CA2261956A1 - Method and apparatus for searching an excitation codebook in a code excited linear prediction (clep) coder - Google Patents
Method and apparatus for searching an excitation codebook in a code excited linear prediction (clep) coder Download PDFInfo
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- CA2261956A1 CA2261956A1 CA002261956A CA2261956A CA2261956A1 CA 2261956 A1 CA2261956 A1 CA 2261956A1 CA 002261956 A CA002261956 A CA 002261956A CA 2261956 A CA2261956 A CA 2261956A CA 2261956 A1 CA2261956 A1 CA 2261956A1
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L15/00—Speech recognition
- G10L15/02—Feature extraction for speech recognition; Selection of recognition unit
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; 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
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; 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
Abstract
Method and apparatus for selecting a code vector in an algebraic codebook wherein the analysis window for the coder is extended beyond the length of the target speech frame. An input signal is filtered by a perceptual weighting filter (76). Then, the filter is set to ring out for a number of samples equal to the length of the perceptual weighting filter (76), while a zero input vector is applied as input. By extending the analysis window, the two dimensional impulse response matrix can be stored as a one dimensional autocorrelation matrix in memory (60, 80), greatly saving on the computational complexity and memory required for the search.
Description
CA 022619~6 1999-01-29 W O38,'1~'C PCT~US97/13594 METHOD AND APPARATUS FOR SEARCHING AN
EXCITATION CODEBOOK IN A CODE EX~ L) LINEAR
PREDICTION (CELP) CODER
BACKGROUND OF THE INVENTION
I. ~ield of the Invention The present invention relates to speech processing. More 10 particularly, the present invention relates to a novel and improved method and apparatus for locating an optimal excitation vector in a code excited linear prediction (CELP) coder.
II. Description of the Related Art ~ ransmission of voice by digital techniques has become widespread, particularly in long distance and digital radio telephone applications. This in turn has created interest in determining methods which minimize the amount of information sent over the transmission channel while 20 maintaining high quality in the reconstructed speech. If speech is transmitted by simply sampling and digitizing, a data rate on the order of 64 kilobits per second (kbps) is required to achieve a speech quality of conventional analog telephone. However, through the use of speech analysis, followed by the appropriate coding, transmission, and resynthesis 25 at the receiver, a significant reduction in the data rate can be achieved.
Devices which employ techniques to compress voiced speech by extracting parameters that relate to a model of human speech generation are typically called vocoders. Such devices are composed of an encoder, which analyzes the incoming speech to extract the relevant parameters, and a 30 decoder, which resynthesizes the speech using the parameters which it receives over the transmission channel. The model is constantly changing ~ to accuràtely'model the time varying speech signal. Thus, the speech is divided into blocks of time, or analysis frames, during-which the parameters are calculated. The parameters are then updated for each new frame.
Of the various classes of speech coders, the Code Excited Linear Predictive Coding (CELP), Stochastic Coding, or Vector Excited Speech Coding coders are of one class. An example of a coding algorithm of this particular class is described in the paper "A 4.8 kbps Code Excited Linear Predictive Coder" by Thomas E. Tremain et al., Proceedings of the Mobile 40 Satel}ite Conference, 1988. Similarly, examples of other vocoders of this CA 022619~6 1999-01-29 W 0 98/05030 PCT~US97113594 type are detailed in U.S. Patent No. 5,414,796, entitled "Variable Rate Vocoder" and assigned to the assignee of the present invention and incorporated by reference herein.
The function of the vocoder is to compress the digitized speech signal 5 into a low bit rate signal by removing all of the natural redundancies inherent in speech. In a CELP coder, redundancies are removed by means of a short term formant (or LPC) filter. Once these redundancies are removed, the resulting residual signal can be modeled as white Gaussian noise, which also must be encoded.
The process of determining the coding parameters for a given frame of speech is as follows. First, the parameters of the LPC filter are determined by finding the filter coefficients which remove the short term redundancy, due to the vocal tract filtering, in the speech. Next, an excitation signal, which is input to LPC filter at the decoder, is chosen by driving the LPC filter15 with a number of random excitation waveforms in a codebook, and selecting the particular excitation waveform which causes the output of the LPC filter to be the closest approximation to the original speech. Thus, the transmitted parameters relate to (1) the LPC filter and (2) an identification ofthe codebook excitation vector.
A promising excitation codebook structure is referred to as an algebraic codebook. The actual structure of algebraic codebooks is well known in the art and is described in the paper "Fast CELP coding based on Algebraic Codes" by J.P. Adoul, et al., Proceedings of ICASSP April 6-9, 1987.
The use of algebraic codes is further disclosed in U.S. Patent No. 5,444,816, 25 entitled "Dynamic Codebook for Efficient Speech Coding Based on Algebraic Codes", the disclosure of which is incorporated by reference.
SUMMARY OF THE INVENTION
Ana1ysis by synthesis based CELP coders use a minimum mean square error measure to match the best synthesized speech vector to the target speech vector. This measure is used to search the codevector codebook to choose the optimum vector for the current subframe. This mean square error measure is typically limited to the window over which 35 the excitation codevector is being chosen and thus fails to account for the contribution this codevector will make on the next subframe being searched.
In the present invention, the window size over which the mean square error measure is minimized is extended to account for this ringing of CA 022619~6 1999-01-29 the codevector in the current subframe into the next subframe. The window extension is equal to the length of the impulse response of the perceptual weighting filter, h(n). The mean square error approach in the current invention is analogous to the autocorrelation approach to the 5 minimum mean square error used in LPC analysis as described in the paper "A 4.8kbps Code Excited Linear Predictive Coder" by Thomas E. Tremain et al., Proceedings of the Mobile Satellite Conference. 1988.
Formulating the mean square error problem from this perspective, the present invention has the following advantages over the current 10 approach:
1.) The ringing of the codevector from the current subframe to the next subframe is accounted for in the measure and thus pu~ses placed at the end o~ the vector are weighted equivalently to pulses placed at the beginning of the vector.
EXCITATION CODEBOOK IN A CODE EX~ L) LINEAR
PREDICTION (CELP) CODER
BACKGROUND OF THE INVENTION
I. ~ield of the Invention The present invention relates to speech processing. More 10 particularly, the present invention relates to a novel and improved method and apparatus for locating an optimal excitation vector in a code excited linear prediction (CELP) coder.
II. Description of the Related Art ~ ransmission of voice by digital techniques has become widespread, particularly in long distance and digital radio telephone applications. This in turn has created interest in determining methods which minimize the amount of information sent over the transmission channel while 20 maintaining high quality in the reconstructed speech. If speech is transmitted by simply sampling and digitizing, a data rate on the order of 64 kilobits per second (kbps) is required to achieve a speech quality of conventional analog telephone. However, through the use of speech analysis, followed by the appropriate coding, transmission, and resynthesis 25 at the receiver, a significant reduction in the data rate can be achieved.
Devices which employ techniques to compress voiced speech by extracting parameters that relate to a model of human speech generation are typically called vocoders. Such devices are composed of an encoder, which analyzes the incoming speech to extract the relevant parameters, and a 30 decoder, which resynthesizes the speech using the parameters which it receives over the transmission channel. The model is constantly changing ~ to accuràtely'model the time varying speech signal. Thus, the speech is divided into blocks of time, or analysis frames, during-which the parameters are calculated. The parameters are then updated for each new frame.
Of the various classes of speech coders, the Code Excited Linear Predictive Coding (CELP), Stochastic Coding, or Vector Excited Speech Coding coders are of one class. An example of a coding algorithm of this particular class is described in the paper "A 4.8 kbps Code Excited Linear Predictive Coder" by Thomas E. Tremain et al., Proceedings of the Mobile 40 Satel}ite Conference, 1988. Similarly, examples of other vocoders of this CA 022619~6 1999-01-29 W 0 98/05030 PCT~US97113594 type are detailed in U.S. Patent No. 5,414,796, entitled "Variable Rate Vocoder" and assigned to the assignee of the present invention and incorporated by reference herein.
The function of the vocoder is to compress the digitized speech signal 5 into a low bit rate signal by removing all of the natural redundancies inherent in speech. In a CELP coder, redundancies are removed by means of a short term formant (or LPC) filter. Once these redundancies are removed, the resulting residual signal can be modeled as white Gaussian noise, which also must be encoded.
The process of determining the coding parameters for a given frame of speech is as follows. First, the parameters of the LPC filter are determined by finding the filter coefficients which remove the short term redundancy, due to the vocal tract filtering, in the speech. Next, an excitation signal, which is input to LPC filter at the decoder, is chosen by driving the LPC filter15 with a number of random excitation waveforms in a codebook, and selecting the particular excitation waveform which causes the output of the LPC filter to be the closest approximation to the original speech. Thus, the transmitted parameters relate to (1) the LPC filter and (2) an identification ofthe codebook excitation vector.
A promising excitation codebook structure is referred to as an algebraic codebook. The actual structure of algebraic codebooks is well known in the art and is described in the paper "Fast CELP coding based on Algebraic Codes" by J.P. Adoul, et al., Proceedings of ICASSP April 6-9, 1987.
The use of algebraic codes is further disclosed in U.S. Patent No. 5,444,816, 25 entitled "Dynamic Codebook for Efficient Speech Coding Based on Algebraic Codes", the disclosure of which is incorporated by reference.
SUMMARY OF THE INVENTION
Ana1ysis by synthesis based CELP coders use a minimum mean square error measure to match the best synthesized speech vector to the target speech vector. This measure is used to search the codevector codebook to choose the optimum vector for the current subframe. This mean square error measure is typically limited to the window over which 35 the excitation codevector is being chosen and thus fails to account for the contribution this codevector will make on the next subframe being searched.
In the present invention, the window size over which the mean square error measure is minimized is extended to account for this ringing of CA 022619~6 1999-01-29 the codevector in the current subframe into the next subframe. The window extension is equal to the length of the impulse response of the perceptual weighting filter, h(n). The mean square error approach in the current invention is analogous to the autocorrelation approach to the 5 minimum mean square error used in LPC analysis as described in the paper "A 4.8kbps Code Excited Linear Predictive Coder" by Thomas E. Tremain et al., Proceedings of the Mobile Satellite Conference. 1988.
Formulating the mean square error problem from this perspective, the present invention has the following advantages over the current 10 approach:
1.) The ringing of the codevector from the current subframe to the next subframe is accounted for in the measure and thus pu~ses placed at the end o~ the vector are weighted equivalently to pulses placed at the beginning of the vector.
2.) The impulse response of the perceptual weighting filter becomes stationary for the entire subframe making the autocorrelation matrix of h(n), ~)(i,j), Toeplitz, or stated another way, ~)(i,j) = ~) li-jl . Thus the present invention turns a 2-D matrix into a 1-D vector and thus reduces RAM requirements for the codebook search as well as computational 20 operations.
BRIEF DESCRIPTION OF THE DRAWINGS
The features, objects, and advantages of the present invention will 25 become more apparent from the detailed description set forth below when taken in conjunction with the drawings in which like reference characters identify correspondingly throughout and wherein:
FIG. 1 is an illustration of the traditional apparatus for selecting a code vector in an ACELP coder;
FIG. 2 is a block diagram of the apparatus of the present invention for selecting a code vector in an ACELP coder; and FIG. 3 is a flowchart describing the method for selecting a code vector in the present invention.
DETAILED DESCRIPTION OF THE PREFERRED
EMBODIMENTS
FIG. 1 illustrates the traditional apparatus and method used to perform an algebraic codebook search. Codebook generator 6 includes a , , ~ .
W 098105030 PCTrUS97/13594 pulse generator 2 which in response to a pulse position signal, Pi, generates a signal with a unit pulse in the ith position. In the exemplary embodiment, the codebook excitation vector comprises forty samples and the possible positions for the unit impulse are divided into tracks TO to T4 5 as shown in TABLE 1 below.
Track Positions T0 0,5,10,15,20,25,30,35 T1 1,6,11,16,21,26,31,36 T2 2, 7, 12, 17, 22, 27, 32, 37 T3 3,8, 13, 18,23,28,33,38 T4 4, 9, 10, 19, 24, 29, 34, 39 In the exemplary embodiment, one pulse is provided for each track by pulse 10 generator 2. Np is the number of pulses in an excitation vector. In the exemplary embodiment, Np is 5. For each pulse, pi, a corresponding sign si is assigned to the pulse. The sign of the pulse which is illustrated by multiplier 4 which multiplies the unit impulse at position, pi, by the sign value, si. The resulting code vector, ck, is given by equation (1) below.
Np -I
ck(j) = ~ Sj S(j-pj) (1) i=O
Filter generator 12 generates the tap values for formant filter, h(n), as is well known in the art and described in detail in the aforementioned U.S.
20 Patent No. 5,414,796. Typically, the impulse function, h(n), would be computed for M samples where M is the length of the subframe being searched, for example 40.
The composite filter coefficients, h(n), are provided to and stored as two dimensional triangular Toeplitz matrix (H) in memory element 13 25 where the diagonal is h(0) and the lower diagonals are h(1)..., h(M-1) as shown below.
~ h(0) 0 0 -- 0 ~
h(1) h(0) 0 0 H =h(2) h(1) h(0) ~ (2) .
h(~l - 1) h(M - 2) h(M - 3) - h(0) .
CA 0226l956 l999-0l-29 W O 98/05030 PCTrUS97/13594 s The values are provided by memory 13 to matrix multiplication element 14. H is then multiplied by its transpose to give the correlation of the impulse response matrix (P in accordance with equation (3) below.
M
~P(i,j) = Ht ~ H= ~,h(n - i)h(n - j), for i > j (3) n=j The result of the correlation operation is then provided to memory element 18 and stored as a two dimensional matrix which requires 402 or 1600 10 positions of memory for this embodiment.
The input speech frame s(n) is provided to and filtered by perceptual weighting filter 32 to provide the target signal, x(n). The design and implementation of perceptual weighting filter 32 is well known in the art and is described in detail in the aforementioned U.S. Patent No. 5,414,796.
The sample values of the target signal, x(n), and values of the impulse matrix, H(n), are provided to matrix multiplication element 16 which computes the cross corre}ation between the target signal and the impulse response in accordance with equation (4) below.
M
d(i)= Ht ~ x= ~x(i)h(i - j), for j=0 to M.
j=j The values from memory element 20, d(i), and the codebook vector amplitude elements, ck, are provided to matrix multiplication element 22 which multiplies the codebook vector amplitude elements by the vector d(n) and squares the resulting value in accordance with equation (5) below.
~Np-l ~2 E2y= ~,ck(p;) d(pj) (5) ~ i=O
Codebook vector amplitude elements, ck, and codebook pulse positioning vector p are provided to matrix multiplication element 26.
Matrix multiplication element 26 computes the value, Eyy, in accordance with equation (6) below.
Np-l Np-l Np-l Eyy= ~,~(pi,pj)+2 ~, ~,ck(pj)ck(pj)~(pi~p;) (6) i=o i=o j=i+l CA 0226l9~6 l999-0l-29 The values of Eyy and (Exy)2 are provided to divider 28, which computes the value Tk in accordance with equation (7) below.
Tk= ( E ) (7) The values Tk for each codebook vector amplitude element, ck, and codebook pulse positioning vector p are provided to minimization element 30 and the codebook vector that maximizes the value Tk is selected.
Referring to FIG. 2, the apparatus for selecting the code vector in the present invention is illustrated. In FIG. 3, a flowchart describing the operational flow of the present invention is illustrated. First in block 100, the present invention precomputes the values of d(k), which can be computed ahead of time and stored since its values do not change with the code vector being searched.
The speech frame, s(n) is provided to perceptual weighting filter 76 which generates the target signal, x(n). The resulting target speech segment, x(n), consists of M+L-1 perceptually weighted samples which are provided to multiply and accumulate element 78. L is the length of the impulse response of perceptual weighting filter 76. This extended length target speech vector, x(n), is created by filtering M samples of the speech signal through the perceptual weighting filter 76 and then continuing to let this filter ring out for L-1 additional samples while a zero input vector is applied as input to perceptual weighting filter 76.
As deseribed previously with respect to filter generator 12, filter generator 56 computes the filter tap coefficients for the formant filter and from those coefficients determines the impulse response, h(n). However filter generator 56 generates a filter response for delays from 0 to L-1, where L is the length of the impulse response, h(n). It should be noted that 30 though, described in the exemplary embodiment, without a pitch filter the present invention is equally applicable for cases where there is a pitch filter by simple modification of the impulse response as is well known in the art.
The values of h(n) from filter generator 56 are provided to multiply and accumulate element 78. Multiply and accumulate element 78 computes 35 the cross correlation of the target sequence, x(n), with the filter impulse response, h(n), in accordance with equation (8) below.
CA 0226l9~6 l999-0l-29 W O9~/05030 PCTrUS97/13594 n+L-l d(n)= ~,x(n)h(n- j), for n=0 to M-1. (8) . j=n The computed values of d(n) are then stored in memory element 80.
In block 102, the present invention precomputes the values of ~
5 needed for the computation of Eyy. It is at this point where the biggest gain in memory savings of the present invention is reali~e~l. Because the mean square error measure has been extended over a larger window, h(n3 is now stationary over the entire subframe and consequently the 2-D ~(i,j) matrix becomes a 1-D vector because (~(i,j) = ~)( l i-j I ). In the present embodiment 10 as described in Table 1, this means that the traditional method requires 1600Ram locations while the present invention requires only 40. Operation count savings are also obtained in the computation and store of the 1-D
vector over the 2-D matrix also. In the present invention, the values of are computed in accordance with equation (9) below.
~(i)= ~ h(n)h(n - i) (9) n=0 The values of ~(i) are stored in memory element 80, which only requires L
memory locations, as opposed to the traditional method which requires the 20 storage of M2 elements. In this embodiment, L=M.
In block 104, the present invention computes the cross correlation valùe Exy. The values of d(k) stored in memory element 80 and the current codebook vector ci(k) from codebook generator 50 are provided to multiply and accumulate element 62. Multiply and accumulate element 62 computes 25 the cross correlation of the target vector, x(k), and the codebook vector amplitude elements, ci(k) in accordance with equation (10).
Np Exy = ~,Cj(pk) d(Pk) (10) k=O
30 The value of Exy is then provided to squaring means 64 which computes the square of Exy.
In block 106, the present invention computes the value of the autocorrelation of the synthesized speech, Eyy. The codebook vector amplitude elements cj(k) and cj(k) are provided from codebook generator 50 35 to multiply and accumulate element 70. In addition, the values of ~) l i-j I
are provided to multiply and accumulate element 70 from memory element CA 022619~6 1999-01-29 W O~8'GS~0 PCTAUS97tl3594 60. Multiply and accumulate element 70 computes the value given in equation (11) below.
Np Np ~, ~,Ck(pj)-ck(Pi) ~) Pi Pi (11) i=O j=i+l The value computed by multiply and accumulate means 70 is provided to multiplier 72 where its value is multiplied by 2. The product from multiplier 72 is provided to a first input of summer 74.
Memory element 60 provides the value of q)(0) to multiplier 75 10 where it is multiplied by the value Np. The product from multiplier 75 is provided to a second input of summer 74. ~he sum from summer 74 is the value Eyy which is given by equation (12) below.
Np Np Eyy =Np ~(0)+2 ~, ~,ck(pj) ck(pj) ~)lpj--pjI (12) i=O 1=i+l ~5 An appreciation of the savings of computational resource can be attained by comparing equation (12) of the present invention with equation ~6) of the traditional search method. This savings results from faster addressing of a 1-D matrix (~ l pi-pj I ) over a 2-D access of ~(pi,pj), from less adds required20 for Eyy computation (for the exemplary embodiment equation (6) takes 15 adds while equation (12) takes 11 assuming ck(pi) are just 1 or -1 sign terms), and from the 1360 Ram location savings since ~'(i,j) does not need to be stored.
In block 108, the present invention computes the value of (EXy32/Eyy.
25 The value of Eyy from summing element 74 is provided to a first input of divider 66. The value of (Exy)2 is provided from squaring means 64 is provided to the second input of divider 66. Divider 66 then computes the quotient given in equation (13) below.
xy (13) Eyy The quotient value from divider 66 is provided to minimization element 66. In block 110, if the all vectors Ck have not been tested the flow moves back to block 104 and the next code vector is tested as described 35 above. If all vectors have been tested then, in block 112, minimization _ CA 022619~6 1999-01-29 W 0 98/0S030 PCT~US97/13594 element 68 selects the code vector which results in the maximum value of (EXy)2/Eyy~
The previous description of the preferred embodiments is provided to enable any person skilled in the art to make or use the present invention.
5 The various modifications to these embodiments will be readily apparent to those skilled in the art, and the generic principles defined herein may be applied to other embodiments without the use of the inventive faculty.
Thus, the present invention is not intended to be limited to the embodiments shown herein but is to be accorded the widest scope consistent 10 with the principles and novel features disclosed herein.
I CLAIM:
BRIEF DESCRIPTION OF THE DRAWINGS
The features, objects, and advantages of the present invention will 25 become more apparent from the detailed description set forth below when taken in conjunction with the drawings in which like reference characters identify correspondingly throughout and wherein:
FIG. 1 is an illustration of the traditional apparatus for selecting a code vector in an ACELP coder;
FIG. 2 is a block diagram of the apparatus of the present invention for selecting a code vector in an ACELP coder; and FIG. 3 is a flowchart describing the method for selecting a code vector in the present invention.
DETAILED DESCRIPTION OF THE PREFERRED
EMBODIMENTS
FIG. 1 illustrates the traditional apparatus and method used to perform an algebraic codebook search. Codebook generator 6 includes a , , ~ .
W 098105030 PCTrUS97/13594 pulse generator 2 which in response to a pulse position signal, Pi, generates a signal with a unit pulse in the ith position. In the exemplary embodiment, the codebook excitation vector comprises forty samples and the possible positions for the unit impulse are divided into tracks TO to T4 5 as shown in TABLE 1 below.
Track Positions T0 0,5,10,15,20,25,30,35 T1 1,6,11,16,21,26,31,36 T2 2, 7, 12, 17, 22, 27, 32, 37 T3 3,8, 13, 18,23,28,33,38 T4 4, 9, 10, 19, 24, 29, 34, 39 In the exemplary embodiment, one pulse is provided for each track by pulse 10 generator 2. Np is the number of pulses in an excitation vector. In the exemplary embodiment, Np is 5. For each pulse, pi, a corresponding sign si is assigned to the pulse. The sign of the pulse which is illustrated by multiplier 4 which multiplies the unit impulse at position, pi, by the sign value, si. The resulting code vector, ck, is given by equation (1) below.
Np -I
ck(j) = ~ Sj S(j-pj) (1) i=O
Filter generator 12 generates the tap values for formant filter, h(n), as is well known in the art and described in detail in the aforementioned U.S.
20 Patent No. 5,414,796. Typically, the impulse function, h(n), would be computed for M samples where M is the length of the subframe being searched, for example 40.
The composite filter coefficients, h(n), are provided to and stored as two dimensional triangular Toeplitz matrix (H) in memory element 13 25 where the diagonal is h(0) and the lower diagonals are h(1)..., h(M-1) as shown below.
~ h(0) 0 0 -- 0 ~
h(1) h(0) 0 0 H =h(2) h(1) h(0) ~ (2) .
h(~l - 1) h(M - 2) h(M - 3) - h(0) .
CA 0226l956 l999-0l-29 W O 98/05030 PCTrUS97/13594 s The values are provided by memory 13 to matrix multiplication element 14. H is then multiplied by its transpose to give the correlation of the impulse response matrix (P in accordance with equation (3) below.
M
~P(i,j) = Ht ~ H= ~,h(n - i)h(n - j), for i > j (3) n=j The result of the correlation operation is then provided to memory element 18 and stored as a two dimensional matrix which requires 402 or 1600 10 positions of memory for this embodiment.
The input speech frame s(n) is provided to and filtered by perceptual weighting filter 32 to provide the target signal, x(n). The design and implementation of perceptual weighting filter 32 is well known in the art and is described in detail in the aforementioned U.S. Patent No. 5,414,796.
The sample values of the target signal, x(n), and values of the impulse matrix, H(n), are provided to matrix multiplication element 16 which computes the cross corre}ation between the target signal and the impulse response in accordance with equation (4) below.
M
d(i)= Ht ~ x= ~x(i)h(i - j), for j=0 to M.
j=j The values from memory element 20, d(i), and the codebook vector amplitude elements, ck, are provided to matrix multiplication element 22 which multiplies the codebook vector amplitude elements by the vector d(n) and squares the resulting value in accordance with equation (5) below.
~Np-l ~2 E2y= ~,ck(p;) d(pj) (5) ~ i=O
Codebook vector amplitude elements, ck, and codebook pulse positioning vector p are provided to matrix multiplication element 26.
Matrix multiplication element 26 computes the value, Eyy, in accordance with equation (6) below.
Np-l Np-l Np-l Eyy= ~,~(pi,pj)+2 ~, ~,ck(pj)ck(pj)~(pi~p;) (6) i=o i=o j=i+l CA 0226l9~6 l999-0l-29 The values of Eyy and (Exy)2 are provided to divider 28, which computes the value Tk in accordance with equation (7) below.
Tk= ( E ) (7) The values Tk for each codebook vector amplitude element, ck, and codebook pulse positioning vector p are provided to minimization element 30 and the codebook vector that maximizes the value Tk is selected.
Referring to FIG. 2, the apparatus for selecting the code vector in the present invention is illustrated. In FIG. 3, a flowchart describing the operational flow of the present invention is illustrated. First in block 100, the present invention precomputes the values of d(k), which can be computed ahead of time and stored since its values do not change with the code vector being searched.
The speech frame, s(n) is provided to perceptual weighting filter 76 which generates the target signal, x(n). The resulting target speech segment, x(n), consists of M+L-1 perceptually weighted samples which are provided to multiply and accumulate element 78. L is the length of the impulse response of perceptual weighting filter 76. This extended length target speech vector, x(n), is created by filtering M samples of the speech signal through the perceptual weighting filter 76 and then continuing to let this filter ring out for L-1 additional samples while a zero input vector is applied as input to perceptual weighting filter 76.
As deseribed previously with respect to filter generator 12, filter generator 56 computes the filter tap coefficients for the formant filter and from those coefficients determines the impulse response, h(n). However filter generator 56 generates a filter response for delays from 0 to L-1, where L is the length of the impulse response, h(n). It should be noted that 30 though, described in the exemplary embodiment, without a pitch filter the present invention is equally applicable for cases where there is a pitch filter by simple modification of the impulse response as is well known in the art.
The values of h(n) from filter generator 56 are provided to multiply and accumulate element 78. Multiply and accumulate element 78 computes 35 the cross correlation of the target sequence, x(n), with the filter impulse response, h(n), in accordance with equation (8) below.
CA 0226l9~6 l999-0l-29 W O9~/05030 PCTrUS97/13594 n+L-l d(n)= ~,x(n)h(n- j), for n=0 to M-1. (8) . j=n The computed values of d(n) are then stored in memory element 80.
In block 102, the present invention precomputes the values of ~
5 needed for the computation of Eyy. It is at this point where the biggest gain in memory savings of the present invention is reali~e~l. Because the mean square error measure has been extended over a larger window, h(n3 is now stationary over the entire subframe and consequently the 2-D ~(i,j) matrix becomes a 1-D vector because (~(i,j) = ~)( l i-j I ). In the present embodiment 10 as described in Table 1, this means that the traditional method requires 1600Ram locations while the present invention requires only 40. Operation count savings are also obtained in the computation and store of the 1-D
vector over the 2-D matrix also. In the present invention, the values of are computed in accordance with equation (9) below.
~(i)= ~ h(n)h(n - i) (9) n=0 The values of ~(i) are stored in memory element 80, which only requires L
memory locations, as opposed to the traditional method which requires the 20 storage of M2 elements. In this embodiment, L=M.
In block 104, the present invention computes the cross correlation valùe Exy. The values of d(k) stored in memory element 80 and the current codebook vector ci(k) from codebook generator 50 are provided to multiply and accumulate element 62. Multiply and accumulate element 62 computes 25 the cross correlation of the target vector, x(k), and the codebook vector amplitude elements, ci(k) in accordance with equation (10).
Np Exy = ~,Cj(pk) d(Pk) (10) k=O
30 The value of Exy is then provided to squaring means 64 which computes the square of Exy.
In block 106, the present invention computes the value of the autocorrelation of the synthesized speech, Eyy. The codebook vector amplitude elements cj(k) and cj(k) are provided from codebook generator 50 35 to multiply and accumulate element 70. In addition, the values of ~) l i-j I
are provided to multiply and accumulate element 70 from memory element CA 022619~6 1999-01-29 W O~8'GS~0 PCTAUS97tl3594 60. Multiply and accumulate element 70 computes the value given in equation (11) below.
Np Np ~, ~,Ck(pj)-ck(Pi) ~) Pi Pi (11) i=O j=i+l The value computed by multiply and accumulate means 70 is provided to multiplier 72 where its value is multiplied by 2. The product from multiplier 72 is provided to a first input of summer 74.
Memory element 60 provides the value of q)(0) to multiplier 75 10 where it is multiplied by the value Np. The product from multiplier 75 is provided to a second input of summer 74. ~he sum from summer 74 is the value Eyy which is given by equation (12) below.
Np Np Eyy =Np ~(0)+2 ~, ~,ck(pj) ck(pj) ~)lpj--pjI (12) i=O 1=i+l ~5 An appreciation of the savings of computational resource can be attained by comparing equation (12) of the present invention with equation ~6) of the traditional search method. This savings results from faster addressing of a 1-D matrix (~ l pi-pj I ) over a 2-D access of ~(pi,pj), from less adds required20 for Eyy computation (for the exemplary embodiment equation (6) takes 15 adds while equation (12) takes 11 assuming ck(pi) are just 1 or -1 sign terms), and from the 1360 Ram location savings since ~'(i,j) does not need to be stored.
In block 108, the present invention computes the value of (EXy32/Eyy.
25 The value of Eyy from summing element 74 is provided to a first input of divider 66. The value of (Exy)2 is provided from squaring means 64 is provided to the second input of divider 66. Divider 66 then computes the quotient given in equation (13) below.
xy (13) Eyy The quotient value from divider 66 is provided to minimization element 66. In block 110, if the all vectors Ck have not been tested the flow moves back to block 104 and the next code vector is tested as described 35 above. If all vectors have been tested then, in block 112, minimization _ CA 022619~6 1999-01-29 W 0 98/0S030 PCT~US97/13594 element 68 selects the code vector which results in the maximum value of (EXy)2/Eyy~
The previous description of the preferred embodiments is provided to enable any person skilled in the art to make or use the present invention.
5 The various modifications to these embodiments will be readily apparent to those skilled in the art, and the generic principles defined herein may be applied to other embodiments without the use of the inventive faculty.
Thus, the present invention is not intended to be limited to the embodiments shown herein but is to be accorded the widest scope consistent 10 with the principles and novel features disclosed herein.
I CLAIM:
Claims (9)
1. In a linear prediction coder to provide synthesized speech in which short term and long term redundancies by a filter means having L
taps wherein said filter means has an impulse response, h(n), are removed from a frame of N digitized speech samples resulting in a residual waveform of N samples, a method for encoding said residual waveform using k codebook vector, c k, comprising:
convolving a target signal, x(n), and said impulse response, h(n) to provide a first convolution;
autocorrelating an impulse response matrix wherein said impulse response matrix is a lower triangular toeplitz matrix with diagonal h(0) where h(0) is the zeroth impulse response value and the lower diagonals h(1),...,h(L-1) and wherein said impulse response autcorrelation is computed in accordance with the equation:
;
autocorrelating said synthesized speech in accordance with said autocorrelation of said impulse response matrix and said codebook vectors, C k to provide a synthesized speech autocorrelation, Eyy;
cross correlating said synthesized speech and said target speech in accordance with said first convolution and said codebook vectors to provide a cross correlation Exy; and selecting a codebook vector in accordance with said cross correlation, Exy, and said synthesized speech autocorrelation, Eyy.
taps wherein said filter means has an impulse response, h(n), are removed from a frame of N digitized speech samples resulting in a residual waveform of N samples, a method for encoding said residual waveform using k codebook vector, c k, comprising:
convolving a target signal, x(n), and said impulse response, h(n) to provide a first convolution;
autocorrelating an impulse response matrix wherein said impulse response matrix is a lower triangular toeplitz matrix with diagonal h(0) where h(0) is the zeroth impulse response value and the lower diagonals h(1),...,h(L-1) and wherein said impulse response autcorrelation is computed in accordance with the equation:
;
autocorrelating said synthesized speech in accordance with said autocorrelation of said impulse response matrix and said codebook vectors, C k to provide a synthesized speech autocorrelation, Eyy;
cross correlating said synthesized speech and said target speech in accordance with said first convolution and said codebook vectors to provide a cross correlation Exy; and selecting a codebook vector in accordance with said cross correlation, Exy, and said synthesized speech autocorrelation, Eyy.
2. The method of Claim 1 further comprising the steps of:
generating a first set of filter coefficients;
generating a second set of filter coefficients;
combining said first set of filter coefficients and said second set of filter coefficients to provide said impulse response, h(n).
generating a first set of filter coefficients;
generating a second set of filter coefficients;
combining said first set of filter coefficients and said second set of filter coefficients to provide said impulse response, h(n).
3. The method of Claim 1 further comprising:
receiving said input frame of N digitized samples; and perceptual weighting said input frame to provide said target signal.
receiving said input frame of N digitized samples; and perceptual weighting said input frame to provide said target signal.
4. The method of claim 1 wherein said step of convolving said target signal and said impulse response is performed in accordance with the equation:
.
.
5. The method of Claim 1 further comprising the step of storing said impulse response autcorrelation in a memory of L memory locations.
6. The method of Claim 1 wherein said step of cross correlating said synthesized speech and said target speech is performed in accordance with the equation:
, where d(k) is the cross correlation of the target signal and the impulse response.
, where d(k) is the cross correlation of the target signal and the impulse response.
7. The method of Claim 1 wherein step of autocorrelating said synthesized speech is performed in accordance with the equation:
.
.
8. The method of Claim 1 wherein said step of selecting a codebook vector comprises the steps of:
for each code vector, c k, squaring the value Exy;
dividing computed value of Eyy by said square of Exy for each code vector, c k; and selecting the code vector which maximizes the quotient of Eyy and the square of Exy.
for each code vector, c k, squaring the value Exy;
dividing computed value of Eyy by said square of Exy for each code vector, c k; and selecting the code vector which maximizes the quotient of Eyy and the square of Exy.
9. The method of Claim 1 wherein said codebook vectors, c k, are selected in accordance with an algebraic codebook format.
Applications Claiming Priority (3)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| US08/690,709 US5751901A (en) | 1996-07-31 | 1996-07-31 | Method for searching an excitation codebook in a code excited linear prediction (CELP) coder |
| US08/690,709 | 1996-07-31 | ||
| PCT/US1997/013594 WO1998005030A1 (en) | 1996-07-31 | 1997-07-31 | Method and apparatus for searching an excitation codebook in a code excited linear prediction (clep) coder |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| CA2261956A1 true CA2261956A1 (en) | 1998-02-05 |
Family
ID=24773618
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CA002261956A Abandoned CA2261956A1 (en) | 1996-07-31 | 1997-07-31 | Method and apparatus for searching an excitation codebook in a code excited linear prediction (clep) coder |
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| Country | Link |
|---|---|
| US (1) | US5751901A (en) |
| EP (1) | EP0917710B1 (en) |
| JP (1) | JP2000515998A (en) |
| KR (1) | KR100497788B1 (en) |
| CN (1) | CN1124589C (en) |
| AT (1) | ATE259532T1 (en) |
| AU (1) | AU719568B2 (en) |
| BR (1) | BR9710640A (en) |
| CA (1) | CA2261956A1 (en) |
| DE (1) | DE69727578D1 (en) |
| FI (1) | FI990181A (en) |
| IL (1) | IL128285A0 (en) |
| WO (1) | WO1998005030A1 (en) |
Families Citing this family (20)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| JPH11513813A (en) * | 1995-10-20 | 1999-11-24 | アメリカ オンライン インコーポレイテッド | Repetitive sound compression system |
| US6714907B2 (en) * | 1998-08-24 | 2004-03-30 | Mindspeed Technologies, Inc. | Codebook structure and search for speech coding |
| KR100576024B1 (en) * | 2000-04-12 | 2006-05-02 | 삼성전자주식회사 | Codebook searching apparatus and method in a speech compressor having an acelp structure |
| US7363219B2 (en) * | 2000-09-22 | 2008-04-22 | Texas Instruments Incorporated | Hybrid speech coding and system |
| US7606703B2 (en) * | 2000-11-15 | 2009-10-20 | Texas Instruments Incorporated | Layered celp system and method with varying perceptual filter or short-term postfilter strengths |
| US6766289B2 (en) * | 2001-06-04 | 2004-07-20 | Qualcomm Incorporated | Fast code-vector searching |
| US6789059B2 (en) * | 2001-06-06 | 2004-09-07 | Qualcomm Incorporated | Reducing memory requirements of a codebook vector search |
| DE10140507A1 (en) * | 2001-08-17 | 2003-02-27 | Philips Corp Intellectual Pty | Method for the algebraic codebook search of a speech signal coder |
| US7054807B2 (en) * | 2002-11-08 | 2006-05-30 | Motorola, Inc. | Optimizing encoder for efficiently determining analysis-by-synthesis codebook-related parameters |
| US7047188B2 (en) * | 2002-11-08 | 2006-05-16 | Motorola, Inc. | Method and apparatus for improvement coding of the subframe gain in a speech coding system |
| JP3887598B2 (en) * | 2002-11-14 | 2007-02-28 | 松下電器産業株式会社 | Coding method and decoding method for sound source of probabilistic codebook |
| US7249014B2 (en) * | 2003-03-13 | 2007-07-24 | Intel Corporation | Apparatus, methods and articles incorporating a fast algebraic codebook search technique |
| GB0307752D0 (en) * | 2003-04-03 | 2003-05-07 | Seiko Epson Corp | Apparatus for algebraic codebook search |
| CN101615396B (en) * | 2003-04-30 | 2012-05-09 | 松下电器产业株式会社 | Voice encoding device and voice decoding device |
| KR100668300B1 (en) * | 2003-07-09 | 2007-01-12 | 삼성전자주식회사 | Bitrate scalable speech coding and decoding apparatus and method thereof |
| CN1886781B (en) * | 2003-12-02 | 2011-05-04 | 汤姆森许可贸易公司 | Method for coding and decoding impulse responses of audio signals |
| JP3981399B1 (en) * | 2006-03-10 | 2007-09-26 | 松下電器産業株式会社 | Fixed codebook search apparatus and fixed codebook search method |
| US8920343B2 (en) | 2006-03-23 | 2014-12-30 | Michael Edward Sabatino | Apparatus for acquiring and processing of physiological auditory signals |
| WO2009033288A1 (en) | 2007-09-11 | 2009-03-19 | Voiceage Corporation | Method and device for fast algebraic codebook search in speech and audio coding |
| CA2979948C (en) | 2012-10-05 | 2019-10-22 | Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V. | An apparatus for encoding a speech signal employing acelp in the autocorrelation domain |
Family Cites Families (72)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US3633107A (en) * | 1970-06-04 | 1972-01-04 | Bell Telephone Labor Inc | Adaptive signal processor for diversity radio receivers |
| JPS5017711A (en) * | 1973-06-15 | 1975-02-25 | ||
| US4076958A (en) * | 1976-09-13 | 1978-02-28 | E-Systems, Inc. | Signal synthesizer spectrum contour scaler |
| US4214125A (en) * | 1977-01-21 | 1980-07-22 | Forrest S. Mozer | Method and apparatus for speech synthesizing |
| CA1123955A (en) * | 1978-03-30 | 1982-05-18 | Tetsu Taguchi | Speech analysis and synthesis apparatus |
| DE3023375C1 (en) * | 1980-06-23 | 1987-12-03 | Siemens Ag, 1000 Berlin Und 8000 Muenchen, De | |
| US4379949A (en) * | 1981-08-10 | 1983-04-12 | Motorola, Inc. | Method of and means for variable-rate coding of LPC parameters |
| USRE32580E (en) * | 1981-12-01 | 1988-01-19 | American Telephone And Telegraph Company, At&T Bell Laboratories | Digital speech coder |
| JPS6011360B2 (en) * | 1981-12-15 | 1985-03-25 | ケイディディ株式会社 | Audio encoding method |
| US4535472A (en) * | 1982-11-05 | 1985-08-13 | At&T Bell Laboratories | Adaptive bit allocator |
| EP0111612B1 (en) * | 1982-11-26 | 1987-06-24 | International Business Machines Corporation | Speech signal coding method and apparatus |
| US4667340A (en) * | 1983-04-13 | 1987-05-19 | Texas Instruments Incorporated | Voice messaging system with pitch-congruent baseband coding |
| US4787925A (en) * | 1983-04-15 | 1988-11-29 | Figgie International Inc. | Gas filter canister housing assembly |
| EP0127718B1 (en) * | 1983-06-07 | 1987-03-18 | International Business Machines Corporation | Process for activity detection in a voice transmission system |
| US4672670A (en) * | 1983-07-26 | 1987-06-09 | Advanced Micro Devices, Inc. | Apparatus and methods for coding, decoding, analyzing and synthesizing a signal |
| EP0163829B1 (en) * | 1984-03-21 | 1989-08-23 | Nippon Telegraph And Telephone Corporation | Speech signal processing system |
| DE3411844A1 (en) * | 1984-03-30 | 1985-10-10 | Robert Bosch Gmbh, 7000 Stuttgart | IGNITION COIL FOR THE MULTI-PLUGED AND DISTRIBUTORLESS IGNITION SYSTEM OF AN INTERNAL COMBUSTION ENGINE |
| US4617676A (en) * | 1984-09-04 | 1986-10-14 | At&T Bell Laboratories | Predictive communication system filtering arrangement |
| US4885790A (en) * | 1985-03-18 | 1989-12-05 | Massachusetts Institute Of Technology | Processing of acoustic waveforms |
| US4856068A (en) * | 1985-03-18 | 1989-08-08 | Massachusetts Institute Of Technology | Audio pre-processing methods and apparatus |
| US4937873A (en) * | 1985-03-18 | 1990-06-26 | Massachusetts Institute Of Technology | Computationally efficient sine wave synthesis for acoustic waveform processing |
| US4831636A (en) * | 1985-06-28 | 1989-05-16 | Fujitsu Limited | Coding transmission equipment for carrying out coding with adaptive quantization |
| JPS628031A (en) * | 1985-07-04 | 1987-01-16 | Matsushita Electric Ind Co Ltd | Hydraulic oil sensor |
| US4827517A (en) * | 1985-12-26 | 1989-05-02 | American Telephone And Telegraph Company, At&T Bell Laboratories | Digital speech processor using arbitrary excitation coding |
| CA1299750C (en) * | 1986-01-03 | 1992-04-28 | Ira Alan Gerson | Optimal method of data reduction in a speech recognition system |
| US4797929A (en) * | 1986-01-03 | 1989-01-10 | Motorola, Inc. | Word recognition in a speech recognition system using data reduced word templates |
| US4726037A (en) * | 1986-03-26 | 1988-02-16 | American Telephone And Telegraph Company, At&T Bell Laboratories | Predictive communication system filtering arrangement |
| JPH0748695B2 (en) * | 1986-05-23 | 1995-05-24 | 株式会社日立製作所 | Speech coding system |
| US4899384A (en) * | 1986-08-25 | 1990-02-06 | Ibm Corporation | Table controlled dynamic bit allocation in a variable rate sub-band speech coder |
| US4697261A (en) * | 1986-09-05 | 1987-09-29 | M/A-Com Government Systems, Inc. | Linear predictive echo canceller integrated with RELP vocoder |
| US4771465A (en) * | 1986-09-11 | 1988-09-13 | American Telephone And Telegraph Company, At&T Bell Laboratories | Digital speech sinusoidal vocoder with transmission of only subset of harmonics |
| US4903301A (en) * | 1987-02-27 | 1990-02-20 | Hitachi, Ltd. | Method and system for transmitting variable rate speech signal |
| US5054072A (en) * | 1987-04-02 | 1991-10-01 | Massachusetts Institute Of Technology | Coding of acoustic waveforms |
| US4868867A (en) * | 1987-04-06 | 1989-09-19 | Voicecraft Inc. | Vector excitation speech or audio coder for transmission or storage |
| US5202953A (en) * | 1987-04-08 | 1993-04-13 | Nec Corporation | Multi-pulse type coding system with correlation calculation by backward-filtering operation for multi-pulse searching |
| US4890327A (en) * | 1987-06-03 | 1989-12-26 | Itt Corporation | Multi-rate digital voice coder apparatus |
| US4899385A (en) * | 1987-06-26 | 1990-02-06 | American Telephone And Telegraph Company | Code excited linear predictive vocoder |
| CA1337217C (en) * | 1987-08-28 | 1995-10-03 | Daniel Kenneth Freeman | Speech coding |
| US4852179A (en) * | 1987-10-05 | 1989-07-25 | Motorola, Inc. | Variable frame rate, fixed bit rate vocoding method |
| IL84902A (en) * | 1987-12-21 | 1991-12-15 | D S P Group Israel Ltd | Digital autocorrelation system for detecting speech in noisy audio signal |
| US4817157A (en) * | 1988-01-07 | 1989-03-28 | Motorola, Inc. | Digital speech coder having improved vector excitation source |
| US4896361A (en) * | 1988-01-07 | 1990-01-23 | Motorola, Inc. | Digital speech coder having improved vector excitation source |
| DE3871369D1 (en) * | 1988-03-08 | 1992-06-25 | Ibm | METHOD AND DEVICE FOR SPEECH ENCODING WITH LOW DATA RATE. |
| DE3883519T2 (en) * | 1988-03-08 | 1994-03-17 | Ibm | Method and device for speech coding with multiple data rates. |
| US5023910A (en) * | 1988-04-08 | 1991-06-11 | At&T Bell Laboratories | Vector quantization in a harmonic speech coding arrangement |
| US4864561A (en) * | 1988-06-20 | 1989-09-05 | American Telephone And Telegraph Company | Technique for improved subjective performance in a communication system using attenuated noise-fill |
| FR2636163B1 (en) * | 1988-09-02 | 1991-07-05 | Hamon Christian | METHOD AND DEVICE FOR SYNTHESIZING SPEECH BY ADDING-COVERING WAVEFORMS |
| JPH0783315B2 (en) * | 1988-09-26 | 1995-09-06 | 富士通株式会社 | Variable rate audio signal coding system |
| US5077798A (en) * | 1988-09-28 | 1991-12-31 | Hitachi, Ltd. | Method and system for voice coding based on vector quantization |
| EP0364647B1 (en) * | 1988-10-19 | 1995-02-22 | International Business Machines Corporation | Improvement to vector quantizing coder |
| NL8901032A (en) * | 1988-11-10 | 1990-06-01 | Philips Nv | CODER FOR INCLUDING ADDITIONAL INFORMATION IN A DIGITAL AUDIO SIGNAL WITH A PREFERRED FORMAT, A DECODER FOR DERIVING THIS ADDITIONAL INFORMATION FROM THIS DIGITAL SIGNAL, AN APPARATUS FOR RECORDING A DIGITAL SIGNAL ON A CODE OF RECORD. OBTAINED A RECORD CARRIER WITH THIS DEVICE. |
| JP3033060B2 (en) * | 1988-12-22 | 2000-04-17 | 国際電信電話株式会社 | Voice prediction encoding / decoding method |
| US5357594A (en) * | 1989-01-27 | 1994-10-18 | Dolby Laboratories Licensing Corporation | Encoding and decoding using specially designed pairs of analysis and synthesis windows |
| US5222189A (en) * | 1989-01-27 | 1993-06-22 | Dolby Laboratories Licensing Corporation | Low time-delay transform coder, decoder, and encoder/decoder for high-quality audio |
| DE68916944T2 (en) * | 1989-04-11 | 1995-03-16 | Ibm | Procedure for the rapid determination of the basic frequency in speech coders with long-term prediction. |
| US5060269A (en) * | 1989-05-18 | 1991-10-22 | General Electric Company | Hybrid switched multi-pulse/stochastic speech coding technique |
| GB2235354A (en) * | 1989-08-16 | 1991-02-27 | Philips Electronic Associated | Speech coding/encoding using celp |
| US5091945A (en) * | 1989-09-28 | 1992-02-25 | At&T Bell Laboratories | Source dependent channel coding with error protection |
| DE59002768D1 (en) * | 1989-10-06 | 1993-10-21 | Telefunken Fernseh & Rundfunk | METHOD FOR TRANSMITTING A SIGNAL. |
| DE59002222D1 (en) * | 1989-10-06 | 1993-09-09 | Telefunken Fernseh & Rundfunk | METHOD FOR TRANSMITTING A SIGNAL. |
| JPH03181232A (en) * | 1989-12-11 | 1991-08-07 | Toshiba Corp | Variable rate encoding system |
| CA2010830C (en) * | 1990-02-23 | 1996-06-25 | Jean-Pierre Adoul | Dynamic codebook for efficient speech coding based on algebraic codes |
| US5103459B1 (en) * | 1990-06-25 | 1999-07-06 | Qualcomm Inc | System and method for generating signal waveforms in a cdma cellular telephone system |
| US5235671A (en) * | 1990-10-15 | 1993-08-10 | Gte Laboratories Incorporated | Dynamic bit allocation subband excited transform coding method and apparatus |
| US5187745A (en) * | 1991-06-27 | 1993-02-16 | Motorola, Inc. | Efficient codebook search for CELP vocoders |
| US5175769A (en) * | 1991-07-23 | 1992-12-29 | Rolm Systems | Method for time-scale modification of signals |
| US5734789A (en) * | 1992-06-01 | 1998-03-31 | Hughes Electronics | Voiced, unvoiced or noise modes in a CELP vocoder |
| IT1257065B (en) * | 1992-07-31 | 1996-01-05 | Sip | LOW DELAY CODER FOR AUDIO SIGNALS, USING SYNTHESIS ANALYSIS TECHNIQUES. |
| DE69428612T2 (en) * | 1993-01-25 | 2002-07-11 | Matsushita Electric Ind Co Ltd | Method and device for carrying out a time scale modification of speech signals |
| US5526464A (en) * | 1993-04-29 | 1996-06-11 | Northern Telecom Limited | Reducing search complexity for code-excited linear prediction (CELP) coding |
| JP3137805B2 (en) * | 1993-05-21 | 2001-02-26 | 三菱電機株式会社 | Audio encoding device, audio decoding device, audio post-processing device, and methods thereof |
| JP3182032B2 (en) * | 1993-12-10 | 2001-07-03 | 株式会社日立国際電気 | Voice coded communication system and apparatus therefor |
-
1996
- 1996-07-31 US US08/690,709 patent/US5751901A/en not_active Expired - Lifetime
-
1997
- 1997-07-31 CN CN97197717A patent/CN1124589C/en not_active Expired - Lifetime
- 1997-07-31 BR BR9710640-2A patent/BR9710640A/en not_active IP Right Cessation
- 1997-07-31 WO PCT/US1997/013594 patent/WO1998005030A1/en active IP Right Grant
- 1997-07-31 JP JP10509169A patent/JP2000515998A/en active Pending
- 1997-07-31 CA CA002261956A patent/CA2261956A1/en not_active Abandoned
- 1997-07-31 KR KR10-1999-7000852A patent/KR100497788B1/en active IP Right Grant
- 1997-07-31 AT AT97937095T patent/ATE259532T1/en not_active IP Right Cessation
- 1997-07-31 IL IL12828597A patent/IL128285A0/en unknown
- 1997-07-31 AU AU39694/97A patent/AU719568B2/en not_active Expired
- 1997-07-31 DE DE69727578T patent/DE69727578D1/en not_active Expired - Lifetime
- 1997-07-31 EP EP97937095A patent/EP0917710B1/en not_active Expired - Lifetime
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1999
- 1999-02-01 FI FI990181A patent/FI990181A/en not_active Application Discontinuation
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| ATE259532T1 (en) | 2004-02-15 |
| KR20000029745A (en) | 2000-05-25 |
| AU3969497A (en) | 1998-02-20 |
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| CN1229502A (en) | 1999-09-22 |
| CN1124589C (en) | 2003-10-15 |
| US5751901A (en) | 1998-05-12 |
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| EP0917710B1 (en) | 2004-02-11 |
| AU719568B2 (en) | 2000-05-11 |
| DE69727578D1 (en) | 2004-03-18 |
| WO1998005030A1 (en) | 1998-02-05 |
| JP2000515998A (en) | 2000-11-28 |
| BR9710640A (en) | 2002-08-06 |
| IL128285A0 (en) | 2000-02-17 |
| KR100497788B1 (en) | 2005-06-29 |
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