EP0347307B1 - Kodierungsverfahren und linearer Prädiktionssprachkodierer - Google Patents

Kodierungsverfahren und linearer Prädiktionssprachkodierer Download PDF

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Publication number
EP0347307B1
EP0347307B1 EP89401644A EP89401644A EP0347307B1 EP 0347307 B1 EP0347307 B1 EP 0347307B1 EP 89401644 A EP89401644 A EP 89401644A EP 89401644 A EP89401644 A EP 89401644A EP 0347307 B1 EP0347307 B1 EP 0347307B1
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European Patent Office
Prior art keywords
filtering
vector
excitation
vectors
subjected
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EP89401644A
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English (en)
French (fr)
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EP0347307A3 (en
EP0347307A2 (de
Inventor
Michel Lever
Marc Delprat
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Nortel Networks France SAS
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Matra Communication SA
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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
    • G10L19/005—Correction of errors induced by the transmission channel, if related to the coding algorithm
    • 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
    • G10L19/113—Regular pulse excitation
    • 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
    • G10L2019/0003—Backward prediction of gain
    • 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
    • G10L2019/0013—Codebook search algorithms
    • G10L2019/0014—Selection criteria for distances
    • 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
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/03—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters
    • G10L25/06—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters the extracted parameters being correlation coefficients

Definitions

  • the subject of the present invention is a coding method and a speech coder of the type known as linear prediction analysis. It relates more particularly to methods and speech coders of this type with excitation by excitation vector, often designated by the English abbreviation CELP, which are to be distinguished from coding methods with analysis by linear prediction with multi-pulse excitation. (MPLPC), an example of which is given in document EP-A-0 195 487 to which reference may be made.
  • Vector-driven linear prediction analysis coding provides an interesting solution to the problem of speech transmission in a narrow band channel, for example, transmission between mobiles and to mobiles in a 12.5 kHz channel which reduces the bit rate available at around 8 kbits / s; in the latter case, the bit rate assigned to the transmission of the parameters representing the speech signal is reduced to about 6 kbits / s since part of the overall bit rate must be assigned to the transmission of an error correction code.
  • Speech coders with linear prediction and vector excitation are already known, usable with a low bit rate, usually between a quarter of a bit and a half bit per speech sample.
  • SCHROEDER and ATAL Code excited linear prediction (CELP): high quality speech at very low bit rates
  • FIG. 1 gives a schematic diagram of such an encoder 10.
  • the speech signal is applied to this encoder via a digitization chain.
  • the chain comprises, from a microphone 12, a low-pass filter 14 limiting the bandwidth to approximately 4000 Hz and a sampler-encoder 16.
  • the sampler takes samples speech at a rate which is for example 8 kHz and provides successive samples, grouped by vocoder frames occupying time windows of fixed duration, for example 20 ms.
  • the coder 10 transforms the speech signal into a coded signal having a lower bit rate, transmitted to the transmission equipment by a multiplexer 18 which receives, for each frame, the indices k of the optimal excitation vectors Ck , the associated gains G k and coefficients identifying prediction parameters, for each of the constituent blocks of the frame, each occupying a sub-window.
  • the coder 10 shown by way of example in FIG. 1 uses analysis by synthesis: the speech spectrum in each window is modeled by a linear predictor filter whose coefficients are variable over time.
  • the residual signal, obtained by subtraction, is subject to vector quantization using a dictionary of waveforms.
  • excitation vectors stored in the dictionary 20 are chosen either empirically taking account of statistical data on the language, or randomly, or else from conventional binary digital codes such as the Golay codes.
  • the article by SCHROEDER and others mentioned above proposes for example a dictionary comprising 1024 excitation vectors each made up of 40 samples. This number of vectors is placed between the minimum below which the excitation would be poorly represented and the maximum beyond which the number of bits left free would be insufficient to transmit the parameters of the predictors.
  • the output of amplifier 22 is applied to a predictive synthesis filter consisting of a long-term predictor filter 24, intended to introduce the periodicity of the long-term signal, and of a short-term predictor filter 26.
  • the output Sn of the predictor filter which represents a synthesis of estimation of the speech signal, is applied to the subtractive input of a subtractor 28 which receives, on its additive input, the sampled and digitized speech signal Sn.
  • the coding operation consists in determining the optimal sequence of innovation c k and the gain G k for each speech frame by a synthetic analysis process.
  • the synthesis signal obtained S k is compared to the original signal S and the difference signal obtained in the subtractor 28 is processed in a perceptual weighting filter 30 having a transfer function W (z ), whose function is to attenuate the frequencies for which the errors are less important from the perceptual point of view and on the contrary to amplify the frequencies for which the errors are more important from the perceptual point of view.
  • a circuit 32 searches for the coding sequence for which the energy contained in the weighted error signal e k for a sub-window is minimal; this sequence is selected for the current block, then the optimum gain G k is calculated.
  • the function A (z) of the short-term predictive filter 26 is of the form:
  • the coefficients a (i) constitute the parameters of linear prediction. Their number is generally between 8 and 16 for windows of 20 ms.
  • the transfer function B (z) can be of the form 1-bz- T and involve a delay T ranging from 40 to 120 samples.
  • the perceptual weighting filter 30 has a transfer function W (z) which is generally of the form:
  • CELP coding method in accordance with the preamble of claim 1 (IEEE Journal on selected areas in communications, Vol. 6, n ° 2, February 1988, pages 353-363); the present invention aims to provide a coding method with linear prediction and excitation by coding vectors of this type, which meets the requirements of practice better than those previously known, in particular in that it reduces by at least an order of magnitude the volume of calculation to be carried out for the coding of a segment.
  • the invention notably proposes a speech coding method, with linear prediction and vector excitation, according to the characterizing part of claim 1.
  • each coding sequence consists of several equidistant pulses separated by zeros, advantageously binary, that is to say that an excitation by regular pulse sequences, or RPCELP is used, we reduce in very large proportions the duration of the search for the optimal sequence, especially if an appropriate choice is made of the characteristics of the perceptual weighting filter.
  • the perceptual weighting filter 30, placed at the output of the subtractor 28 in FIG. 1 is transferred to the two input branches of the subtractor in the form of filters 34 and 36, of transfer function 1 / A (z / y). There is thus in cascade, on the branch assigned to the original signal S (n), the filter 33 of transfer function A (z) and the filter 36 having the same transfer function as the filter 34.
  • the filtering of all the vectors by the synthesis filter, of transfer function 1 / A (z / y) whose coefficients vary over time, represents an enormous volume of calculations. This volume is reduced very considerably according to a first aspect of the invention, by adopting a perceptual weighting filter with small number of fixed coefficients in time, chosen according to the average characteristics of the speech over a long time interval.
  • the perceptual weighting filter then has a transfer function W (z) which can be written: where C (z / -y) is the transfer function of a short-term speech predictor, for example of the form:
  • the contribution of the memory in the long-term predictor filter 24, of transfer function 1 / B (z), and in the short-term predictor weighted function filter transfer 1 / A (z / y), is subtracted from the original signal having undergone the weighting to obtain a signal x n , before the start of the search in the vector dictionary 20.
  • This operation is carried out in the figure 3 using a subtractor 38 which receives only the memory component of the long-term predictor filter 24.
  • each vector C k is only processed by the weighted synthesis filter 34.
  • each of the filters 34 and 36 has been shown broken down into a filter 34a or 36a of transfer function 1 / ⁇ (z / ⁇ ), without memory, and a filter 34b or 36b corresponding only to the contribution of the memory terms.
  • the filtering operation by filter 34a is expressed above by the convolution of two finite sequences, represented by the product of a matrix and a vector: where H is a lower triangular matrix LxL (L being the common length of the sequences) whose elements are taken from the impulse response h (i) of 1 / A (z / y), of the form: which merges with that of 1 / ⁇ (z / y)
  • the next step in the process consists in eliminating the memory terms, that is to say the operations shown diagrammatically in 34a and 36a, to arrive at the constitution shown in FIG. 5.
  • W '(z) A (z) / C (z / -y)
  • Yet another embodiment of the invention implements a modified error evaluation criterion to be minimized.
  • the sample frames each occupying a window are successively applied; consequently, the impulse response of the weighted synthesis filter for a frame (or a block) occurs on the next frame (or the next block).
  • we use the damping of the filters and we apply to their input instead of a sequence consisting only of L samples, a sequence consisting of L samples and J zeros, J being chosen so that the impulse response of the synthesis filter W (z) / A (z) is practically zero after J samples.
  • the impulse response matrix then becomes a rectangular "strip" type matrix with (L + J) xL terms of the type:
  • A will then be calculated for each frame while k and G k will be calculated for each block.
  • a particularly interesting solution in this case consists in using sequences of pulses of length L having a regular structure made up of q equidistant pulses separated by D-1 zeros, the first pulse occupying one of the positions 0 to D-1 and the number of sequences being such that all of these positions are successively occupied. It is thus possible to give a satisfactory representation of the phase information in the excitation signal.
  • the dictionary consists of a basic set of K / D sequences, with a zero phase and with three successive shifts, ie in all K sequences.
  • Excitation by regular excitation sequences reduces the number of operations to be performed, since many of the products to be performed are zero, one of the factors being a zero whose position is known for each sample.
  • EP-A-0 195 487 relates to an MPLPC coding method according to which it is necessary successively to determine an optimal phase of pulses, then to seek the optimal amplitude of all the pulses constituting sequence among discrete values , quantified for example on 3 bits.
  • H c k all become equal and we have: where d m denotes one of the sequences (the number of K / D) resulting from the decimation of the components of the K vectors by elimination of zeros; the sequence d m for 0 ⁇ k 3 3 is given in FIG. 7 by way of example.
  • the coder then presents the constitution of principle shown in FIG. 6.
  • a single filtering operation is performed on the speech signal frame by the filter 33.
  • the sequence c k tested in a form which no longer needs to be prefiltered , is applied to the circuit 32 for calculating the scalar product c k t .y and for determining the maximum, for which an index selection order is sent at 40.
  • the sequence c k amplified at 22 is applied to the long-term predictor 24, shown with a single coefficient b.
  • the term r is formed by subtracting the output of the long-term predictor 24 from the output of the filter 34 on the speech channel, in the subtractor 38.
  • the filter 42 which receives the residue has a fixed response R (z) represented by a symmetric Toeplitz matrix.
  • This process reduces the number of calculations required in a report which is typically about three orders of magnitude compared to the conventional CELP method, regardless of the length L chosen for the speech blocks.
  • the gain G k is, for transmission, quantized in a quantizer 46.
  • Each signal frame is split into several blocks, an intermediate memory 48 must be interposed between the components 33 and 44.

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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)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)

Claims (6)

1. Kodierungsverfahren für Sprache mit linearer Prädiktion und vektorieller Anregung, das die Kodierung von Sprachsignalen erlaubt, die in Form von in Raster verteilten numerierten Abtastproben gebracht sind, gemäß welchem:
ein Signalraster dargestellt wird einerseits durch Prädiktionsparameter und andererseits durch eine Folge von Anregungsvektoren, welche in einem Verzeichnis (20) enthalten sind, und durch Verstärkungsfaktoren (Gk) dieser Vektoren, wobei die erhaltenen Vektoren durch Suche (32) des Energieminimums eines Fehlersignals bestimmt werden, das seinerseits durch Subtraktion jedes zuvor einer Filterung unterworfenen Vektors von dem Raster des Sprachsignals erhalten wurde, und wobei
vor der Substraktion jedes Sprachsignalraster einer Kurzterm-Analysefilterung (33) und einer merklich gewichteten Synthesefilterung (36) und die Verstärkungsvektoren einer Langterm-Prädiktionsfilterung (24) und der gleichen merklich gewichteten Synthesefilterung (34, 36) wie das Sprachsignal unterworfen werden,

dadurch gekennzeichnet, daß alle Anregungsvektoren aus der gleichen Impulsanzahl gebildet sind, die äquidistant und durch Nullen getrennt sind.
2. Verfahren nach Anspruch 1, dadurch gekennzeichnet, daß die durch Nullen getrennte Impulse binar sind.
3. Verfahren nach Anspruch 1 oder 2, dadurch gekennzeichnet, daß man jede Suche (32) nach dem Energieminimumn des Fehlersignals ausführt durch Filterung (34,36 oder 34a,36a) einer Gesamtheit, die zusätzlich zu den reellen Abtastproben eines Blocks, welcher einen Teil des Rasters bildet, Nullproben in ausreichender Zahl umfaßt, damit die Impulsantworten der Prädiktionsfilterung, welche der letzten reellen Abtastprobe entspricht, im wesentlich getilgt ist, wobei die Filterung (34,36 oder 34a, 36a) ohne Speicherung von einem Block zu einem anderen ausgeführt wird.
4. Verfahren nach Anspruch 1 oder 2, dadurch gekennzeichnet, daß jedes Sprachsignalraster nachdem es der Kurzterm-Analyse-Prädiktionsfilterung A(z) unterworfen wurde, auf den Addiereingang eines Subtrahierers (38) gegeben wird, der auf seinem Subtrahiereingang den Beitrag aus dem Speicher des Langterm-Prädiktionsfilters Ausdruck (24) empfängt, wobei
- der Ausgang des Subtrahierers einer Filterung (42) unterworfen wird,
- das skalare Produkt (32) des gefilterten Ausgangs und jeder nicht verstärkten Sequenz berechnet wird, und wobei man die Sequenz sucht, für welche das skalare Produkt maximal ist.
5. Verfahren nach Anspruch 4, dadurch gekennzeichnet, daß die Filterung (42) feste Koeffizienten hat.
6. Kodierungsverfahren für Sprache mit linearer Prädiktion und vektorieller Anregung, das die Kodierung von Sprachsignalen erlaubt, die in Form von in Raster verteilten numerierten Abtastproben gebracht sind, gemäß welchem:
jeder Block, der einen Teil des Signalrasters darstellt, wiedergibt durch einen der Vektoren, welche in einem Verzeichnis (20) enthalten sind, durch Verstärkungsfaktoren (Gk) der Vektoren, und durch Prädiktionsparameter, wobei die erhaltenen Vektoren durch Suche (32) des Energie-minimums eines Fehlersignals bestimmt werden, das seinerseits durch Subtraktion jedes zuvor einer Filterung unterworfenen Vektors von dem Raster des Sprachsignals erhalten wurde, und wobei
vor der Substraktion jedes Sprachsignalraster einer Kurzterm-Analysefilterung A(z) unterworfen wird,

dadurch gekennzeichnet, daß das Ergebnis der Subtraktion (38) einer merklich gewichteten Synthesefilterung (36) mit Festzeitkoeffizienten unterworfen wird, und daß
die Anregungsvektoren, welche in vorberechneter und gefilterter Weise gespeichert sind, einer Filterung unterworfen werden durch einen merklich gewichteten Synthesefilter 1/C(z/y) fest und ohne Speicherung, wobei alle Anregungsvektoren bestehend aus der gleichen Anzahl von Impulsen äquidistant und durch Nullen getrennt sind.
EP89401644A 1988-06-13 1989-06-13 Kodierungsverfahren und linearer Prädiktionssprachkodierer Expired - Lifetime EP0347307B1 (de)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
FR8807846 1988-06-13
FR8807846A FR2632758B1 (fr) 1988-06-13 1988-06-13 Procede de codage et codeur de parole a prediction lineaire

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EP0347307A2 EP0347307A2 (de) 1989-12-20
EP0347307A3 EP0347307A3 (en) 1990-12-27
EP0347307B1 true EP0347307B1 (de) 1994-05-04

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EP (1) EP0347307B1 (de)
DE (1) DE68915057T2 (de)
ES (1) ES2052043T3 (de)
FR (1) FR2632758B1 (de)

Families Citing this family (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
GB2266822B (en) * 1990-12-21 1995-05-10 British Telecomm Speech coding
ES2042410B1 (es) * 1992-04-15 1997-01-01 Control Sys S A Metodo de codificacion y codificador de voz para equipos y sistemas de comunicacion.
FR2720849B1 (fr) * 1994-06-03 1996-08-14 Matra Communication Procédé et dispositif de prétraitement d'un signal acoustique en amont d'un codeur de parole.
CN101615394B (zh) 2008-12-31 2011-02-16 华为技术有限公司 分配子帧的方法和装置

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Publication number Priority date Publication date Assignee Title
CA1337217C (en) * 1987-08-28 1995-10-03 Daniel Kenneth Freeman Speech coding

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EP0347307A3 (en) 1990-12-27
ES2052043T3 (es) 1994-07-01
EP0347307A2 (de) 1989-12-20
DE68915057T2 (de) 1994-08-18
FR2632758B1 (fr) 1991-06-07
DE68915057D1 (de) 1994-06-09
FR2632758A1 (fr) 1989-12-15

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