EP0119033B1 - Dispositif de codage de la parole - Google Patents

Dispositif de codage de la parole Download PDF

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Publication number
EP0119033B1
EP0119033B1 EP19840301302 EP84301302A EP0119033B1 EP 0119033 B1 EP0119033 B1 EP 0119033B1 EP 19840301302 EP19840301302 EP 19840301302 EP 84301302 A EP84301302 A EP 84301302A EP 0119033 B1 EP0119033 B1 EP 0119033B1
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EP
European Patent Office
Prior art keywords
signal
weighting
parameters
encoder
speech
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Expired
Application number
EP19840301302
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German (de)
English (en)
Other versions
EP0119033A1 (fr
Inventor
Gideon Abraham Senensieb
Anthony John Milbourn
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Prutec Ltd
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Prutec Ltd
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Publication date
Priority claimed from GB8306685A external-priority patent/GB2137054B/en
Application filed by Prutec Ltd filed Critical Prutec Ltd
Publication of EP0119033A1 publication Critical patent/EP0119033A1/fr
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/10—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a multipulse excitation

Definitions

  • This invention relates to a speech encoder, this being a circuit for converting a speech signal into a pulse train.
  • the pulse train may either be transmitted, encrypted, or stored and from it the original speech can be reproduced.
  • the linear predictor in Figure 1 is a recursive digital filter comprising a summation circuit 10 which has an input line 12 and an output line 14.
  • the output line 14 is connected to a shift register or to a tapped delay line 16 each tapping of which is fed back to the summation circuit by way of a respective multiplication circuit 18, to 18 n .
  • the output signal has a first component determined by the weighted summed outputs from the tappings of the delay line and a second component determined by the value of the input signal at that instant.
  • the first of these two components may be regarded as the predicted value based on previous values of the output signal and the second as the residual error. If the weighting parameters p, to p " of the circuits 18 are optimised then the residual error will be minimised.
  • the residual error if used as the excitation, yields perfect reproduction of the original speech.
  • the configuration of the vocal tract being due to physical movement of articulatory organs, can only change quite slowly.
  • the analogy between the configuration of the vocal tract and the weighting parameters allows much of the information in the speech signal to be transmitted at a low data rate. While this ensures good intelligibility, the quality and naturalness of the reproduced speech is largely dependent on the excitation signal used.
  • the parameters of the predictor are transmitted or stored and the excitation signal is selected either as white noise or as a regular series of pulses depending on the type of sound to be produced. Even using such crude simulation of the residual signal it was possible to produce recognisable speech. However, though the quality was acceptable for certain applications, for example military applications where maximum signal compression was of most importance, it fell below acceptable commercial standards.
  • the predictor should be excited by a train of pulses, in which the timing and the magnitude of each pulse in the train should be selected in order to minimise the difference between the re-synthesised speech and the original speech signal.
  • the excitation signal does not depend on the type of sound to be produced but for each frame the ideal excitation pulse train is computed.
  • the multi-pulse excitation signal, u(n), is a sequence of samples whose values are zero in all but a few positions.
  • the amplitudes and positions of the non-zero samples are chosen so as to minimise a perceptually meaningful error.
  • the linear predictor has a transfer function H(z), corresponding to an impulse response h(n).
  • the positions n k and the values u(n k ) are to be chosen so as to minimise the energy in the error sequence.
  • s(n) is the sequence of samples of original speech
  • w(n) is the impulse response corresponding to a spectral weighting function W(z)
  • * denotes convolution.
  • the problem is therefore to determine so as to minimise
  • an interative procedure can be adopted in which position and amplitude are evaluated for one non-zero sample at a time.
  • u(n k ) set to zero for k>j.
  • the sequence e(n) and values of u(n j ) for all possible n j must be recomputed at each iteration over the interval of interest.
  • the procedure can be refined by re-adjusting the amplitudes of all selected samples simultaneously, once their positions are all known.
  • the present invention is intended to encode speech using linear predictive coding in which the LPC filter is excited by a series of pulses whose positions and amplitude are capable of being computed in real time.
  • an encoder for encoding speech signals comprising means for sampling frames of the speech signal to be encoded, a linear prediction analyser for determining for each frame parameters of a linear predictor to minimise the residual signal for the sampled frame, and means for producing an excitation signal for transmission or storage in conjunction with the parameters to enable each frame of the speech signal to be resynthesised, characterised in that the encoder comprises a weighting filter with weighting parameters for damping reverberations within the speech signal caused by resonances in the vocal tract and a circuit for time weighting the parameters determined by the analyser by multiplication with a factor and in that the means for producing an excitation signal comprises correlating means for correlating the outputs of the weighting filter and the circuit.
  • a linear recursive filter if excited by a single pulse may have an impulse response of very long time duration and provided that it is not unstable will eventually decay rather than oscillate.
  • the effect of a long time response is that responses from consecutive excitation pulses tend to run into each other and it is difficult when performing a correlation to separate the pulse response of one excitation from another.
  • the speech signal is passed through a weighting filter, preferably a pole-zero filter, which has the effect of damping reverberations.
  • the weighting filter has a non-recursive part the weighting parameters of which are of the same magnitude as, but of opposite sign to, those of the linear predictor in the decoder.
  • the purpose of the non-recursive side of the weighting filter may regard the purpose of the non-recursive side of the weighting filter as negating the effect of the vocal tract on the pulses originally generated within the throat of the speaker.
  • the other side of the filter on the other hand, the recursive part, has weighting coefficients which are related to those of the linear predictor but are weighted by a factor which follows a power law of k", (k ⁇ 1), so that time-weighting of the impulse response is achieved.
  • the correlator will produce a high correlation output at the times when impulses should be applied to the linear prediction filter in order to simulate the speech signal.
  • the weighting filter is followed by a correlator of which the output is fed to an impulse selector.
  • the purpose of the impulse selector is to select from amongst the peaks of the output of the correlator a number of peaks having the highest magnitude. These peaks determine the time at which the residual signal should be applied to the linear predictor in the decoder in order to resynthesise the speech signal.
  • the peaks are selected such that they are all of the same polarity.
  • This polarity can be set so as to match the polarity of the microphone being used. If the polarities of the peak selection and microphone are correctly matched, then this improves the quality of the resynthesised speech by helping to preserve its harmonic content.
  • the excitation pulses should have an amplitude related to the amplitude of the - peak produced by the correlator. Because the auto-correlation functions of the pulse responses of the LPC filter are not constant but vary with the weighting parameters, it is preferred that the excitation pulse amplitude should be derived by dividing the correlator output by the value of the auto-correlation function of the impulse response of the filter with the prevailing time weighted parameters.
  • the speech signal to be encoded is received over an input line 30.
  • the input signal is applied to a known circuit 32 which is a linear prediction analyser.
  • This circuit computes the values of the weighting parameters of the digital recursive filter which would minimise the residual signal and outputs these parameters.
  • a linear prediction analyser more readily computes so called reflection co-efficients which are not the same as the weighting parameters but from which these parameters can be computed.
  • the reflection co-efficients are applied to a line 34.
  • the speech signal is also applied via a line 36 to a weighting filter 38 which will now be described by reference to Figure 3.
  • the weighting filter comprises an input line 40 connected to a summation circuit 42 having an output line 44.
  • a multi-tapped delay line (or shift register) 46 is connected to the input line 40 and a similar multi-tapped delay line 48 is connected to the output line 44.
  • the tappings of the delay line 46 are connected by way of a first set of weighting circuits 50 to the circuit 42 which also receives signals from the tappings of the delay line 48 through weighting circuits 52.
  • the values of the parameters used in the multiplication circuits of the weighting filter 38 in Figure 3 are derived from the linear prediction analyser 32.
  • the weighting parameters p, to Pn equivalent to the reflection coefficients are computed.
  • the coefficient weighting circuits 32 two sets of parameters are derived from the parameters p, to p " for setting the parameters of the weighting filter 38.
  • the first set of parameters is applied to the weighting circuits 50 and are equal to -p i to -p n .
  • the combination of the summation circuit with the delay line 46 and the weighting circuits 50 results in a digital non-recursive filter having parameters which are the opposite of those used in the receiving circuit to resynthesize the speech signal.
  • the effect of the non-recursive part of the weighting filter is to negate the effect of the vocal tract.
  • the second set of parameters evaluated by the coefficient weighting circuit 32 is equal to k - P1 to k" - Pn , where k is less than 1.
  • the delay line 48 and the weighting circuits 52 produce in conjunction with the summation circuit 42 a recursive digital filter whose pulse response is similar to that of the filter used to resynthesize the speech but with more rapid decay.
  • the effect of combination of the non-recursive and recursive filters which constitute the weighting filter 38, which is also termed a pole-zero filter, is to produce from the speech signal one in which reverberations are more severely damped to reduce the interaction between the effects of consecutive excitation pulses.
  • the output of the digital weighting filter 38 is applied to a correlator 64 connected to a circuit 66 which evaluates the impulse response of a digital recursive filter of the same construction as that shown in Fig. 1 but with weighting parameters k ⁇ p, to k" - p " .
  • the correlator 64 may consist of a shift register whose tapping are connected to multiplication circuits the multiplication factors of which are determined by the impulse response evaluating circuit 66. When there is a high level of correlation between the output of the weighting filter 38 and the impulse response evaluated by the circuit 66, a high output is produced by the correlator.
  • the output of the correlator 64 thus contains peaks which coincide with impulses in the excitation signal which, if applied to the linear predictor at the decoder, will cause a good approximation to the original speech signal to be produced.
  • the purpose of the pulse selector circuit 70 in Figure 2 is to select the timing of the pulses which are to be encoded.
  • One possible algorithm would be to disregard high values adjacent a local maximum or minimum if they are not separated from the local maximum or minimum by a zero crossing or a turning point.
  • Another possible algorithm is to select a fixed number of the greatest peaks in each time frame and to ensure that they are separated by at least some minimum time.
  • the amplitude of the selected pulses will be related to the amplitude of an optimal excitation signal.
  • the impulse response circuit 66 additionally evaluates the auto-correlation function of each pulse response and applies a signal over a line 72 to a divider circuit 74.
  • the selected pulses are divided by the auto-correlation value and the output signal from the divider is fed to a multiplexer 76 which encodes the reflection coefficient received over the line 34 and the signals from the divider 74 to produce the encoded signal on output line 78 for transmission or storage.
  • the preferred embodiment of the invention proposes making some simplifying assumptions in order to derive a modified algorithm which permits implementation in real-time of a 7.2 kbits/s vocoder using standard components on a double Eurosize circuit board.
  • ng is an arbitrary positive integer.
  • the approximation in (12) can be improved by increasing ng and/or by reducing r.
  • (12) can be applied to (5) to yield
  • sequence e o (n) is defined as where S o (n) is the output of the linear predictor driven with zero input.
  • the modified computation exploits an alternative interpretation of the role of the weighting function W(z).
  • the effect of the weighting function can be viewed as an attempt to separate the response of the system to successive non-zero excitation samples. If these samples are far enough apart, their values can be optimized independently.

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  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)

Claims (6)

1. Un encodeur pour encoder des signaux vocaux, comprenant un moyen d'échantillonnage de trames du signal vocal à encoder, un analyseur de prédiction linéaire (32) pour la détermination, pour chaque trame, des paramètres (Pn) d'un dispositif de prédiction linéaire (10, 16, 18) afin de minimiser le signal résiduel pour la trame échantillonnée, et un moyen (64, 70, 74, 76) de production d'un signal d'excitation à transmettre ou stocker en conjonction avec les paramètres, pour permettre de resynthétiser chaque trame du signal vocal, caractérisé en ce que l'encodeur comprend un filtre de pondération (38) avec des paramètres de pondération (-pn) pour l'amortissement des réverbérations dans le signal vocal causées par des résonances dans l'appareil vocal et un circuit (66) pour une pondération dans le temps des paramètres (pn) déterminés par l'analyseur (32) par multiplication avec un facteur (K"), et en ce que le moyen de production d'un signal d'excitation comprend un moyen de corrélation (64) afin de mettre en corrélation les sorties du filtre de pondération (38) et le circuit (66).
2. Un encodeur de signal selon la revendication 1, dans lequel le filtre de pondération (38) comprend un filtre des pôles et zéros.
3. Un encodeur de signal selon la revendication 1 ou 2, dans lequel le moyen de corrélation (64) comprend une ligne de retard à prises, un moyen pour multiplier les signaux captés par ladite réponse d'impulsion pondérée dans le temps, et un moyen pour additionner les sorties des circuits de multiplication.
4. Un encodeur de signal selon l'une quelconque des revendications précédentes, dans lequel la sortie du moyen de corrélation est reliée à un sélecteur d'impulsion (70) dont l'action vise à sélectionner un nombre d'impulsions issues de la sortie du corrélateur.
5. Un encodeur de signal selon la revendication 4, dans lequel le sélecteur d'impulsion comprend un moyen de détection des crêtes locales et un moyen de sélection, parmi les crêtes locales, de celles qui ont les amplitudes les plus positives ou les plus négatives.
6. Un encodeur de signal selon l'une quelconque des revendications précédentes, comprenant un moyen (66,74) de détermination de l'ampleur des impulsions transmises, en divisant la sortie du moyen de corrélation (64) par la fonction d'autocorrélation de ladite réponse d'impulsion pondérée dans le temps.
EP19840301302 1983-03-11 1984-02-28 Dispositif de codage de la parole Expired EP0119033B1 (fr)

Applications Claiming Priority (4)

Application Number Priority Date Filing Date Title
GB8306685 1983-03-11
GB8306685A GB2137054B (en) 1983-03-11 1983-03-11 Speech encoder
GB8333037 1983-12-10
GB8333037 1983-12-10

Publications (2)

Publication Number Publication Date
EP0119033A1 EP0119033A1 (fr) 1984-09-19
EP0119033B1 true EP0119033B1 (fr) 1987-04-15

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EP19840301302 Expired EP0119033B1 (fr) 1983-03-11 1984-02-28 Dispositif de codage de la parole

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CA (1) CA1202419A (fr)
DE (1) DE3463192D1 (fr)

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* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4472832A (en) * 1981-12-01 1984-09-18 At&T Bell Laboratories Digital speech coder

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DE3463192D1 (en) 1987-05-21
CA1202419A (fr) 1986-03-25
EP0119033A1 (fr) 1984-09-19

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