A method of speech signal compression, by variable rate coding of frames of digitized speech samples, comprising the steps of: determining a level of speech activity for a frame of digitized speech samples; selecting an encoding rate from a set of rates based upon said determined level of speech activity within said frame; coding said frame according to a predetermined coding format for said selected rate wherein each rate has a corresponding different coding format; providing for said frame a corresponding output data packet at said selected rate.

Variable rate vocoder

Apparatus and method for encoding speech using a codebook excited linear predictive (CELP) speech processor and an algebraic codebook for use therewith The CELP speech processor receives a digital speech input representative of human speech and performs linear predictive code analysis and perceptual weighting filtering to produce a short term speech information and a long term speech information The CELP speech processor utilizes an organized, non-overlapping, algebraic codebook containing a predetermined number of vectors, uniformly distributed over a multi-dimensional sphere to generate a remaining speech residual The short term speech information, long term speech information and remaining speech residual are combinable to form a quality reproduction of the digital speech input

Method and system for CELP speech coding and codebook for use therewith

An impulse sequence of a pitch frequency is detected from a phase-equalized prediction residual of an input speech signal S(t), and a quasi-periodic impulse sequence is obtained by processing the impulse sequence so that a fluctuation in its pitch frequency is within an allowed limit range. The magnitudes of the quasi-­periodic impulse sequence are so determined as to minimize an error between the waveform of a synthesized speech obtainable by exciting an all-pole filter (18) with the quasi-periodic impulse sequence and the waveform of a phase-equalized speech obtainable by applying the input speech signal to a phase equalizing filter (5). Preferably, the quasi-periodic impulse sequence is supplied to the all-pole filter after being applied to a zero filter (10) in which it is given features of the prediction residual of the speech. Coefficients of the zero filter are also determined so that the error of the waveforms of the synthesized speech and the phase-equalized speech is minimum.

Speech analysis-synthesis method and apparatus therefor

It is an objective of the present invention to provide an optimized method of selection of the encoding mode that provides rate efficient coding of the input speech. It is a second objective of the present invention to identify and provide a means for generating a set of parameters ideally suited for this operational mode selection. Third, it is an objective of the present invention to provide identification of two separate conditions that allow low rate coding with minimal sacrifice to quality. The two conditions are the coding of unvoiced speech and the coding of temporally masked speech. It is a fourth objective of the present invention to provide a method for dynamically adjusting the average output data rate of the speech coder with minimal impact on speech quality.

Method and apparatus for performing speech frame encoding mode selection in a variable rate encoding system

A speech coder and decoder methodology wherein pitch excitation and codebook excitation source energies (100) are represented by parameters that are readily transmissible with minimal transmission capacity requirements. The parameters are the long term energy value, a short term correction factor which is applied to the long term energy value to match the short term energy, and proportionality factor(s) that specify the relative energy contribution of the excitation sources to the short term energy value (101).

Digital speech coder having optimized signal energy parameters

A vector excitation coder efficiently com-presses vectors by using an optimum codebook designed off line, using an initial arbitrary codebook and a set of speech training vectors exploiting codevector sparsity (i.e., by making zero all but a selected number of samples of lowest amplitude in each of N codebook vectors). A fast-search method selects a number Nc of good excitation vectors from the code-book, where Nc is much smaller than N, and uses only the Nc vectors in an exhaustive search for the best match between a perceptually weighted input vector Zn~ and an estimate Zn derived from a codebook vector processed through long-term and short-terms filters, and a perceptual weighting filter. The zero input response of these cascaded filters is calculated and subtracted from an input speech vector sn after per-ceptual weighting to produce a vector rn. The code-book search operation is performed using

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Vector excitation speech or audio coder for transmission or storage

LPC spectral parameter encoding is often a challenging task in the development of low bit-rate speech coding systems. An efficient, low complexity method called Switched-Adaptive Interframe Vector Prediction (SIVP) has been developed for this purpose. SIVP utilizes vector linear prediction to exploit the high frame-to-frame redundancy present in successive frames of LPC parameters. When SIVP is combined with scalar quantization, we have found that the LPC parameter bit-rate required to achieve high-quality synthetic speech is only 1300 bits per second. With vector quantization, the bit-rate can be reduced even further (to 1000 hits per second) without introducing any perceivable quantization noise in the reconstructed

s9.9 ENCODING OF LPC SPECTRAL PARAMETERS USING SWITCHED-ADAPTIVE INTERFRAME VECTOR PREDICTION?

A speech compression project has recently been completed to develop a speech coding algorithm suitable for operation in a mobile satellite environment aimed at providing telephone quality natural speech at 4.8 kbps. The work has resulted in two alternative techniques which achieve reasonably good communications quality at 4.8 kbps while tolerating vehicle noise and rather severe channel impairments. The algorithms are embodied in a compact self-contained prototype consisting of two AT and T 32-bit floating-point DSP32 digital signal processors (DSP). A Motorola 68HC11 microcomputer chip serves as the board controller and interface handler. On a wirewrapped card, the prototype's circuit footprint amounts to only 200 sq cm, and consumes about 9 watts of power.

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Grant Davidson

Papers

Vector excitation speech or audio coder for transmission or storage

s9.9 ENCODING OF LPC SPECTRAL PARAMETERS USING SWITCHED-ADAPTIVE INTERFRAME VECTOR PREDICTION?

Speech coding at 4800 bps for mobile satellite communications