor.