Method of and device for speech signal coding and decoding by means of a multipulse excitation

Claim 2

1 - A method of speech signal coding and decoding, using a multipulse analysis-by-synthesis excitation technique, which method comprises a coding phase including the following operations: - speech signal conversion into frames of digital samples [s(n)]; - short-term analysis of the speech signal, to determine a group of linear prediction coefficients [a(k)] relevant to a current frame and a representation thereof as line spectrum pairs; - coding of said representation of the linear prediction coefficients; - spectral shaping of the speech signal, by weighting the digital samples [s(n)] in a frame by a first and a second weighting functions [A(z); 1/A(z,γ)], the weighting by the first weighting function generating a residual signal [r(n)], which is then weighted by the second function to generate a spectrally-weighted speech signal [s w (n)]; - long-term analysis of the speech signal, by using said residual signal [r(n)] and said spectrally weighted signal [s w (n)], to determine the lag (M) separating a current sample from a preceding sample [r(n)-M)] used to process said current sample, and the gain (B) by which said preceding sample is weighted for the processing; - determination of the positions and amplitudes of the excitation pulses, by exploiting the results of short-term and long-term analysis, said determination being performed in closed loop as a part of the procedure by which excitation pulse positions are determined; - coding of the values of said lag and gain of long-term analysis and of said amplitudes and position of the excitation pulses, the coded values forming, jointly with the coded representation of the linear prediction coefficients and with coded r.m.s. values of said excitation pulses, the coded speech signal; and also comprising a decoding phase, where the excitation is reconstructed starting from the coded values of the amplitudes, the positions and the r.m.s. values of the pulses and where, by using the reconstructed excitation (ê, a synthesized speech signal [ŝ(n)] is generated, by means of a long-term synthesis filtering followed by a short-term synthesis filtering, which filterings exploit the long-term analysis parameters and respectively the quantized linear prediction coefficients, characterized in that said long-term analysis and excitation pulse generation are performed in successive steps, in the first of which long-term analysis gain (B) and lag (M) are determined by minimizing a mean squared error between the spectrally-shaped speech signal [s w (n)] and a further signal [s w0 (n)] obtained by weighting by said second weighting function the signal resulting from a long-term synthesis filtering, which is similar to that performed during decoding and in which the signal used for the synthesis is a null signal, while in the second step the amplitudes and positions of the excitation pulses [e(i)] are actually determined by minimizing the mean squared error between a signal [s we (n)] representing the difference between the spectrally-shaped speech signal [s w (n)] and said further signal [s w0 (n)], and a third weighted signal [ŝ we (n)], obtained by submitting the excitation pulses to a long-term synthesis filtering and to a weighting by said second weighting function, and in that the coding of said representation of the linear prediction coefficients consists in a vector quantization of the line spectrum pairs or of the adjacent line pair differences according to a split-codebook quantization technique.

Claim 3

A method as claimed in claim 1, characterized in that the lag (M) and the gain (B) are determined in two successive steps, in the first of which an optimum value of the lag is determined by minimizing said error for a predetermined gain value, while in the second the optimum gain value is determined, by using said optimum lag value.

Claim 4

A method as claimed in claim 1, characterized in that the lag (M) and the gain (B) are determined in two successive steps, in the first of which the mean squared error is minimized between the residual signal [r(n)] and a signal [ ₀(n)] which is the signal [r₀(n)] resulting from said long-term synthesis filtering with null input, if the synthesis relevant to a sample of the current frame is performed on the basis of a sample of a preceding frame, and is said residual signal [r(n)] if the synthesis relevant to a sample of the current frame is performed on the basis of a preceding sample of the same frame, while in the second step the gain (B) is calculated with the following sequence of operations: a value [s′ w0 (n)] of said further signal is determined for a unitary gain value; a first error value E(M,1) is hence determined, and the operations for determining the value of the signal weighted with said second weighting function and of the error are repeated for each value possible for the gain, the value adopted being the one which minimizes the error.

Claim 5

A method as claimed in claim 3, characterized in that the lag (M) is computed every frame, and the gain (B) every half frame.

Claim 6

Device for speech signal coding and decoding by multipulse analysis-by-synthesis excitation techniques, for implementing the method as claimed in any of claims 1, 3 or 4, comprising, for speech signal coding: - means for converting the speech signal into frames of digital samples [s(n)]; - means (STA) for the short-term analysis of the speech signal, which means receive a group of samples from said converting means, compute a set of linear prediction coefficients [a(k)] relevant to a current frame and emit a representation of said linear prediction coefficients [a(k)] as line spectrum pairs; - means (VQ) for coding said representation of the linear prediction coefficients; - means (DCO) for obtaining quantized linear prediction coefficients [â(k)] from said coded representation; - a circuit (SW) for the spectral shaping of the speech signal, connected to the converting means and to the means (DCO) obtaining the quantized linear prediction coefficients and comprising a pair of cascaded weighting digital filters (F1, F2), weighting the digital samples [s(n)] according to a first and a second weighting function [A(z); 1/A(z,γ)], respecively, said first filter (F1) supplying a residual signal; - means (LTA) for the long-term analysis of the speech signal, connected to the outputs of said first filter (F1) and of the spectral shaping circuit (SW) to determine the lag (M) which separates a current sample from a preceding sample [r(n-M)], used to process said current sample, and the gain (B) by which said preceding sample is weighted for the processing; - an excitation generator (EG) for determining the positions and the amplitudes of the excitation pulses, connected to said short-tem and long-term analysis means (STA, LTA) and to said spectral shaping circuit (SW); - means (LTC, PAC) for coding the values of said long-term analysis lag and gain and excitation pulse positions and amplitudes, the coded values forming, jointly with the coded representation of the linear prediction coefficients and with r.m.s. values of said excitation pulses, the coded speech signal; and also comprising, for speech signal decoding (synthesis): - means (ED, LTD, STD) for reconstructing the excitation, the long-term analysis lag (M) and gain (B) and the linear prediction coefficients [a(k)] starting from the coded signal; and - a synthesizer, comprising the cascade of a first long-term synthesis filter (LTP1), which receives the reconstructed excitation pulses, gain and lag and filters them according to a first transfer function [1/P(z)] dependent on said gain and lag, and a short-term synthesis filter (STP) having a second transfer function [1/A(z)] which is the reciprocal of said first spectral weighting function [A(z)], characterized in that the long-term analysis means (LTA) are apt to determine said lag (M) and gain (B) in two successive steps, preceding a step in which the amplitudes and positions of the excitation pulses are determined by said excitation generator (EG), and comprise: - a second long-term synthesis filter (LTP2), which is fed with a null signal and in which, for the computation of the lag (M), there is used a predetermined set of values of the number of samples separating a current sample being synthesized from a previous sample used for the synthesis, and, for the computation of the gain (B), a predetermined set of possible values of the gain itself is used; - a multipleser (MX1) receiving at a first input a residual signal sample and at a second input a sample of the output signal of the second long-term synthesis filter (LTP2) and supplies the samples present at either input depending on whether or not said number of samples is lower than a frame length; - a third weighting filter (F3), which has the same transfer function as said second digital filter (F2) of the spectral shaping circuit (SW), is connected to the output of said second long-term synthesis filter (LTP2) and is enabled only during the determination of the long-term analysis gain (B); - a first adder (SM1), which receives at a first input the spectrally-shaped signal (s w ) and at a second input the output signal of said third weighting filter (F3) and supplies the difference between the signals present at its first and second input; - a first processing unit (CMB), which receives in said first operation step the signal outgoing from said multiplexer (MX1) and determines the optimum value of said number of samples, and in a second operation step receives the output signal of said first adder (SM1) and determines, by using the lag computed in the first step, the value of the gain which minimizes the mean squared error, within a validity period of the excitation pulses, between the input signals of the first added (SM1); and in that the excitation generator (EG) for generating the excitat­ion pulses [e(i)] comprise: - a third long-term synthesis filter (LTP3), which has the same transfer function as the first one and is fed with the excitation pulses generated; - a fourth weighting filter (F4), connected to the output of the third synthesis filter (LTP3) and having the same transfer function as said second and third weighting filters (F2, F3); - a second adder (SM2), which receives at a first input the output signal of said first adder (SM1) and at a second input the output signal of the fourth weighting filter (F4), and supplies the difference between the signals present at its first and second input; - a second processing unit (CE) which is connected to the output of said second adder and determines the amplitudes and positions of said pulses by minimizing the mean squared error, within a pulse validity period, between the input signals of the fourth adder (SM2).

Claim 7

A device as claimed in claim 5 characterized in that the means (VQ) coding said representation of the linear prediction coefficient consist of a vector quantizer (VQ) for split-codebook vector quantization of the line spectrum pairs or of the differences between adjacent line spectrum pairs.