EP0481895B1 - Procédé de transmission, à bas débit, par codage CELP d'un signal de parole et système correspondant - Google Patents

Procédé de transmission, à bas débit, par codage CELP d'un signal de parole et système correspondant Download PDF

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Publication number
EP0481895B1
EP0481895B1 EP91402774A EP91402774A EP0481895B1 EP 0481895 B1 EP0481895 B1 EP 0481895B1 EP 91402774 A EP91402774 A EP 91402774A EP 91402774 A EP91402774 A EP 91402774A EP 0481895 B1 EP0481895 B1 EP 0481895B1
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Prior art keywords
vector
vectors
values
dictionary
basis
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German (de)
English (en)
French (fr)
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EP0481895A3 (en
EP0481895A2 (fr
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Renaud Di Francesco
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Orange SA
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France Telecom 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
    • G10L19/04—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
    • G10L19/083—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being an excitation 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
    • 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
    • G10L2019/0004—Design or structure of the codebook
    • G10L2019/0005—Multi-stage vector quantisation
    • 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
    • G10L2019/0007—Codebook element generation

Definitions

  • the invention relates to a method of transmission, at low speed, by CELP coding of a speech signal and to the corresponding system.
  • CELP Code Excited Linear Prediction
  • This technique for coding digital samples representing the speech signal is a hybrid coding technique in which the speech signal is modeled by linear prediction filters and residues of this prediction.
  • the CELP coders as shown schematically in FIG. 1a and 1b, exhaustively test all the elements of a list of waveforms. The waveform providing the best synthesis of the signal is retained, and its index, or characteristic address, is transmitted to the decoder. This method is called synthesis analysis.
  • the list of waveforms stored in the encoder and the decoder is called a dictionary.
  • CELP coder The quality of a CELP coder depends closely on the dictionary chosen, on the method of determination-modeling of the linear prediction filters used, these two parameters constituting two degrees of freedom, not independent, making it possible to adapt a particular CELP coder to the needs of 'a specific application.
  • Such a CELP coding technique is suitable for low bit rate coding applications (between 4 to 24 kbits / s).
  • bit rate coding applications between 4 to 24 kbits / s.
  • the digital signal to be analyzed, transmitted and reconstructed is divided into blocks, or frames.
  • Each block comprising L values is considered as a vector of a vector space of dimension L.
  • the current excitation signal constituted by a vector v, read in the dictionary of waveforms, must minimize a criterion of perceptual distortion of the form: min ⁇ -Hv ⁇ 2 , in which ⁇ denotes a target signal derived from the original signal O to be transmitted after perceptual weighting and H denotes a matrix of dimension LxL of impulse response derived from the product of the transfer functions of the synthesis filter and the perceptual weighting.
  • each reference vector vi is associated with an adaptive gain value gk taken from a dictionary of gain values G, which allows, following application of the gain gk to the vector vi to form a vector v k , i , to satisfy the aforementioned minimum distortion criterion.
  • Such an operating mode therefore does not make it possible to take into account, as a reference vector, the totality of the possibilities of combinations of the ternary values of the components of the reference vectors, the minimization of the distortion criterion not being able in all cases to be optimal.
  • document EP-A-0 379 296 describes a method of transmitting a low-speed speech signal, of the CELP type. This method implements a process for calculating the minimum quadratic deviation of an initial vector and a reference vector from a vector dictionary factorized into the product of two dictionaries of basic and gain vectors. The gain is separated in amplitude and in sign and the calculation process consists in finding three indices satisfying the minimum quadratic error criterion.
  • the object of the present invention is to remedy the abovementioned drawbacks, in order in particular to bring about a simplification of the calculations by the introduction as reference vector, into the dictionary of reference vectors, or directions, of almost all the combinations of the n-ary values of the components of the vectors, n being an odd number.
  • Another object of the present invention is the implementation, prior to the conventional process of applying an adaptive gain to each of the reference vectors, of a correction process by applying a scale factor, introducing the distribution of the energy of the excitation signal as a function of the frequency spectrum thereof, in order to take account of the non-uniformity of the energy distribution of the signal in the frequency domain.
  • Another object of the present invention is finally the implementation of a low-speed transmission method of a speech signal in which each reference vector, constituting the excitation signal, can be regenerated at the level of a decoder from the only index or address values of the optimal reference vector satisfying the minimum distortion criterion at the level of the coder, which has the effect of considerably simplifying and reducing the manufacturing costs of the aforementioned decoders.
  • the method for transmitting a low bit rate speech signal comprises a process for coding digital speech samples by code-excited linear prediction to generate a code signal, a process for transmitting the code signal and a process for decoding the received code signal.
  • the coding process corresponds to a process in which a waveform is represented by a block of samples comprising L sample values and constituting an initial vector (o) of dimension L is represented, from a filter of synthesis, by a reference waveform selected from a dictionary of reference waveforms each forming a reference vector (v) on criterion of minimum quadratic deviation of the initial vector (o) with respect to the waveform or reference vector (v), min ⁇ -Hv ⁇ 2 , where ⁇ represents a target vector obtained by perceptual weighting of the initial vector (o) and H a LxL dimension matrix of impulse response from the product of the synthesis filter and the linear perceptual weighting.
  • n / 2 corresponds to the integer division of n by 2.
  • H. ⁇ i.yi> and all the energies perceptual ⁇ Hy ⁇ 2 , which allows to attribute to the initial vector (o) the corresponding optimal reference vector v k *, i * , with v k *, i * gk * . ⁇ i * .yi *, this vector of optimal reference being represented by the only values of index k *, i * satisfying the criterion min ⁇ -gk.H. ⁇ i.yi ⁇ 2 .
  • the process of transmission at low speed of a speech signal consists in transmitting, as code signal, the only values of the indices k *, i * representative of each optimal reference vector v k *, i * .
  • the process of decoding a coded speech signal transmitted at low bit rate according to a code signal is remarkable in that, in order to ensure the decoding of the code signal, this process consists in discriminating the values of the indices k *, i * constituting the code signal, in decomposing the value of the index i *, representative of the optimal reference vector, in base n to regenerate the corresponding base vector yi *, to be carried out, from the value of the index i *, of the scale factor ⁇ i * and of the corresponding adaptive gain gk *, a correction of the corresponding regenerated base vector to constitute the regenerated reference vector vk *, i *.
  • a synthesis filtering operation is performed on the regenerated reference vector vk *, i * to generate the reconstructed speech signal.
  • the method which is the subject of the invention comprises a process for coding digital speech samples by linear prediction excited by codes. This process generates a code signal.
  • the method further includes a process for transmitting the code signal and a process for decoding the received code signal.
  • the coding process corresponds to a process in which a waveform represented by a block of samples comprising L values of samples, or frames, constitutes an initial vector denoted by o of dimension L, this vector being represented, and the corresponding waveform, from a synthesis filter by a reference waveform, denoted v, selected from a dictionary of reference waveforms each forming an abovementioned reference vector.
  • the selection is made on the criterion of minimum quadratic deviation of the initial vector o vis-à-vis the waveform or reference vector v, this criterion written: min ⁇ -Hv ⁇ 2 .
  • ⁇ represents a target vector obtained by perceptual weighting of the initial vector o and H represents a matrix of dimension LxL of impulse response resulting from the product of the synthesis filter and from the aforementioned linear perceptual weighting.
  • the coding process is such that the selection criterion consists in establishing a dictionary factored into the product of a first dictionary Yn of basic vectors denoted yi.
  • Each basic vector is a basic vector of n-ary form, that is to say that the components yi (j) of these basic vectors, with j ⁇ [O, L-1], can take n values discrete different.
  • each basic vector yi is corrected by a scale factor ⁇ i taking into account the distribution of the excitation energy in the frequency domain of the signal.
  • the scale factors ⁇ i are determined from of a database, experimentally, the database being constituted by recording significant speech samples over several hours for example and for several speakers of the same language of expression or of several distinct languages, l experience showing that the diversity of languages of expression only intervenes at the second level in the determination of the aforementioned scale factors ⁇ i.
  • the scale factors vi are determined for each corresponding base vector yi by a process of identification of each base vector yi on a delocalized sequence of L successive recursive speech samples from the database , sorting the lowest adaptation coefficients and averaging a number u of identification or adaptation coefficients to obtain the corresponding scale factor vi associated with the aforementioned basic vector yi.
  • the factorized dictionary previously mentioned also consists of a second dictionary constituting the aforementioned product, this second dictionary being denoted G (y) and being formed by a dictionary of gains gk.
  • each scale factor coefficient vi is representative of the distribution of excitation energy in the frequency domain of a speech signal.
  • this optimal reference vector is represented by the only values of the parameters of index k *, i * satisfying the aforementioned criterion: min ⁇ -gk.H. ⁇ i.yi ⁇ 2 .
  • the minimum value of the quadratic deviation min ⁇ -gk.H. ⁇ i.yi ⁇ 2 is evaluated by selecting the corresponding gain element gk from the second dictionary G (y) making it possible to minimize the difference
  • where g checks the relation: g ⁇
  • the dictionary Yn of basic vectors yi of n-ary form [-n / 2, ..., O, ... n / 2] of dimension L includes all the basic vectors whose L components have for value the aforementioned n-values, with the exception of the null vector.
  • the index i of the basic vectors is taken equal to the value in base n of each base vector after transcoding the values [-n / 2 ..., O, ... n / 2] into values corresponding (0,1,2, ... n). It will thus be understood that the basic vectors yi of n-ary form are arranged as a function of their index i, this index i having for value the value in base n of each vector.
  • the set of basic vectors yi constituting the dictionary Yn is defined from n / 2.L impulse vectors of which only one component yi (j) of order j, with j ⁇ [O, L-1 ], is equal to -1, -2, ... -n / 2.
  • FIGS. 3a and 3b operator cells have been shown respectively making it possible to generate, from the previously defined pulse vectors and from sub-dictionaries constituted by the pulse vector considered and the related vectors corresponding to each pulse vector, the complete dictionary comprising the collection of all of the sub-dictionaries.
  • Each operator as represented in FIG. 3a comprises an operator called delay operator R whose transfer function is denoted Z +1 , according to the classical notation of transform into Z, a symmetrizer operator denoted Sy which has the function of multiplying the components of all the vectors presented at its entry by the value +1, by the value O then by the value -1 and a summator, noted S, receiving the output of the delay operator R and the symmetrizer Sy.
  • the summator S receives the output of the delay operator R via a switch I, in position F, or the zero vector [0,0,0,0,0] of dimension L in position O.
  • the operators represented in FIG. 3a are constituted by a single operator represented in 1), 2) and 3) at different stages of the processing process to generate the basic vectors yi of the abovementioned dictionary Yn.
  • the initial pulse or pulse vector ⁇ L-1 is present at the input of the delay operator R
  • the symmetrizer Sy is then supplied by a sub-dictionary noted DO ⁇ which is initially formed by the above-mentioned pulse vector ⁇ L-1.
  • Sy balancer delivers a sub-dictionary symmetrical noted DO ⁇ , as shown in FIG.
  • the pulse vector ⁇ L-2 is associated with the sub-dictionary D1 formed by the related vectors y1, y2, y3 with the pulse vector ⁇ L-2 and by the pulse vector to initial L-1 forming the basic vector y0, as well as the zero vector.
  • point 2 the summator S
  • the operator making it possible to generate the basic vectors yi is such that the latter receives the impulse vector at the level of the delay operator R at Lm, at the symmetrizer Sy, the dictionary denoted D m-1 formed recursively like the dictionary D1, the summator S as represented in point 2 of FIG. 3a then delivering from the above-mentioned pulse vector ⁇ Lm-1 delivered by the delay operator R or the zero vector and by the sub-dictionary D m-1 ⁇ the Dm dictionary.
  • the * represented at the level of the components yi (j) for the processing process of level m correspond to values 0, -1 or +1 when the vectors are ternary vectors.
  • the * represent values between -n / 2 and + n / 2, under the conditions previously mentioned.
  • the total ternary dictionary, sum or union of all the sub-dictionaries of intermediate level m, up to L can be obtained for the only positive or negative values of the components yi (j) , the dictionary total can then be obtained by symmetrization through a symmetrization operator such as Sy.
  • this operator is such that the impulse responses of the system H at the relative time 0, 1, 2, L-1, that is to say the values h 0 , h 1 , h L-2 , h L-1 are applied to the aforementioned operator.
  • the symmetrization operator Sy multiplies the elements of S L-1 (Dm-1) by +1, 0, -1 and realizes, as described above, the union of the distinct elements obtained.
  • FIGS. 5a and 5b A schematic diagram of the numbering and calculation process of the different entities implemented by the selection criterion in accordance with the object of the present invention will be described in connection with FIGS. 5a and 5b.
  • the elementary decryption cell is represented in Figure 5b from the pulse vectors noted not-1, ⁇ 0 and ⁇ 1. It will be noted that the summation of the pulse vectors e Frankfurts1, ⁇ 0 and ⁇ -1 amounts to replacing the last coordinate of the incident base vector by the component values +1, O or -1.
  • FIGS. 5a and 5b the architecture as represented in FIGS. 5a and 5b is that of a linear structure of ternary graphs. For an n-ary structure we get an n-ary graph.
  • the global graph for obtaining the energies is traversed from right to left, the initial energy E (0) being equal to SL-1 (0) 2 .
  • the elementary cell constituting the graph represented in FIG. 5c is represented in FIG. 5d.
  • each reference vector v k *, i * can advantageously be weighted by a factor of predicted level, noted a.
  • This predicted level factor a is representative of the average energy of the excitation signal estimated on at least three successive previous excitation vectors.
  • the previous expression is then calculated by filtering the expression 2x / ⁇ by the matrix transposed from the matrix H, ie H t .
  • the calculation process as represented by the operator in FIG. 6 allows, in a manner analogous to the calculation of the partial responses S L-1 (yi) previously described, to obtain the quantities x'0, x'Lm-1, x'L-2 and therefore the aforementioned scalar products, the null vector being replaced by the null value.
  • each scale factor vi can be determined from a plurality N of frames, from a speech signal database, the scale factor ⁇ i for each base vector yi being chosen so as to make the filter residue of the aforementioned frames minimal for the frame considered. It will be recalled that several processes for determining each scale factor vi can be envisaged.
  • the table of the scale factors vi is given below by the table of 121 values of the scale factors.
  • the first value multiplies (-1, -1, -1, -1, -1) ..., the last (0,0,0,0, -1).
  • the transmission of speech at low speed is effected by the only transmission, as code signal, of the values of the indices k * and i * representative of each reference vector v k *, i * .
  • the transmission can be carried out using conventional transmission protocols in which a redundancy of the information transmitted is introduced in order to ensure transmission at a rate substantially zero error.
  • the value i * can be transmitted either by direct numbering or by retrograde numbering, or according to a translated numbering whose translation table is known to the coder as well as to the decoder.
  • the decoding process consists in discriminating in 1000 the values of the indices k * and i * constituting the code signal and then in decomposing into 1001 the value of the index i * representative of the optimal reference vector in base n in order to regenerate the corresponding base vector yi *.
  • the decoding process consists in carrying out a synthesis filtering operation 1003 of the reference vector to generate the reconstructed speech signal.
  • each reference vector v k *, i * before the synthesis filtering is weighted by a predicted level factor ⁇ which is estimated on at least three excitation vectors successive previous.
  • the determination of the predicted level ⁇ will not be described in detail since it corresponds to the level of the process of decoding the operations normally known to those skilled in the art.
  • FIGS. 8 and 9 A more detailed description of a system for transmitting a low bit rate speech signal in accordance with the object of the present invention will be described in conjunction with FIGS. 8 and 9.
  • the coding circuit comprises a generator 1 of a first dictionary Yn of basic vectors yi of n-ary shape of dimension L, the components of these vectors, as mentioned previously, being able to take the values between -n / 2 to n / 2.
  • the generator of the dictionary Yn can advantageously be constituted by calculating means comprising the operators as described in FIGS. 3a, 3b for example and / or a storage circuit which can be constituted by a random access memory associated with this circuit computer or by ROM.
  • the read only memory is associated with a fast sequencer which makes it possible to carry out a successive reading of the basic vectors yi according to the indices in direct or retrograde numbering as described previously.
  • the coding circuit as represented in FIG. 8 comprises a circuit 2 correcting the basic vectors yi by a scale factor ⁇ i.
  • a fast multiplexer denoted MUX makes it possible successively to read the corresponding values of the corrected base vector yi ⁇ and deliver this value corresponding to a circuit 3 generator of a second adaptive gain dictionary gk.
  • the generator circuit 3 of the second dictionary G (y) may advantageously include an amplifier circuit, denoted 30, connected to a table of values gk constituting the aforementioned second dictionary.
  • the coding circuit which is the subject of the present invention also comprises an amplifier circuit 4 which makes it possible to apply to each reference vector v k, i the level prediction coefficient a as defined above. in the description.
  • the coding circuit object of the present invention then comprises, arranged in cascade, the synthesis filter noted 5 and the perceptual weighting filter noted 6 of transmission H as described previously in the description.
  • a summator 7 makes it possible to receive on the one hand the original signal via a same perceptual weighting filter 6 after inversion of the difference of the signals delivered by the algebraic summator 7, allowing the application on the signal thus obtained from the minimum distortion criterion.
  • the coding circuit which is the subject of the present invention comprises a circuit for calculating the minimum distortion 8 which comprises a first calculating circuit 80 of the product 2 gk ⁇ x / ⁇
  • the first calculator circuit 80 delivers a first calculation result r1.
  • a second calculator circuit 81 makes it possible to calculate the energy of the perceptually reconstructed and weighted vector, this energy being of the form gk 2 ⁇ H. ⁇ i.yi ⁇ 2 .
  • the computer circuits 80 and 81 can be constituted by program modules whose calculation graphs have been explained in FIGS. 4 and 5 a) to d) respectively.
  • the second calculation circuit 81 delivers a second calculation result denoted r2.
  • a comparator 83 makes it possible to compare the value of the calculation results r1 and r2 which makes it possible to determine by discrimination of the values of the indices i and k, the indices i * and k * for which the criterion of minimum of the quadratic difference is satisfied .
  • the discrimination of the indices i * and k * is carried out for example by a sorting program noted 84 in FIG. 8.
  • the values of the indices k * and i * are then delivered, these indices being representative of the corresponding reference vector v k * , i * .
  • FIG. 8 also shows the transmission circuit according to the subject of the present invention, this transmission circuit making it possible to deliver as a code signal representative of the speech signal the only values of the indices k * and i *.
  • This transmission circuit does not have any particular characteristic insofar as it can in fact be constituted by a transmission system of conventional type used in the devices for transmitting speech signals by coding of CELP type of the prior art.
  • FIG. 9 A more detailed description of a decoding circuit allowing the implementation of the method which is the subject of the invention is shown in FIG. 9.
  • the decoding circuit comprises a module 10 for discriminating the values of the indices i *, k * of the code signal received, the code signal being of course transmitted according to a particular protocol which does not enter into the object of the present invention.
  • the discrimination circuit 10 thus effecting a parallel series transformation of the information relating to the indices i *, k *
  • the decoding circuit comprises a circuit n-base decomposition of the value of the index i *.
  • the decoding circuit as shown in FIG. 9 comprises a table of the adaptive gain values gk denoted 11, which, on reception of the value of the index k *, makes it possible to deliver the corresponding adaptive gain value gk *.
  • This circuit 11 can advantageously consist of a read-only memory in which the adaptive gain values gk are stored.
  • a generator circuit 12 of the scale factor ⁇ i * is provided.
  • This circuit can consist of a read only memory forming a look-up table, which with the value i * makes correspond the value ⁇ i *.
  • the decoding circuit comprises a circuit 13 generating the regenerated base vector by decomposition into base n of the value of the index i *.
  • a circuit 14 corresponds to the value i * by transcoding the components in base n of the value of the index i *, the value [-n / 2, ..., 0, ... n / 2], which makes it possible to generate a regenerated reference vector v k *, i * of the product of the vector regenerated base and product A.
  • a synthesis filter 15 allows starting from the regenerated reference vector to generate the reconstructed speech signal.

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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)
EP91402774A 1990-10-19 1991-10-17 Procédé de transmission, à bas débit, par codage CELP d'un signal de parole et système correspondant Expired - Lifetime EP0481895B1 (fr)

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FR9012980 1990-10-19
FR9012980A FR2668288B1 (fr) 1990-10-19 1990-10-19 Procede de transmission, a bas debit, par codage celp d'un signal de parole et systeme correspondant.

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EP0481895A2 EP0481895A2 (fr) 1992-04-22
EP0481895A3 EP0481895A3 (en) 1992-08-12
EP0481895B1 true EP0481895B1 (fr) 1997-12-10

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Families Citing this family (18)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2658794B2 (ja) * 1993-01-22 1997-09-30 日本電気株式会社 音声符号化方式
WO1994025959A1 (en) * 1993-04-29 1994-11-10 Unisearch Limited Use of an auditory model to improve quality or lower the bit rate of speech synthesis systems
FR2722631B1 (fr) * 1994-07-13 1996-09-20 France Telecom Etablissement P Procede et systeme de filtrage adaptatif par egalisation aveugle d'un signal telephonique numerique et leurs applications
JP3058028B2 (ja) * 1994-10-31 2000-07-04 三菱電機株式会社 画像符号化データ再符号化装置
SE504010C2 (sv) * 1995-02-08 1996-10-14 Ericsson Telefon Ab L M Förfarande och anordning för prediktiv kodning av tal- och datasignaler
DE19516106C2 (de) * 1995-05-05 2003-04-03 Philips Corp Intellectual Pty Verfahren zum Bestimmen von Referenzwerten
US5864820A (en) * 1996-12-20 1999-01-26 U S West, Inc. Method, system and product for mixing of encoded audio signals
US6477496B1 (en) 1996-12-20 2002-11-05 Eliot M. Case Signal synthesis by decoding subband scale factors from one audio signal and subband samples from different one
US5864813A (en) * 1996-12-20 1999-01-26 U S West, Inc. Method, system and product for harmonic enhancement of encoded audio signals
US5845251A (en) * 1996-12-20 1998-12-01 U S West, Inc. Method, system and product for modifying the bandwidth of subband encoded audio data
US6516299B1 (en) 1996-12-20 2003-02-04 Qwest Communication International, Inc. Method, system and product for modifying the dynamic range of encoded audio signals
US6782365B1 (en) 1996-12-20 2004-08-24 Qwest Communications International Inc. Graphic interface system and product for editing encoded audio data
US6463405B1 (en) 1996-12-20 2002-10-08 Eliot M. Case Audiophile encoding of digital audio data using 2-bit polarity/magnitude indicator and 8-bit scale factor for each subband
FI113571B (fi) 1998-03-09 2004-05-14 Nokia Corp Puheenkoodaus
US7536298B2 (en) * 2004-03-15 2009-05-19 Intel Corporation Method of comfort noise generation for speech communication
JP4993676B2 (ja) * 2006-09-01 2012-08-08 キヤノン株式会社 画像符号化装置及び画像符号化方法
CN101430879B (zh) * 2007-11-05 2011-08-10 华为技术有限公司 一种多速率语音频编码的方法
PT3364411T (pt) * 2009-12-14 2022-09-06 Fraunhofer Ges Forschung Dispositivo de quantização de vetor, dispositivo de codificação de voz, método de quantização de vetor e método de codificação de voz

Family Cites Families (11)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
NL8302985A (nl) * 1983-08-26 1985-03-18 Philips Nv Multipulse excitatie lineair predictieve spraakcodeerder.
NL8500843A (nl) * 1985-03-22 1986-10-16 Koninkl Philips Electronics Nv Multipuls-excitatie lineair-predictieve spraakcoder.
US4944013A (en) * 1985-04-03 1990-07-24 British Telecommunications Public Limited Company Multi-pulse speech coder
IT1195350B (it) * 1986-10-21 1988-10-12 Cselt Centro Studi Lab Telecom Procedimento e dispositivo per la codifica e decodifica del segnale vocale mediante estrazione di para metri e tecniche di quantizzazione vettoriale
US4868867A (en) * 1987-04-06 1989-09-19 Voicecraft Inc. Vector excitation speech or audio coder for transmission or storage
US4910781A (en) * 1987-06-26 1990-03-20 At&T Bell Laboratories Code excited linear predictive vocoder using virtual searching
US4899385A (en) * 1987-06-26 1990-02-06 American Telephone And Telegraph Company Code excited linear predictive vocoder
DE68923771T2 (de) * 1988-12-23 1995-12-14 Nippon Electric Co Sprachübertragungssystem unter Anwendung von Mehrimpulsanregung.
CA2005115C (en) * 1989-01-17 1997-04-22 Juin-Hwey Chen Low-delay code-excited linear predictive coder for speech or audio
US4980916A (en) * 1989-10-26 1990-12-25 General Electric Company Method for improving speech quality in code excited linear predictive speech coding
CA2010830C (en) * 1990-02-23 1996-06-25 Jean-Pierre Adoul Dynamic codebook for efficient speech coding based on algebraic codes

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
Academic Publishers, Dordrecht, NL; R.A. SALAMI: "Binary pulse excitation: a novel approach to low complexity CELP coding" *

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DE69128407D1 (de) 1998-01-22
FR2668288A1 (fr) 1992-04-24
EP0481895A3 (en) 1992-08-12
DE69128407T2 (de) 1998-06-04
EP0481895A2 (fr) 1992-04-22
FR2668288B1 (fr) 1993-01-15
JP3130348B2 (ja) 2001-01-31
JPH04264500A (ja) 1992-09-21
US5226085A (en) 1993-07-06

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