EP0347307A2 - Kodierungsverfahren und linearer Prädiktionssprachkodierer - Google Patents

Kodierungsverfahren und linearer Prädiktionssprachkodierer Download PDF

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Publication number
EP0347307A2
EP0347307A2 EP89401644A EP89401644A EP0347307A2 EP 0347307 A2 EP0347307 A2 EP 0347307A2 EP 89401644 A EP89401644 A EP 89401644A EP 89401644 A EP89401644 A EP 89401644A EP 0347307 A2 EP0347307 A2 EP 0347307A2
Authority
EP
European Patent Office
Prior art keywords
filtering
vector
frame
speech
subjected
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Granted
Application number
EP89401644A
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English (en)
French (fr)
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EP0347307B1 (de
EP0347307A3 (en
Inventor
Michel Lever
Marc Delprat
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Nortel Networks France SAS
Original Assignee
Matra Communication SA
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Publication of EP0347307A3 publication Critical patent/EP0347307A3/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
    • G10L19/005—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 present invention relates to a coding method and a speech coder of the type known as linear analysis prediction. 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 around 6 kbits / s because 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 4,000 Hz and a sampler-coder 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 c k , 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 by 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 consisting 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 Ck and the gain G k for each frame of speech by a process of analysis by synthesis.
  • the synthesis signal obtained S is compared with 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 for its part a transfer function W (z) which is generally of the form:
  • 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 proposes in particular 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.
  • each of the filters 34 and 36 has been shown broken down into a filter 34a or 36a of transfer function 1 / ⁇ (z / y), without memory, and a filter 34b or 36b corresponding only to the contribution of the memory terms.
  • the next step in the process consists in eliminating the memory terms, that is to say the operations shown diagrammatically at 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 matrix of the "strip" type 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 pulse sequences 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 is made up 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.
  • the calculations can be further simplified by constituting the sequences only of binary samples which can only take the values + 1, -1 (and 0), as indicated in FIG. 8. In fact all the sequences then contain the same energy; the search for the optimum sequence is carried out with purely scalar products and amounts to looking for the binary vector which gives the best result.
  • the document EP-AO 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, quantized values.
  • 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 the zeros; the sequence d m for 0: 5 k: 5 3 is given in FIG. 7 by way of example.
  • the coder then presents the principle constitution shown in FIG. 6.
  • a single filtering operation is carried out 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 long-term predictor 24, represented 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)
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
FR8807846A FR2632758B1 (fr) 1988-06-13 1988-06-13 Procede de codage et codeur de parole a prediction lineaire
FR8807846 1988-06-13

Publications (3)

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

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EP89401644A Expired - Lifetime EP0347307B1 (de) 1988-06-13 1989-06-13 Kodierungsverfahren und linearer Prädiktionssprachkodierer

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

Cited By (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
ES2042410A2 (es) * 1992-04-15 1993-12-01 Control Sys S A Metodo de codificacion y codificador de voz para equipos y sistemas de comunicacion.
EP0685836A1 (de) * 1994-06-03 1995-12-06 Matra Communication Verfahren und Gerät zum Vorverarbeitung von einem akoustischen Signal vor der Sprachcodierung
US6016468A (en) * 1990-12-21 2000-01-18 British Telecommunications Public Limited Company Generating the variable control parameters of a speech signal synthesis filter
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

Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO1989002147A1 (en) * 1987-08-28 1989-03-09 British Telecommunications Public Limited Company Speech coding

Patent Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO1989002147A1 (en) * 1987-08-28 1989-03-09 British Telecommunications Public Limited Company Speech coding

Non-Patent Citations (4)

* Cited by examiner, † Cited by third party
Title
ELECTRONICS LETTERS, vol. 25, no. 6, 16 mars 1989, pages 401-403, Stevenage GB; R.A. SALAMI: "Binary code excited linear prediction (BCELP): new approach to CELP coding of speech without codebooks" *
ICASSP'85 PROCEEDINGS - IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, 26-29 mars 1985, Tampa, Florida, vol. 3, pages 965-968, IEEE, New York, US; E.F. DEPRETTERE et al.: "Regular excitation reduction for effective and efficient LP-coding of speech" *
ICASSP'86 PROCEEDINGS - IEEE-IECEJ-ASJ INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSNG, 7-11 avril 1986, Tokyo, vol. 4, pages 3055-3058, IEEE, New York, US; G. DAVIDSON et al.: "Complexity reduction methods for vector excitation coding" *
PROCEEDINGS: ICASSP'87 - 1987 INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, 6-9 avril 1987, Dallas, Texas, vol. 4, pages 1957-1960, IEEE, New York, US; J.-P. ADOUL et al.: "Fast CELP coding based on algebraic codes" *

Cited By (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6016468A (en) * 1990-12-21 2000-01-18 British Telecommunications Public Limited Company Generating the variable control parameters of a speech signal synthesis filter
ES2042410A2 (es) * 1992-04-15 1993-12-01 Control Sys S A Metodo de codificacion y codificador de voz para equipos y sistemas de comunicacion.
EP0685836A1 (de) * 1994-06-03 1995-12-06 Matra Communication Verfahren und Gerät zum Vorverarbeitung von einem akoustischen Signal vor der Sprachcodierung
FR2720849A1 (fr) * 1994-06-03 1995-12-08 Matra Communication Procédé et dispositif de prétraitement d'un signal acoustique en amont d'un codeur de parole.
US5644679A (en) * 1994-06-03 1997-07-01 Matra Communication Method and device for preprocessing an acoustic signal upstream of a speech coder
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
US8843366B2 (en) 2008-12-31 2014-09-23 Huawei Technologies Co., Ltd. Framing method and apparatus

Also Published As

Publication number Publication date
EP0347307B1 (de) 1994-05-04
EP0347307A3 (en) 1990-12-27
FR2632758B1 (fr) 1991-06-07
DE68915057D1 (de) 1994-06-09
DE68915057T2 (de) 1994-08-18
ES2052043T3 (es) 1994-07-01
FR2632758A1 (fr) 1989-12-15

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