EP0450064B1 - Codeur de parole numerique a predicteur a long terme ameliore a resolution au niveau sous-echantillon - Google Patents

Codeur de parole numerique a predicteur a long terme ameliore a resolution au niveau sous-echantillon Download PDF

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EP0450064B1
EP0450064B1 EP91905041A EP91905041A EP0450064B1 EP 0450064 B1 EP0450064 B1 EP 0450064B1 EP 91905041 A EP91905041 A EP 91905041A EP 91905041 A EP91905041 A EP 91905041A EP 0450064 B1 EP0450064 B1 EP 0450064B1
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speech
vector
samples
circuitry
filter
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EP0450064A1 (fr
EP0450064B2 (fr
EP0450064A4 (en
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Ira Alan Gerson
Mark A. Jasiuk
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Motorola Solutions Inc
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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
    • G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
    • 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
    • G10L2019/0001—Codebooks
    • G10L2019/0011—Long term prediction filters, i.e. pitch estimation
    • 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
    • G10L2019/0001—Codebooks
    • G10L2019/0012—Smoothing of parameters of the decoder interpolation

Definitions

  • the present invention generally relates to digital speech coding at low bit rates, and more particularly, is directed to an improved method for determining long-term predictor output responses for code-excited linear prediction speech coders.
  • Code-excited linear prediction is a speech coding technique which has the potential of producing high quality synthesized speech at low bit rates, i.e., 4.8 to 9.6 kilobits-per-second (kbps).
  • This class of speech coding also known as vector-excited linear prediction or stochastic coding, will most likely be used in numerous speech communications and speech synthesis applications.
  • CELP may prove to be particularly applicable to digital speech encryption and digital radiotelephone communication systems wherein speech quality, data rate, size, and cost are significant issues.
  • code-excited or "vector-excited” is derived from the fact that the excitation sequence for the speech coder is vector quantized, i.e., a single codeword is used to represent a sequence, or vector, of excitation samples. In this way, data rates of less than one bit per sample are possible for coding the excitation sequence.
  • the stored excitation code vectors generally consist of independent random white Gaussian sequences. One code vector from the codebook is chosen to represent each block of N excitation samples. Each stored code vector is represented by a codeword, i.e., the address of the code vector memory location. It is this codeword that is subsequently sent over a communications channel to the speech synthesizer to reconstruct the speech frame at the receiver. See M.R.
  • CELP Code-Excited Linear Prediction
  • the excitation code vector from the codebook is applied to two time-varying linear filters which model the characteristics of the input speech signal.
  • the first filter includes a long-term predictor in its feedback loop, which has a long delay, i.e., 2 to 15 milliseconds, used to introduce the pitch periodicity of voiced speech.
  • the second filter includes a short-term predictor in its feedback loop, which has a short delay, i.e., less than 2 msec, used to introduce a spectral envelope or format structure.
  • the speech coder applies each individual code vector to the filters to generate a reconstructed speech signal, and compares the original input speech signal to the reconstructed signal to create an error signal.
  • the error signal is then weighted by passing it through a weighting filter having a response based on human auditory perception.
  • the optimum excitation signal is determined by selecting the code vector which produces the weighted error signal having the minimum energy for the current frame.
  • the codeword for the optimum code vector is then transmitted over a communications channel.
  • the codeword received from the channel is used to address the codebook of excitation vectors.
  • the single code vector is then multiplied by a gain factor, and filtered by the long-term and short-term filters to obtain a reconstructed speech vector.
  • the gain factor and the predictor parameters are also obtained from the channel. It has been found that a better quality synthesized signal can be generated if the actual parameter used by the synthesizer are used in the analysis stage, thus minimizing the quantization errors.
  • the use of these synthesis parameters in the CELP speech analysis stage to produce higher quality speech is referred to as analysis-by-synthesis speech coding.
  • the predictor order p typically ranges from 8 to 12, assuming an 8 kiloHertz (kHz) sampling rate.
  • the weights ⁇ 1 , ⁇ 2 , ⁇ p , in this equation are called the predictor coefficients.
  • the short-term predictor coefficients are determined from the speech signal using conventional linear predictive coding (LPC) techniques.
  • LPC linear predictive coding
  • the output response of the short-term filter may be expressed in Z-transform notation as:
  • the long-term filter must predict the next output sample from preceding samples that extend over a much longer time period. If only a single past sample is used in the predictor, then the predictor is a single-tap predictor. Typically, one to three taps are used.
  • the output response for a long-term filter incorporating a single-tap, long-term predictor is given in z-transform notation as:.
  • this response is a function of only the delay or lag L of the filter and the filter coefficient ⁇ .
  • the lag L would typically be the pitch period of the speech, or a multiple of it.
  • a suitable range for the lag L would be between 16 and 143, which corresponds to a pitch range between 56 and 500 Hz.
  • the long-term predictor lag L and long-term predictor coefficient ⁇ can be determined from either an open-loop or a closed loop configuration. Using the open-loop configuration, the lag L and coefficient ⁇ are computed from the input signal (or its residual) directly. In the closed loop configuration, the lag-L, and the coefficient ⁇ are computed at the frame rate from coded data representing the past output of the long-term filter and the input speech signal. In using the coded data, the long-term predictor lag determination is based on the actual long-term filter state that will exist at the synthesizer. Hence, the dosed-loop configuration gives better performance than the open-loop method, since the pitch filter itself is would be contributing to the optimization of the error signal. Moreover, a single-tap predictor works very well in the closed-loop configuration.
  • the frame length N would be equal to 60 samples. This means only 32 past samples would be available to predict the next 60 samples of the frame. hence, if the long-term predictor lag L is less than the frame length N, only L past samples of the required N samples are defined.
  • a third solution is to reduce the size of the frame length N.
  • the long-term predictor lag L can always be determined from past samples. This approach, however, suffers from a severe bit rate penalty.
  • a shorter frame length a greater number of long-term predictor parameters and excitation vectors must be coded, and accordingly, the bit rate of the channel must be greater to accommodate the extra coding.
  • the sampling rate used in the coder places an upper limit on the performance of a single-tap pitch predictor. For example, if the pitch frequency is actually 485 Hz, the closest lag value would be 16 which corresponds to 500 Hz. This results in an error of 15 Hz for the fundamental pitch frequency which degrades voice quality. This error is multiplied for the harmonics of the pitch frequency causing further degradation.
  • the optimum solution must address both the problems of computational complexity and voice quality for the coding of high-pitched speech. degrades voice quality. This error is multiplied for the harmonics of the pitch frequency causing further degradation.
  • the optimum solution must address both the problems of computational complexity and voice quality for the coding of high-pitched speech.
  • a general object of the present invention is to provide an improved digital speech coding technique that produces high quality speech at low bit rates.
  • a more specific object of the present invention is to provide a method to determine long-term predictor parameters using the closed-loop approach.
  • Another object of the present invention is to provide an improved method for determining the output response of a long-term predictor in the case when the long-term predictor lag parameter L is a non-integer number.
  • a further object of the present invention is to provide an improved CELP speech coder which permits joint optimization of the gain factor gamma and the long-term predictor coefficient beta during the codebook search for the optimum excitation code vector.
  • FIG. 1 there is shown a general block diagram of code excited linear predictive speech coder 100 utilizing the long-term filter in accordance with the present invention.
  • An acoustic input signal to be analyzed is applied to speech coder 100 at microphone 102.
  • the input signal typically a speech signal, is then applied to filter 104.
  • Filter 104 generally will exhibit bandpass filter characteristics. However, if the speech bandwidth is already adequate, filter 104 may comprise a direct wire connection.
  • the analog speech signal from filter 104 is then converted into a sequence of N pulse samples, and the amplitude of each pulse sample is then represented by a digital code in analog-to-digital (A/D) converter 108, as known in the art.
  • the sampling rate is determined by sample dock SC, which represents an 8.0 kHz rate in the preferred embodiment.
  • the sample clock SC is generated along with the frame clock FC via clock 112.
  • A/D 108 which may be represented as input speech vector s(n)
  • This input speech vector s(n) is repetitively obtained in separate frames, i.e., blocks of time, the length of which is determined by the frame clock FC.
  • LPC linear predictive coding
  • the short-term predictor parameters ⁇ i , long-term predictor coefficient ⁇ , nominal long-term predictor lag parameter L, weighting filter parameters WFP, and excitation gain factor ⁇ (along with the best excitation codeword I as described later) area applied to multiplexer 150 and sent over the channel for use by the speech synthesizer.
  • the input speech vector s(n) is also applied to subtractor 130 the function of which will subsequently be described.
  • Codebook ROM 120 contains a set of M excitation vectors u i (n),. wherein 1 ⁇ i ⁇ M, each comprised of N samples, wherein 0 ⁇ n ⁇ N-1 . Codebook ROM 120 generates these pseudorandom excitation vectors in response to a particular one of a set of excitation codewords i. Each of the M excitation vectors are comprised of a series of random white Gaussian samples, although other types of excitation vectors may be used with the present invention. If the excitation signal were coded at a rate of 0.2 bits per sample for each of the 60 samples, then there would be 4096 codewords i corresponding to the possible excitation vectors.
  • Gain block 122 For each individual excitation vector u i (n), a reconstructed speech vector s' i (n) is generated for comparison to the input speech vector s(n).
  • Gain block 122 scales the excitation vector u i (n) by the excitation gain factor ⁇ , which is constant for the frame.
  • the excitation gain factor ⁇ may be pre-computed by coefficient analyzer 110 and used to analyze all excitation vectors as shown in Figure 1, or may be optimized jointly with the search for the best excitation codeword I and generated by codebook search controller 140.
  • the scaled excitation signal ⁇ u i (n) is then filtered by longter filter 124 and short-term filter 126 to generate the reconstructed speech vector s' i (n).
  • Filter 124 utilizes the long-term predictor parameters ⁇ and L to introduce voice periodicity
  • filter 126 utilizes the short-term predictor parameters ⁇ i to introduce the spectral envelope, as described above.
  • Long-term filter 124 will be described in detail in the following figures. Note that blocks 124 and 126 are actually recursive filters which contain the long-term predictor and short-term predictor in their respective feedback paths.
  • the reconstructed speech vector s' i (n) for the i-th excitation code vector is compared to the same block of the input speech vector s(n) by subtracting these two signals in subtractor 130.
  • the difference vector e i (n) represents the difference between the original and the reconstructed blocks of speech.
  • the difference vector is perceptually weighted by weighting filter 132, utilizing the weighting filter parameters WTP generated by coefficient analyzer 110. Refer to the preceding reference for a representative weighting filter transfer function. Perceptual weighting accentuates those frequencies where the error is perceptually more important to the human ear, and attenuates other frequencies.
  • Energy calculator 134 computes the energy of the weighted difference vector e' i (n), and applies this error signal E i to codebook search controller 140.
  • the search controller compares the i-th error signal for the present excitation vector u i (n) against previous error signals to determine the excitation vector producing the minimum error.
  • the code of the i-th excitation vector having a minimum error is then output over the channel as the best excitation code I.
  • search controller 140 may determine a particular codeword which provides an error signal having some predetermined criteria, such as meeting a predefined error threshold.
  • Figure 1 illustrates one embodiment of the invention for a code-excited linear predictive speech coder.
  • the long-term filter parameters L and ⁇ are determined in an open-loop configuration by coefficient analyzer 110.
  • the long-term filter parameters can be determined in a closed-loop configuration as described in the aforementioned Singhal and Atal reference.
  • performance of the speech coder is improved using long-term filter parameters determined in the closed-loop configuration.
  • the novel structure of the long-term predictor according to the present invention greatly facilitates the use of the closed-loop determination of these parameters for lags L less than the frame length N.
  • Figure 2A illustrates an embodiment of long-term filter 124 of Figure 1, where L is constrained to be an integer.
  • Figure 1 shows the scaled excitation vector ⁇ u i (n) from gain block 122 as being input to long-term filter 124
  • a representative input speech vector s(n) has been used in Figure 2A for purposes of explanation. hence, a frame of N samples of input speech vector s(n) is applied to adder 210.
  • the output of adder 210 produces the output vector b(n) for the long-term filter 124.
  • the output vector b(n) is fed back to delay block 230 of the long-term predictor.
  • the nominal long-term predictor lag parameter L is also input to delay block 230.
  • the long-term predictor delay block provides output vector q(n) to long-term predictor multiplier block 220, which scales the long-term predictor response by the long-term predictor coefficient ⁇ .
  • the scaled output ⁇ q(n) is then applied to adder 210 to complete the feedback loop if the recursive filter.
  • the output response H n (z) of long-term filter 124 is defined in Z-transform notation as: wherein n represents a sample number of a frame containing N samples, O ⁇ n ⁇ N-1 , wherein ⁇ represents a filter coefficient, wherein L represents the nominal lag or delay of the long-term predictor, and wherein represents the closest integer less than or equal to (n+L)/L .
  • the long-term predictor delay L varies as a function of the sample number n.
  • the actual long-term predictor delay becomes kL, wherein L is the basic or nominal long-term predictor lag, and wherein k is an integer chosen from the set ⁇ 1, 2, 3, 4, hereby ⁇ as a function of the sample number n.
  • the long-term filter output response b(n) is a function of the nominal long-term predictor lag parameter L and the filter state FS which exists at the beginning of the frame. This statement holds true for all values of L -- even for the problematic case of when the pitch lag L is less than the frame length N.
  • the function of the long-term predictor delay block 230 is to store the current input samples in order to predict future samples.
  • Figure 2B represents a simplified diagram of a shift register, which may be helpful in understanding the operation of long-term predictor delay block 230 of Figure 2A.
  • the current output sample b(n) is applied to the input of the shift register, which is shown on the right on Figure 2B.
  • the previous sample b(n) is shifted left into the shift register. This sample now becomes the first past sample b(n-1).
  • the long-term predictor lag parameter L is shorter than the frame length N, then an insufficient number of samples would have been shifted into the shift register by the beginning of the next frame.
  • the pitch lag L would be equal to 32.
  • b(n-L) would normally be b(27), which represents a future sample with respect to the beginning of the frame of 60 samples.
  • the complete long-term predictor response is needed at the beginning of the frame such that closed-loop analysis of the predictor parameters can be performed.
  • the same stored samples b(n-L), 0 ⁇ n ⁇ L are repeated such that the output response of the long-term predictor is always a function of samples which have been input into the long-term predictor delay block prior to the start of the current frame.
  • the shift register has thus been extended to store another kL samples, which represent modifying the structure of the long-term predictor delay block 230.
  • k must be chosen such that b(n-kL) represents a sample which existed in the shift register prior to he start of the frame.
  • long-term filter 124 of Figure 2A The operation of long-term filter 124 of Figure 2A will now be described in accordance with the flowchart of Figure 3.
  • the sample number n is initialized to zero at step 351.
  • the nominal long-term predictor lag parameter L and the long-term predictor coefficient ⁇ are input from coefficient analyzer 110 in step 352.
  • the sample number n is tested to see if an entire frame has been output. If n ⁇ N, operation ends at step 361. If all samples have not yet been computed, a signal sample s(n) is input in step 354.
  • the output response of long-term predictor delay block 230 is calculated according to the equation: wherein represents the closest integer less than or equal to (n+L)/L .
  • step 357 the sample in the shift register is shifted left one position, for all register locations between b(n-2) and b(n-L MAX ), where L MAX represents the maximum long-term predictor lag that can be assigned. In the preferred embodiment, L MAX would be equal to 143.
  • step 358 the output sample b(n) is input into the first location b(n-1) of the shift register.
  • Step 359 outputs the filtered sample b(n). The sample number n is then incremented in step 360, and then tested in step 353. When all N samples have been computed, the process ends at step 361.
  • Figure 2C is an alternative embodiment of a long-term filter incorporating the present invention.
  • Filter 124' is the feedforward inverse version of the recursive filter configuration of Figure 2A.
  • Input vector s(n) is applied to both subtracter 240 and long-term predictor delay block 260.
  • Delayed vector q(n) is output to multiplier 250, which scales the vector by the long-term predictor coefficient ⁇ .
  • the output response H n (z) of digital filter 124' is given in z-transform notation as: wherein n represents the sample number of a frame containing N samples, 0 ⁇ n ⁇ N-1 , wherein ⁇ represents the long-term filter coefficient, wherein L represents the nominal lag or delay of the long-term predictor, and wherein represents the closest integer less than or equal to (n+L)/L .
  • the output signal b(n) of filter 124' may also be defined in terms of the input signal s(n) as: for 0 ⁇ n ⁇ N-1 .
  • the structure of the long-term predictor has again been modified so as to repeatedly output the same stored samples of the long-term predictor in the case of when the long-term predictor lag L is less than the frame length N.
  • FIG. 5 there is illustrated the preferred embodiment of the long-term filter 124 of Figure 1 which allows for subsample resolution for the lag parameter L.
  • a frame of N samples of input speech vector s(n) is applied to adder 510, The output of adder 510 produces the output vector b(n) for the long term filter 124.
  • the output vector b(n) is fed back to delayed vector generator block 530 of the long-term predictor.
  • the nominal long-term predictor lag parameter L is also input to delayed vector generator block 530.
  • the long-term predictor lag parameter L can take on non-integer values.
  • the preferred embodiment allows L to take on values which are a multiple of one half. Alternate implementations of the sub-sample resolution long-term predictor of the present invention could allow values which are multiples of one third or one fourth or any other rational fraction.
  • the delayed vector generator 530 includes a memory which holds past samples of b(n).
  • interpolated samples of b(n) are also calculated by delayed vector generator 530 and stored in its memory.
  • the state of the long-term predictor which is contained in delayed vector generator 530 has two samples for every stored sample of b(n). One sample is for b(n) and the other sample represents an interpolated sample between two consecutive b(n) samples. In this way, samples of b(n) can be obtained from delayed vector generator 530 which correspond to integer delays or multiples of half sample delays.
  • the interpolation is done using interpolating finite impulse response filters as described in the book by R. Crochiere and L. Rabiner entitled Multirate Digital Signal Processing , published by Prentice Hall in 1983.
  • the operation of vector delay generator 530 is described in further detail hereinbelow in conjunction with the flowcharts in Figure 6A and 6B.
  • Delayed vector generator 530 provides output vector q(n) to long-term multiplier block 520, which scales the long-term predictor response by the long-term predictor coefficient ⁇ . The scaled output ⁇ q(n) is then applied to adder 510 to complete the feedback loop of the recursive filter 124 in Figure 5.
  • the resolution of the long-term predictor memory is extended by mapping an N point sequence b(n), onto a 2N point vector ex(i).
  • the negative indexed samples of ex(i) contain the extended resolution past values of long-term filter output b(n), or the extended resolution long term history.
  • the mapping process doubles the temporal resolution of the long-term predictor memory, each time it is applied.
  • single stage mapping is described, although additional stages may be implemented in other embodiments of the present invention.
  • step 604 the flowchart proceeds to step 604, where L, ⁇ and s(n) are inputted.
  • vector q(n) is constructed according to the equation: for 0 ⁇ n ⁇ N-1 wherein represents the closest integer less than or equal to (n+L)/L and wherein L is the long term predictor lag.
  • long term predictor lag L may be the pitch period or a multiple of the pitch period.
  • L may be an integer or a real number whose fractional part is 0.5 in the preferred embodiment. When the fractional part of L is 0.5, L has an effective resolution of half a sample.
  • step 612 vector b(n) of the long-term filter is outputted.
  • step 614 the extended resolution state ex(n) is updated to generate and store the interpolated values of b(n) in the memory of delayed vector generator 530. Step 614 is illustrated in more detail in Figure 6B.
  • step 616 the process has been completed and stops.
  • the interpolated samples of ex(i) initialized to zero are reconstructed through FIR interpolation, using a symmetric, zero-phase shift filter, assuming that the order of such FIR filter is 2M+1 as explained hereinabove.
  • Only even samples pointed to be the FIR filter taps are used in sample reconstruction, since odd samples have been set to zero.
  • M+1 samples instead of 2M+1 samples are actually weighted and summed for each reconstructed sample.
  • the parameter ⁇ the history extension scaling factor, may be set equal to ⁇ , which is the pitch predictor coefficient, or set to unity.
  • ex(i) ex(i+2N)
  • Max_L the maximum long term predictor delay used.
  • Synthesizer 400 obtains the short-term predictor parameters ⁇ i , long-term predictor parameters ⁇ and L, excitation gain factor ⁇ and the codeword I received from the channel, via de-multiplexer 450.
  • the codeword I is applied to codebook ROM 420 to address the codebook of excitation vectors.
  • the single excitation vector u I (n) is then multiplied by the gain factor ⁇ in block 422, filtered by long-term predictor filter 424 and short-term predictor filter 426 to obtain reconstructed speech vector s' I (n).
  • This vector which represents a frame of reconstructed speech, is then applied to analog-to-digital (A/D) convertor 408 to produce a reconstructed analog signal, which is then low pass filtered to reduce aliasing by filter 404, and applied to an output transducer such as speaker 402.
  • A/D analog-to-digital
  • the CELP synthesizer utilizes the same codebook, gain block, long-term filter, and short-term filter as the CELP analyzer of Figure 1.
  • FIG. 7 is a detailed block diagram of a pitch post filter for intercoupling the short term filter 426 and D/A converter 408 of the speech synthesizer in Figure 4.
  • a pitch post filter enhances the speech quality by removing noise introduced by the filters 424 and 426.
  • a frame of N samples of reconstructed speech vector s' I (n) is applied to adder 710.
  • the output of adder 710 produces the output vector s'' I (n) for the pitch post filter.
  • the output vector s'' I (n) is fed back to delayed sample generator block 730 of the pitch post filter.
  • the nominal long-term predictor lag parameter L is also input to delayed sample generator block 730. L may take on non-integer values for the present invention.
  • Delayed sample generator 730 provides output vector q(n) to multiplier block 720, which scales the pitch post filter response by coefficient R which is a function of the long-term predictor coefficient ⁇ . The scaled output Rq(n) is then applied to adder 710 to complete the feedback loop of the pitch post filter in Figure 7.
  • the excitation gain factor ⁇ and the long-term predictor coefficient ⁇ can be simultaneously optimized for all values of L in a closed-loop configuration.
  • This joint optimization technique was heretofore impractical for values of L ⁇ N, since the joint optimization equations would become non-linear in the single parameter ⁇ .
  • the present invention modifies the structure of the long-term predictor to allow a linear joint optimization equation.
  • the present invention allows the long-term predictor lag to have better resolution than one sample thereby enhancing its performance.
  • the codebook search procedure has been further simplified, since the zero state response of the long-term filter becomes zero for lags less than the frame length.
  • This additional feature permits those skilled in the art to remove the effect of the long-term filter from the codebook search procedure.
  • a CELP speech coder has been shown which can provide higher quality speech for all pitch rates while retaining the advantages of practical implementation and low bit rate.
  • any type of speech coding e.g., RELP, multipulse, RPE, LPC, etc.
  • RELP RELP
  • multipulse RPE
  • LPC LPC
  • additional equivalent configurations of the sub-sample resolution long-term predictor structure may be made which perform the same computations as those illustrated above.

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Claims (16)

  1. Procédé de reconstitution de signaux vocaux comprenant les étapes consistant à :
    recevoir d'une voie de communication un ensemble de paramètres de signaux vocaux comportant un mot de code I et un paramètre de retard L, le paramètre de retard L pouvant prendre une valeur comprise dans un éventail prédéterminé comportant des valeurs entières et non entières, liées à la période de hauteur de son des signaux vocaux ;
    générer un vecteur d'excitation ayant une pluralité d'échantillons en réponse à un mot de code I ;
    filtrer le vecteur d'excitation en fonction d'au moins le paramètre de retard L et des échantillons de l'état du filtre stocké, l'étape de filtrage comprenant les étapes consistant à :
    calculer les échantillons de l'état du filtre interpolé à partir des échantillons de l'état filtré stocké en utilisant un L non entier, et
    combiner le vecteur d'excitation aux échantillons de l'état du filtre interpolé, ce qui permet de former un vecteur de sortie du filtre ayant une pluralité d'échantillons de sortie du filtre ; et
    traiter le vecteur de sortie du filtre afin de produire un signal vocal reconstitué.
  2. Procédé permettant de reconstituer des signaux vocaux selon la revendication 1, caractérisé en outre en ce que l'étape de filtrage comprend la combinaison, en réponse au fait que L est un entier, du vecteur d'excitation avec l'échantillon de l'état du filtre stocké, ce qui permet de former des échantillons de sortie de l'état du filtre.
  3. Procédé permettant de reconstituer des signaux vocaux selon la revendication 1, caractérisé en outre en ce que l'étape de filtrage comprend la mise à jour des échantillons de l'état du filtre stocké en utilisant les échantillons de sortie du filtre.
  4. Procédé permettant de reconstituer des signaux vocaux selon la revendication 1, comprenant en outre les étapes consistant à :
    convertir les signaux vocaux reconstitués en un signal vocal analogique ; et
    convertir au moyen d'un transducteur le signal vocal analogique en une sortie audio audible, de telle sorte que les périodes de hauteur de son des signaux vocaux soient prévues de façon plus précise.
  5. Appareil permettant de reconstituer les signaux vocaux comprenant :
    les circuits de réception (450) permettant de recevoir d'une voie de communication un ensemble de paramètres de signaux vocaux comportant un mot de code I et un paramètre de retard L ;
    des circuits de génération (420) permettant de générer un vecteur d'excitation ayant une pluralité d'échantillons en réponse au mot de code I ;
    des circuits de filtrage (124, 424) permettant de filtrer le vecteur d'excitation en fonction d'au moins le paramètre de retard L et des échantillons de l'état du filtre stocké, caractérisé en ce que L présente une valeur comprise dans un éventail prédéterminé comprenant des valeurs entières et non entières liées à une période de hauteur de son des signaux vocaux, et en ce que les circuits de filtrage comprennent :
    des circuits de calcul (570) permettant de calculer les échantillons de l'état du filtre interpolé à partir des échantillons de l'état filtré stocké en utilisant un L non entiers ; et
    des circuits de combinaison (510, 520) permettant de combiner le vecteur d'excitation aux échantillons de l'état du filtre interpolé, ce qui permet de former un vecteur de sortie du filtre ayant une pluralité d'échantillons de sortie de filtre ; et
    des circuits de traitement (426) permettant de traiter les vecteurs de sortie du filtre afin de produire un signal vocal reconstitué.
  6. Appareil permettant de reconstituer les signaux vocaux selon la revendication 5, dans lequel les circuits de combinaison comprennent en outre la combinaison, en réponse au fait que L est un entier, du vecteur d'excitation avec les échantillons de l'état du filtre stocké, ce qui permet de former les échantillons de sortie de l'état du filtre.
  7. Appareil permettant de reconstituer les signaux vocaux selon la revendication 5, dans lequel les circuits de filtrage comprennent en outre des circuits de mise à jour permettant de mettre à jour les échantillons de l'état du filtre stocké en utilisant les échantillons de sortie du filtre.
  8. Appareil permettant de reconstituer les signaux vocaux selon la revendication 5, comprenant en outre :
    les circuits de conversion permettant de convertir les signaux vocaux reconstitués en un signal vocal analogique ; et
    des circuits de conversion par transducteur, permettant de convertir le signal vocal analogique en une sortie audio audible, de telle sorte que les périodes de hauteur de son des signaux vocaux puissent être prévus de façon plus précise.
  9. Procédé permettant de reconstituer des signaux vocaux comprenant les étapes consistant à :
    recevoir d'une voie de communication un ensemble de paramètres de signaux vocaux comportant un mot de code I et un paramètre de retard L ;
    générer un vecteur d'excitation ayant une pluralité d'échantillons en réponse au mot de code I ;
    filtrer le vecteur d'excitation en fonction d'au moins le paramètre de retard L, un ensemble d'échantillons de l'état du filtre stocké, caractérisé en ce que L présente une valeur comprise dans un éventail prédéterminé comportant des valeurs entières et non entières liées à une période de hauteur de son des signaux vocaux et au moins un ensemble d'échantillons de l'état du filtre interpolé stocké, l'étape de filtrage comprenant les étapes consistant à :
    choisir un ensemble d'échantillons de l'état du filtre dans un groupe composé de l'ensemble d'échantillons de l'état du filtre stocké et d'au moins un ensemble d'échantillons de l'état du filtre interpolé et stocké, l'étape consistant à faire un choix utilisant au moins le paramètre de retard L, et
    combiner le vecteur d'excitation aux échantillons de l'état du filtre choisi, ce qui permet de former un vecteur de sortie du filtre ayant une pluralité d'échantillons de sortie du filtre ; et
    traiter le vecteur de sortie du filtre pour produire les signaux vocaux reconstitués.
  10. Procédé permettant de reconstituer des signaux vocaux selon la revendication 9, comprenant en outre les étapes consistant à :
    convertir le signal vocal reconstitué en un signal vocal analogigue ; et
    convertir au moyen d'un transducteur le signal vocal analogique en une sortie audio audible, de telle sorte que les périodes de hauteur de son des signaux vocaux puissent être prévues de façon plus précise.
  11. Appareil permettant de reconstituer les signaux vocaux comprenant :
    des circuits de réception (450) permettant de recevoir depuis une voie de communication un ensemble de paramètres de signaux vocaux comportant un mot de code I et un paramètre de retard L, L pouvant avoir une valeur comprise dans un éventail prédéterminé comportant des valeurs entières et non entières liées à une période de hauteur de son de signaux vocaux ;
    des circuits de génération (420) permettant de générer un vecteur d'excitation ayant une pluralité d'échantillons en réponse au mot de code I ;
    des circuits de filtrage (124, 424) permettant de filtrer le vecteur d'excitation en fonction d'au moins le paramètre de retard L, un ensemble d'échantillons de l'état du filtre stocké et au moins un ensemble d'échantillons de l'état du filtre interpolé et stocké, les circuits de filtrage comprenant :
    les circuits de choix permettant de choisir un ensemble d'échantillons de l'état du filtre à partir du groupe composé de l'ensemble des échantillons de l'état du filtre stocké et d'au moins un ensemble d'échantillons de l'état du filtre interpolé et stocké, l'étape de choix utilisant au moins le paramètre de retard L, et
    des circuits de combinaison pour combiner le vecteur d'excitation aux échantillons de l'état du filtre choisi, ce qui permet de former un vecteur de sortie du filtre ayant une pluralité d'échantillons de sortie du filtre ; et
    des circuits de traitement (426) permettant de traiter le vecteur de sortie du filtre pour produire un signal vocal reconstitué.
  12. Appareil permettant de reconstituer des signaux vocaux selon la revendication 11, comprenant en outre :
    des circuits de conversion permettant de convertir les signaux vocaux reconstitués en un signal vocal analogique, et
    des circuits de conversion par transducteur permettant de convertir le signal vocal analogique en une sortie audio audible, de telle sorte que les périodes de hauteur de son des signaux vocaux soient prévues de façon plus précise.
  13. Procédé de codage de signaux vocaux en ensembles de paramètres de signaux vocaux en vue de leur transmission sur une voie de communication, le procédé comprenant les étapes consistant à :
    échantillonner un signal vocal une pluralité de fois pour fournir une pluralité d'échantillons formant un vecteur de signaux vocaux présent ;
    générer un paramètre de retard L ayant une valeur comprise dans un éventail prédéterminé comportant des valeurs entières et non entières liées à la période de hauteur de son des signaux vocaux du vecteur de signaux vocaux présent ;
    rechercher des vecteurs d'excitation pour déterminer un mot de code I qui correspond le mieux au vecteur de signaux vocaux présent, l'étape de recherche comprenant les étapes consistant à :
    générer des vecteurs d'excitation en réponse au mot de code correspondant ;
    filtrer chaque vecteur d'excitation comprenant les étapes consistant à :
    calculer les échantillons de l'état du filtre interpolé à partir des échantillons de l'état filtré et stocké en utilisant un L non entier ; et
    combiner le vecteur d'excitation aux échantillons de l'état du filtre interpolé, ce qui permet de former un vecteur de sortie du filtre ayant une pluralité d'échantillons de sortie du filtre ;
    sélection du mot de code I du vecteur d'excitation pour lequel les signaux vocaux synthétisés en utilisant le L non entier diffèrent le moins des signaux vocaux ; et
    transmettre le paramètre choisi L avec les paramètres de signaux vocaux présélectionné pour le présent vecteur de signaux vocaux sur la voie de communication de telle sorte que les périodes de hauteur de son des signaux vocaux soient prévues de façon plus précise.
  14. Procédé de codage de signaux vocaux selon la revendication 13, caractérisé en outre en ce que l'étape de recherche des vecteurs d'excitation comprend :
    le traitement du vecteur de sortie du filtre pour produire un vecteur de signaux vocaux reconstitués ; et
    la comparaison du vecteur de signaux vocaux reconstitués au vecteur de signaux vocaux présents pour déterminer la différence entre les deux.
  15. Procédé de codage de signaux vocaux selon la revendication 13, caractérisé en outre en ce que l'étape de sélection du mot de code I comprend la sélection du mot de code I du vecteur d'excitation pour lequel le vecteur de signaux vocaux reconstitués diffère le moins du présent vecteur de signaux vocaux.
  16. Appareil de codage de signaux vocaux en ensemble de paramètres de signaux vocaux en vue de leur transmission sur une voie de communication, l'appareil comprenant :
    des circuits d'échantillonnage pour échantillonner un signal vocal une pluralité de fois pour fournir une pluralité d'échantillons formant un vecteur de signaux vocaux présents ;
    des circuits de génération pour générer un paramètre de retard L ayant une valeur comprise dans un éventail prédéterminé comportant des valeurs entières et non entières liées à la période de hauteur de son des signaux vocaux du présent vecteur de signaux vocaux ;
    des circuits de recherche pour rechercher des vecteurs d'excitation afin de déterminer un mot de code I qui corresponde le mieux au présent vecteur de signaux vocaux, les circuits de recherche comprenant :
    des circuits de génération pour générer des vecteurs d'excitation en réponse au mot de code correspondant ;
    des circuits de filtrage pour filtrer chaque vecteur d'excitation en fonction d'au moins le paramètre de retard L, un ensemble d'échantillons d'état du filtre stocké et au moins un ensemble d'échantillon d'état du filtre interpolé stocké, les circuits de filtrage comprenant :
    des circuits de choix pour choisir un ensemble d'échantillons d'état de filtre dans le groupe composé de l'ensemble d'échantillons d'état du filtre stocké et de l'au moins un ensemble d'échantillons d'état du filtre interpolé stocké, les circuits de choix utilisant au moins le paramètre de retard L,
    des circuits de combinaison pour combiner le vecteur d'excitation avec des échantillons d'état du filtre choisi afin de former un vecteur de sortie du filtre ayant une pluralité d'échantillons de sortie du filtre ;
    des circuits de traitement pour traiter le vecteur de sortie du filtre et pour produire un vecteur de signaux vocaux reconstitués ;
    des circuits de comparaison pour comparer le vecteur de signaux vocaux reconstitués au présent vecteur de signaux vocaux afin de déterminer la différence entre les deux ;
    des circuits de sélection pour sélectionner le mot de code I du vecteur d'excitation pour lequel le vecteur des signaux d'excitation reconstitués diffèrent le moins du présent vecteur de signaux vocaux ; et
    des circuits de transmission pour transmettre le mot de code sélectionné i et le paramètre de retard L avec les paramètres de signaux vocaux présélectionnés pour le présent vecteur de signaux vocaux sur la voie de communication de telle sorte que les périodes de hauteur de son de signaux vocaux soient prévues de façon précise.
EP91905041A 1989-09-01 1990-06-25 Codeur de parole numerique a predicteur a long terme ameliore a resolution au niveau sous-echantillon Expired - Lifetime EP0450064B2 (fr)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
US40220689A 1989-09-01 1989-09-01
US402206 1989-09-01
PCT/US1990/003625 WO1991003790A1 (fr) 1989-09-01 1990-06-25 Codeur de parole numerique a prediseur a long terme ameliore

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EP0450064A1 EP0450064A1 (fr) 1991-10-09
EP0450064A4 EP0450064A4 (en) 1995-04-05
EP0450064B1 true EP0450064B1 (fr) 2000-04-19
EP0450064B2 EP0450064B2 (fr) 2006-08-09

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JP (1) JP3268360B2 (fr)
CN (1) CN1026274C (fr)
AT (1) ATE191987T1 (fr)
AU (1) AU634795B2 (fr)
CA (1) CA2037899C (fr)
DE (1) DE69033510T3 (fr)
DK (1) DK0450064T4 (fr)
ES (1) ES2145737T5 (fr)
MX (1) MX167644B (fr)
SG (1) SG47028A1 (fr)
WO (1) WO1991003790A1 (fr)

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FI96248C (fi) 1993-05-06 1996-05-27 Nokia Mobile Phones Ltd Menetelmä pitkän aikavälin synteesisuodattimen toteuttamiseksi sekä synteesisuodatin puhekoodereihin
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GB9408037D0 (en) * 1994-04-22 1994-06-15 Philips Electronics Uk Ltd Analogue signal coder
JP2970407B2 (ja) * 1994-06-21 1999-11-02 日本電気株式会社 音声の励振信号符号化装置
FR2729244B1 (fr) * 1995-01-06 1997-03-28 Matra Communication Procede de codage de parole a analyse par synthese
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CN1026274C (zh) 1994-10-19
MX167644B (es) 1993-03-31
ES2145737T5 (es) 2007-03-01
DE69033510D1 (de) 2000-05-25
DE69033510T2 (de) 2000-11-23
EP0450064A1 (fr) 1991-10-09
AU634795B2 (en) 1993-03-04
JPH04502675A (ja) 1992-05-14
DE69033510T3 (de) 2007-06-06
ATE191987T1 (de) 2000-05-15
EP0450064B2 (fr) 2006-08-09
DK0450064T4 (da) 2006-09-04
DK0450064T3 (da) 2000-10-02
JP3268360B2 (ja) 2002-03-25
CA2037899C (fr) 1996-09-17
SG47028A1 (en) 1998-03-20
EP0450064A4 (en) 1995-04-05
WO1991003790A1 (fr) 1991-03-21
CN1050633A (zh) 1991-04-10
ES2145737T3 (es) 2000-07-16
CA2037899A1 (fr) 1991-03-02
AU5952590A (en) 1991-04-08

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