# Patent

## Title
Method and apparatus for receiving an encoded speech signal based on codebooks

## Abstract

A method and apparatus for processing speech in a wireless communication system uses CELP speech encoded signals. A decoder receives encoded speech including a code index, a code index gain, a pitch lag, a pitch gain, and a line spectral pair (LSP) index. An innovation codevector and an adaptive codevector are determined and scaled. An excitation sequence is generated. Reconstructed speech is then output based on the excitation sequence and LSP index.

## Background
CROSS REFERENCE TO RELATED APPLICATIONS
This application is a continuation of U.S. patent application Ser. No. 12/259,857 filed on Oct. 28, 2008 which is a continuation of Ser. No. 11/490,286 filed Jul. 20, 2006, which is a continuation of U.S. patent application Ser. No. 10/852,047 filed May 24, 2004, issued on Aug. 1, 2006 as U.S. Pat. No. 7,085,714, which is a continuation of U.S. patent application Ser. No. 10/082,412, filed Feb. 25, 2002, issued on Jul. 13, 2004 as U.S. Pat. No. 6,763,330, which is a continuation of U.S. patent application Ser. No. 09/711,252, filed Nov. 13, 2000, issued on May 14, 2002 as U.S. Pat. No. 6,389,388, which is a continuation of U.S. patent application Ser. No. 08/734,356, filed Oct. 21, 1996, issued on May 29, 2001 as U.S. Pat. No. 6,240,382, which is a continuation of U.S. patent application Ser. No. 08/166,223, filed Dec. 14, 1993, issued on Apr. 15, 1997 as U.S. Pat. No. 5,621,852, which are incorporated by reference as if fully set forth.


FIELD OF INVENTION
This invention relates to digital speech encoders using code excited linear prediction coding, or CELP. More particularly, this invention relates a method and apparatus for efficiently selecting a desired codevector used to reproduce an encoded speech segment at the decoder.
BACKGROUND
Direct quantization of analog speech signals is too inefficient for effective bandwidth utilization. A technique known as linear predictive coding, or LPC, which takes advantage of speech signal redundancies, requires much fewer bits to transmit or store speech signals. Originally speech signals are produced as a result of acoustical excitation of the vocal tract. While the vocal cords produce the acoustical excitation, the vocal tract (e.g. mouth, tongue and lips) acts as a time varying filter of the vocal excitation. Thus, speech signals can be efficiently represented as a quasi-periodic excitation signal plus the time varying parameters of a digital filter. In addition, the periodic nature of the vocal excitation can further be represented by a linear filter excited by a noise-like Gaussian sequence. Thus, in CELP, a first long delay predictor corresponds to the pitch periodicity of the human vocal cords, and a second short delay predictor corresponds to the filtering action of the human vocal tract.
CELP reproduces the individual speaker's voice by processing the input speech to determine the desired excitation sequence and time varying digital filter parameters. At the encoder, a prediction filter forms an estimate for the current sample of the input signal based on the past reconstructed values of the signal at the receiver decoder, i.e. the transmitter encoder predicts the value that the receiver decoder will reconstruct. The difference between the current value and predicted value of the input signal is the prediction error. For each frame of speech, the prediction residual and filter parameters are communicated to the receiver. The prediction residual or prediction error is also known as the innovation sequence and is used at the receiver as the excitation input to the prediction filters to reconstruct the speech signal. Each sample of the reconstructed speech signal is produced by adding the received signal to the predicted estimate of the present sample. For each successive speech frame, the innovation sequence and updated filter parameters are communicated to the receiver decoder.
The innovation sequence is typically encoded using codebook encoding. In codebook encoding, each possible innovation sequence is stored as an entry in a codebook and each is represented by an index. The transmitter and receiver both have the same codebook contents. To communicate given innovation sequence, the index for that innovation sequence in the transmitter codebook is transmitted to the receiver. At the receiver, the received index is used to look up the desired innovation sequence in the receiver codebook for use as the excitation sequence to the time varying digital filters.
The task of the CELP encoder is to generate the time varying filter coefficients and the innovation sequence in real time. The difficulty of rapidly selecting the best innovation sequence from a set of possible innovation sequences for each frame of speech is an impediment to commercial achievement of real time CELP based systems, such as cellular telephone, voice mail and the like.
Both random and deterministic codebooks are known. Random codebooks are used because the probability density function of the prediction error samples has been shown to be nearly white Gaussian random noise. However, random codebooks present a heavy computational burden to select an innovation sequence from the codebook at the encoder since the codebook must be exhaustively searched.
To select an innovation sequence from the codebook of stored innovation sequences, a given fidelity criterion is used. Each innovation sequence is filtered through time varying linear recursive filters to reconstruct (predict) the speech frame as it would be reconstructed at the receiver. The predicted speech frame using the candidate innovation sequence is compared with the desired target speech frame (filtered through a perceptual weighting filter) and the fidelity criterion is calculated. The process is repeated for each stored innovation sequence. The innovation sequence that maximizes the fidelity criterion function is selected as the optimum innovation sequence, and an index representing the selected optimum sequence is sent to the receiver, along with other filter parameters.
At the receiver, the index is used to access the selected innovation sequence, and, in conjunction with the other filter parameters, to reconstruct the desired speech.
The central problem is how to select an optimum innovation sequence from the codebook at the encoder within the constraints of real time speech encoding and acceptable transmission delay. In a random codebook, the innovation sequences are independently generated random white Gaussian sequences. The computational burden of performing an exhaustive search of all the innovation sequences in the random code book is extremely high because each innovation sequence must be passed through the prediction filters.
One prior art solution to the problem of selecting an innovation sequence is found in U.S. Pat. No. 4,797,925 in which the adjacent codebook entries have a subset of elements in common. In particular, each succeeding code sequence may be generated from the previous code sequence by removing one or more elements from the beginning of the previous sequence and adding one or more elements to the end of the previous sequence. The filter response to each succeeding code sequence is then generated from the filter response to the preceding code sequence by subtracting the filter response to the first samples and appending the filter response to the added samples. Such overlapping codebook structure permits accelerated calculation of the fidelity criterion.
Another prior art solution to the problem of rapidly selecting an optimum innovation sequence is found in U.S. Pat. No. 4,817,157 in which the codebook of excitation vectors is derived from a set of M basis vectors which are used to generate a set of 2M codebook excitation code vectors. The entire codebook of 2M possible excitation vectors is searched using the knowledge of how the code vectors are generated from. the basis vectors, without having to generate and evaluate each of the individual code vectors
SUMMARY
A receiver is used in decoding a received encoded signal. The received encoded speech signal is encoded using excitation linear prediction. The receiver receives the encoded speech signal. The encoded speech signal comprises a code, a pitch lag and a line spectral pair index. An innovation sequence is produced by selecting a code from each of a plurality of codebooks based on the code index. A line spectral pair quantization of a speech signal is determined using the line spectral pair index. A pitch lag is determined using the pitch lag index. A speech signal is reconstructed using the produced innovation sequence, the determined line spectral pair quantization and pitch lag.



BRIEF DESCRIPTION OF THE DRAWINGS
FIG. 1 is a diagram of a CELP encoder utilizing a ternary codebook in accordance with the present invention.
FIG. 2 is a block diagram of a CELP decoder utilizing a ternary codebook in accordance with the present invention.
FIG. 3 is a flow diagram of an exhaustive search process for finding an optimum codevector in accordance with the present invention.
FIG. 4 is a flow diagram of a first sub-optimum search process for finding a codevector in accordance with the present invention.
FIG. 5 is a flow diagram of a second sub-optimum search process for finding a codevector in accordance with the present invention.
FIGS. 6A, 6B and 6C are graphical representations of a first binary codevector, a second binary codevector, and a ternary codevector, respectively.



DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT(S)
CELP Encoding
The CELP encoder of FIG. 1 includes an input terminal 10 for receiving input speech samples which have been converted to digital form. The CELP encoder represents the input speech samples as digital parameters comprising an LSP index, a pitch lag and gain, and a code index and gain, for digital multiplexing by transmitter 30 on communication channel 31.
LSP Index
As indicated above, speech signals are produced as a result of acoustical excitation of the vocal tract. The input speech samples received on terminal 10 are processed in accordance with known techniques of LPC analysis 26, and are then quantized by a line spectral pair (LSP) quantization circuit 28 into a conventional LSP index.
Pitch Lag and Gain
Pitch lag and gain are derived from the input speech using a weighted synthesis filter 16, and an adaptive codebook analysis 18. The parameters of pitch lag and gain are made adaptive to the voice of the speaker, as is known in the art. The prediction error between the input speech samples at the output of the perceptual weighting filter 12, and predicted reconstructed speech samples from a weighted synthesis filter 16 is available at the output of adder 14. The perceptual weighting filter 12 attenuates those frequencies where the error is perceptually more important. The role of the weighting filter is to concentrate the coding noise in the formant regions where it is effectively masked by the speech signal. By doing so, the noise at other frequencies can be lowered to reduce the overall perceived noise. Weighted synthesis filter 16 represents the combined effect of the decoder synthesis filter and the perceptual weighting filter 12. Also, in order to set the proper initial conditions at the subframe boundary, a zero input is provided to weighted synthesis filter 16. The adaptive codebook analysis 18 performs predictive analysis by selecting a pitch lag and gain which minimizes the instantaneous energy of the mean squared prediction error.
Innovation Code Index and Gain
The innovation code index and gain is also made adaptive to the voice of the speaker using a second weighted synthesis filter 22, and a ternary codebook analysis 24, containing an encoder ternary codebook of the present invention. The prediction error between the input speech samples at the output of the adder 14, and predicted reconstructed speech samples from a second weighted synthesis filter 22 is available at the output of adder 20. Weighted synthesis filter 22 represents the combined effect of the decoder synthesis filter and the perceptual weighting filter 12, and also subtracts the effect of adaptive pitch lag and gain introduced by weighted synthesis filter 16 to the output of adder 14.
The ternary codebook analysis 18 performs predictive analysis by selecting an innovation sequence which maximizes a given fidelity criterion function. The ternary codebook structure is readily understood from a discussion of CELP decoding.
CELP Decoding
A CELP system decoder is shown in FIG. 2. A digital demultiplexer 32 is coupled to a communication channel 31. The received innovation code index (index i and index j), and associated gain is input to ternary decoder codebook 34. The ternary decoder codebook 34 is comprised of a first binary codebook 36, and a second binary codebook 38. The output of the first and second binary codebooks are added together in adder 40 to form a ternary codebook output, which is scaled by the received signed gain in multiplier 42. In general, any two digital codebooks may be added to form a third digital codebook by combining respective codevectors, such as a summation operation.
To illustrate how a ternary codevector is formed from two binary codevectors, reference is made to FIGS. 6A, 6B and 6C. A first binary codevector is shown in FIG. 6A consisting of values {0, 1}. A second binary codevector is shown in FIG. 6B consisting of values {−1, 0}. By signed addition in adder 40 of FIG. 2, the two binary codevectors form a ternary codevector, as illustrated in FIG. 6C.
The output of the ternary decoder codebook 34 in FIG. 2 is the desired innovation sequence or the excitation input to a CELP system. In particular, the innovation sequence from ternary decoder codebook 34 is combined in adder 44 with the output of the adaptive codebook 48 and applied to LPC synthesis filter 46. The result at the output of LPC synthesis filter 46 is the reconstructed speech. As a specific example, if each speech frame is 4 milliseconds, and the sampling rate is 8 Mhz, then each innovation sequence, or codevector, is 32 samples long.
Optimum Innovation Sequence Selection
The ternary codebook analysis 24 of FIG. 1 is illustrated in further detail by the process flow diagram of FIG. 3. In code excited linear prediction coding, the optimum codevector is found by maximizing the fidelity criterion function,




  
    
      
        
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where xt is the target vector representing the input speech sample, F is an N×N matrix with the term in the n th row and the i th column given by fn-i, and Ck is the k th codevector in the innovation codebook. Also, ∥λ2 indicates the sum of the squares of the vector components, and is essentially a measure of signal energy content. The truncated impulse response fn, n=1, 2 . . . N, represents the combined effects of the decoder synthesis filter and the perceptual weighting filter. The computational burden of the CELP encoder comes from the evaluation of the filtered term Fck and the cross-correlation, auto-correlation terms in the fidelity criterion function.

Let Ck=0i+ηj,

k=0, 1, . . . K−1

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Log2 K=Log2 I+Log2 J, where θiηj are codevectors from the two binary codebooks, the fidelity criterion function for the codebook search becomes,





  
    
      
        
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Search Procedures
There are several ways in which the fidelity criterion function Ψ(i,j) may be evaluated.
1. Exhaustive Search.
Finding the maximum Ψ(i,j) involves the calculation of Fθi, Fηj and θit Ft Fηj, which has I and J filtering and the IJ cross-correlation of xt Fθi, xt Fηj and ∥Fθi∥2,∥Fθj∥2, which has I+J cross-correlation and I+J auto-correlation terms.
FIG. 3 illustrates an exhaustive search process for the optimum innovation sequence. All combinations of binary codevectors in binary codebooks 1 and 2 are computed for the fidelity criterion function Ψ(i,j). The peak fidelity criterion function Ψ(i,j) is selected at step 62, thereby identifying the desired codebook index i and codebook index j.
Binary codebook 1 is selectively coupled to linear filter 50. The output of linear filter 50 is coupled to correlation step 52, which provides a correlation calculation with the target speech vector X, the input speech samples filtered in a perceptual weighting filter. Binary codebook 2 is selectively coupled to linear filter 68. The output of linear filter 68 is coupled to correlation step 72, which provides a correlation calculation with the target speech vector X. The output of correlation step 52 is coupled to one input of adder 66. The output of correlation step 72 is coupled to the other input of adder 66. The output of adder 66 is coupled to a square function 64 which squares the output of the adder 66 to form a value equal to the numerator of the fidelity criterion Ψ(i,j) of Equation 2. The linear filters 50 and 68 are each equivalent to the weighted synthesis filter 22 of FIG. 1, and are used only in the process of selecting optimum synthesis parameters. The decoder (FIG. 2) will use the normal synthesis filer.
The output of linear filter 50 is also coupled to a sum of the squares calculation step 54. The output of linear filter 68 is further coupled to a sum of the squares calculation step 70. The sum of the squares is a measure of signal energy content. The linear filter 50 and the linear filter 68 are also input to correlation step 56 to form a cross-correlation term between codebook 1 and codebook 2. The cross-correlation term output of correlation step 56 is multiplied by 2 in multiplier 58. Adder 60 combines the output of multiplier 58, the output of sum of the squares calculation step 54 plus the output of sum of the squares calculation step 70 to form a value equal to the denominator of the fidelity criterion Ψ(i,j) of Equation 2.
In operation, one of 16 codevectors of binary codebook 1 corresponding to a 4 bit codebook index i, and one of 16 codevectors of binary codebook 2 corresponding to a 4 bit codebook index j, is selected for evaluation in the fidelity criterion. The total number of searches is 16×16, or 256. However, the linear filtering steps 50, 68, the auto-correlation calculations 52, 72 and the sum of the squares calculation 54, 70 need only be performed 32 times (not 256 times), or once for each of 16 binary codevectors in two codebooks. The results of prior calculations are saved and reused, thereby reducing the time required to perform an exhaustive search. The number of cross-correlation calculations in correlation step 56 is equal to 256, the number of binary vector combinations searched.
The peak selection step 62 receives the numerator of Equation 2 on one input and the denominator of Equation 2 on the other input for each of the 256 searched combinations. Accordingly, the codebook index i and codebook index j corresponding to a peak of the fidelity criterion function Ψ(i,j) is identified. The ability to search the ternary codebook 34, which stores 256 ternary codevectors, by searching among only 32 binary codevectors, is based on the superposition property of linear filters.
2. Sub-Optimum Search I
FIG. 4 illustrates an alternative search process for the codebook index i and codebook index j corresponding to a desired codebook innovation sequence. This search involves the calculation of Equation 1 for codebook 1 and codebook 2 individually as follows:




  
    
      
        
          
            
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To search all the codevectors in both codebooks individually, only 16 searches are needed, and no cross-correlation terms exist. A subset of codevectors (say 5) in each of the two binary codebooks are selected as the most likely candidates. The two subsets that maximizes the fidelity criterion functions above are then jointly searched to determine the optimum, as in the exhaustive search in FIG. 3. Thus, for a subset of 5 codevectors in each codebook, only 25 joint searches are needed to exhaustively search all subset combinations.

In FIG. 4, binary codebook 1 is selectively coupled to linear filter 74. The output of linear filter 74 is coupled to a squared correlation step 76, which provides a squared correlation calculation with the target speech vector X. The output of linear filter 74 is also coupled to a sum of the squares calculation step 78. The output of the squared correlation step 76, and the sum of the squares calculation step 78 is input to peak selection step 80 to select a candidate subset of codebook 1 vectors.
Binary codebook 2 is selectively coupled to linear filter 84. The output of linear filter 84 is coupled to a squared correlation step 86, which provides a squared correlation calculation with the target speech vector X. The output of linear filter 84 is also coupled to a sum of the squares calculation step 88. The output of the squared correlation step 86, and the sum of the squares calculation step 88 is input to peak selection step 90 to select a candidate subset of codebook 2 vectors. In such manner a fidelity criterion function expressed by Equation 3 is carried out in the process of FIG. 4.
After the candidate subsets are determined, an exhaustive search as illustrated in FIG. 3 is performed using the candidate subsets as the input codevectors. In the present example, 25 searches are needed for an exhaustive search of the candidate subsets, as compared to 256 searches for the full binary codebooks. In addition, filtering and auto-correlation terms from the first calculation of the optimum binary codevector subsets are available for reuse in the subsequent exhaustive search of the candidate subsets.
3. Sub-Optimum Search II
FIG. 5 illustrates yet another alternative search process for the codebook index i and codebook index j corresponding to a desired codebook innovation sequence. This search evaluates each of the binary codevectors individually in both codebooks using the same fidelity criterion function as given in Equation 3 to find the one binary codevector having the maximum value of the fidelity criterion function. The maximum binary codevector, which may be found in either codebook (binary codebook 1 or binary codebook 2), is then exhaustively searched in combination with each binary codevector in the other binary codebook (binary codebook 2 or binary codebook 1), to maximize the fidelity criterion function Ψ(i j).
In FIG. 5, binary codebooks 1 and 2 are treated as a single set of binary codevectors, as schematically represented by a data bus 93 and selection switches 94 and 104.
That is, each binary codevector of binary codebook 1 and binary codebook 2 is selectively coupled to linear filter 96. The output of linear filter 96 is coupled to a squared correlation step 98, which provides a squared correlation calculation with the target speech vector X. The output of linear filter 96 is also coupled to a sum of the squares calculation step 100. The output of the squared correlation step 98, and the sum of the squares calculation step 100 is input to peak selection step 102 to select a single optimum codevector from codebook 1 and codebook 2. A total of 32 searches is required, and no cross-correlation terms are needed.
Having found the optimum binary codevector from codebook 1 and codebook 2, an exhaustive search for the optimum combination of binary codevectors 106 (as illustrated in FIG. 3) is performed using the single optimum codevector found as one set of the input codevectors. In addition, instead of exhaustively searching both codebooks, switch 104 under the control of the peak selection step 102, selects the codevectors from the binary codebook which does not contain the single optimum codevector found by peak selection step 102. In other words, if binary codebook 2 contains the optimum binary codevector, then switch 104 selects the set of binary codevectors from binary codebook 1 for the exhaustive search 106, and vice versa. In such manner, only 16 exhaustive searches need be performed. As before, filtering and auto-correlation terms from the first calculation of the optimum single optimum codevector from codebook 1 and codebook 2 are available for reuse in the subsequent exhaustive search step 106. The output of search step is the codebook index i and codebook index j representing the ternary innovation sequence for the current frame of speech.
Overlapping Codebook Structures
For any of the foregoing search strategies, the calculation of Fθi, Fηj can be further accelerated by using an overlapping codebook structure as indicated in cited U.S. Pat. No. 4,797,925 to the present inventor. That is, the codebook structure has adjacent codevectors which have a subset of elements in common. An example of such structure is the following two codevectors:

θLt=(gL,gL+1, . . . ,gL+N−1)

θL+1t=(gL+1,gL+2, . . . ,gL+N)

Other overlapping structures in which the starting positions of the codevectors are shifted by more than one sample are also possible. With the overlapping structure, the filtering operation of Fθi and Fηj can be accomplished by a procedure using recursive endpoint correction in which the filter response to each succeeding code sequence is then generated from the filter response to the preceding code sequence by subtracting the filter response to the first sample gL, and appending the filter response to the added sample gL+N. In such manner, except for the first codevector, the filter response to each successive codevector can be calculated using only one additional sample.

## Claims

1. A method of receiving an encoded speech signal for use in a decoding device, the method comprising:
receiving the encoded speech signal including a code index, a code index gain, a pitch lag, a pitch gain, and a line spectral pair (LSP) index;
determining an innovation codevector based on the code index and the code index gain, wherein the innovation codevector is generated from at least one of a plurality of codebooks;
determining an adaptive codevector based on the pitch lag and the pitch gain;
generating an excitation sequence based on the innovation codevector and the adaptive codevector; and
outputting a reconstructed speech signal based on the LSP index and the excitation sequence.

2. The method of claim 1, wherein the LSP index is determined by linear predictive coding (LPC) analysis and quantizing results of the LPC analysis.

3. The method of claim 1, wherein the pitch lag and the pitch gain are determined by adaptive codebook analysis on weighted speech samples.

4. The method of claim 1, wherein the code index and the code index gain are determined by codebook analysis on weighted speech samples.

5. The method of claim 1, wherein generating the excitation sequence comprises:
adding the innovation codevector associated with the code index to the adaptive codevector associated with the pitch lag; and
scaling the innovation codevector and the adaptive codevector based on the code index gain and the pitch gain.

6. A decoder for decoding an encoded speech signal, the decoder comprising:
a processor configured to:
receive the encoded speech signal including a code index, a code index gain, a pitch lag, a pitch gain, and a line spectral pair (LSP) index;
determine an innovation codevector based on the code index and the code index gain, wherein the innovation codevector is generated from at least one of a plurality of codebooks;
determine an adaptive codevector based on the pitch lag and the pitch gain; and
generate an excitation sequence based on the innovation codevector and the adaptive codevector; and

a filter configured to generate a reconstructed speech signal based on the LSP index and the excitation sequence.

7. The decoder of claim 6, wherein the filter is a linear predictive coding (LPC) synthesis filter.

8. The decoder of claim 6, wherein the LSP index is determined by linear predictive coding (LPC) analysis and quantizing results of the LPC analysis.

9. The decoder of claim 6, wherein the pitch lag and the pitch gain are determined by adaptive codebook analysis on weighted speech samples.

10. The decoder of claim 6, wherein the code index and the code index gain are determined by codebook analysis on weighted speech samples.

11. The decoder of claim 6, wherein the processor comprises an adder configured to add the innovation codevector and the adaptive codevector.

12. The decoder of claim 6, wherein the processor comprises a multiplier configured to scale the innovation codevector and the adaptive codevector based on the code index gain and the pitch gain.