Mark A. Ireton

System and method for enhanced speech quality in voice storage and retrieval systems

A digital voice data storage and retrieval system using a low bit rate encoder which provides enhanced speech signal quality while also reducing memory size requirements. The system comprises a voice coder/decoder which preferably includes a digital signal processor (DSP) and also preferably includes a local memory. During encoding of the voice data, the voice coder/decoder receives voice input waveforms and generates a parametric representation of the voice data. A storage memory is coupled to the voice coder/decoder for storing the parametric data. During decoding of the voice data, the voice coder/decoder receives the parametric data from the storage memory and reproduces the voice waveforms. According to the invention, an interframe smoothing method is performed on the parametric data after encoding of all of the speech data has completed and the parametric data has been stored in the storage memory. The interframe smoothing is performed either in the background after the coding process has completed or in real time during the decoding process immediately prior to converting the parametric data back to signal waveforms. Since all of the voice input data has already been converted to parametric data and stored in memory, parametric data from a virtually unlimited number of prior and successive frames is available for use by the smoothing algorithm. Therefore, the present invention provides more accurate smoothing and provides enhanced speech signal quality over prior systems.

Detection of tonal signals

The system and method of the present invention uses a zero-crossing rate measurement in order to determine the initiation and/or termination of speech in an audio signal input. It is especially well suited for detecting the termination of a telephone message in a telephone answering device. Specifically, a sample of the zero-crossing rate signal is determined by counting the number of consecutive speech samples required for the occurrence of a pre-defined number of consecutive zero-crossings. The resultant zero-crossing rate signal is smoothed and applied to a differentiator. A short-time magnitude integration is performed to measure the energy in the differentiated signal. The output of the magnitude integration is provided to a threshold detector which produces a sequence of decision values indicating the presence or absence of speech. Finally, the decision values are filtered to produce a more definitive sequence of final decision values.

System and method for improved pitch estimation which performs first formant energy removal for a frame using coefficients from a prior frame

An improved vocoder system and method for estimating pitch in a speech waveform which pre-filters speech data with improved efficiency and reduced computational requirements. The vocoder system is preferably a low bit rate speech coder which analyzes a plurality of frames of speech data in parallel. Once the LPC filter coefficients and the pitch for a first frame have been calculated, the vocoder then looks ahead to the next frame to estimate the pitch, i.e., to estimate the pitch of the next frame. In the preferred embodiment of the invention, the vocoder filters speech data in a second frame using a plurality of the coefficients from a first frame as a multi pole analysis filter. These coefficients are used as a crude two pole analysis filter. The vocoder preferably includes a first processor which performs coefficient calculations for the second frame, and a second processor which performs pre-filtering and pitch estimation, wherein the second processor operates substantially simultaneously with the first processor. Thus, the vocoder system uses LPC coefficients for a first frame as a crude multi pole analysis filter for a subsequent frame of data, thereby performing pre-filtering on a frame without requiring previous coefficient calculations for that frame. This allows pre-filtered pitch estimation and LPC coefficient calculations to be performed in parallel. This provides a more efficient pitch estimation, thus enhancing vocoder performance.

System and method for performing predictive scaling in computing LPC speech coding coefficients

A system and method for calculating linear predictive coding (lpc) coefficients in response to a received speech signal waveform. The system of the present invention uses a method referred to as the FLAT method for computing the lpc coefficients. The FLAT method involves computing 10 lpc filter coefficients and proceeds in a succession of iterations, wherein the 10 filter coefficients are derived in part from 10 reflection coefficients. According to the present invention, the method determines or predicts a scaling factor and, during each iteration, applies the scaling factor to the matrix terms prior to storage of the data while the full precision value is still available internally. Thus the present invention maintains full precision for the speech data during the computation. The scale factor prediction method of the present invention is most effective for high power voiced speech where the greatest loss of precision occurs using prior art post-storage scaling techniques.

Apparatus and method for analyzing speech signals to determine parameters expressive of characteristics of the speech signals

An apparatus and method for locating a plurality of roots of a line spectrum pair expression on a unit circle. The method comprises the steps of: (1) receiving an initial value for locating a first site on the unit circle; (2) receiving a step value for defining an arc-distance on the unit circle; (3) generating intervals on the unit circle, each having a lower limit and an upper limit; the lower limit of the initial interval is the initial value and the upper limit of the initial interval is displaced on the unit circle from the initial value by the arc-distance; each succeeding interval has its lower limit coincident with the upper limit of the next preceding interval and has its upper limit displaced on the unit circle from its lower limit by the arc-distance; (4) evaluating the expression for at least the upper limit and the lower limit of each respective interval; (5) recognizing a root of the expression when the expression changes sign within an interval; (6) designating each such interval as a solution interval; and (7) generating the lower and upper limits of each solution interval to identify where the root is located. The apparatus comprises a waveform generator which receives the initial value and the step value and defines the arc-distance, and generates the intervals; a zero detector for recognizing roots of the expression when the expression changes sign.