Vector excitation speech or audio coder for transmission or storage

Claim 2

An improvement in the method for compressing digitally encoded speech or audio signal by using a permanent indexed codebook of N predetermined excitation vectors of dimension k, each having an assigned codebook index j to find indices which identify the best match between an input speech vector s n that is to be coded and a vector c j from a codebook, where the subscript j is an index which uniquely identifies a codevector in said codebook, and the index of which is to be associated with the vector code, comprising the steps of buffering and grouping said vectors into frames of L samples, with L/k vectors for each frame, performing initial analyses for each successive frame to determine a set of parameters for specifying long-term synthesis filtering, short-term synthesis filtering, and perceptual weighting, computing a zero-input response of a long-term synthesis filter, short-term synthesis filter, and perceptual weighting filter, perceptually weighting each input vector s n of a frame and subtracting from each input vector s n said zero input response to produce a vector z n , obtaining each codevector c j from said codebook one at a time and processing each codevector c j through a scaling unit, said unit being controlled by a gain factor G j , and further processing each scaled codevector c j through a long-term synthesis filter, short-term synthesis filter and perceptual weighting filter in cascade, said cascaded filters being controlled by said set of parameters to produce a set of estimates z j of said vector z n , one estimate for each codevector c j , finding the estimate z j which best matches the vector z n , computing a quantized value of said gain factor G j using said vector z n and the estimate z j which best matches z n , pairing together the index j of the estimate z j which best matches z n and said quantized value of said gain factor G j as index-gain pairs for later reconstruction of said digitally encoded speech or audio signal, associating with each frame said index-gain pairs from said frame along with the quantized values of said parameters otained by initial analysis for use in specifying long-term synthesis filtering and short-term synthesis filtering in said reconstruction of said digitally encoded speech or audio signal, and during said reconstruction, reading out of a codebook a codevector c j that is identical to the codevector c j used for finding said best estimate by processing said reconstruction codevector c j through said scalar and said cascaded long-term and short-term synthesis filters.

Claim 3

An improvement in the method for compressing digitally encoded speech as defined in claim 1 wherein said codebooks are made sparse by extracting vectors from an initial arbitrary codebook, one at a time, and setting all but a selected number of samples of highest amplitude values in each vector to zero amplitude values, thereby generating a sparse vector with the same number of samples as the initial vector, but with only said selected number of samples having nonzero values.

Claim 4

An improvement in the method for compressing digitally encoded speech as defined in claim 1 by use of a codebook to store vectors c j , where the subscript j is an index for each vector stored, a method for designing an optimum codebook using an initial arbitrary codebook and a set of m speech training vectors sn by producing for each vector s n in sequence said perceptually weighted vector z n , clustering said m vectors z n , calculating N centroid vectors from said m clustered vectors, where N<m, update said codebook by replacing N vectors c j with vector s n used to produce vector z n found to be a best match with said vector z j at index location j, and testing for convergence between the updated codebook and said set of m speech training vectors s n , and if convergence has not been achieved, repeating the process using the updated codebook until convergence is achieved.

Claim 5

An improvement as defined in claim 3, including a final step of center clipping vectors in the last updated codebook vector by setting to zero all but a selected number of samples of lowest amplitude in each vector c j , and leaving in each vector c j only said selected number of samples of highest amplitude by extracting the vectors of said last updated codebook, one at a time, and setting all but a selected number of samples of highest amplitude values in each vector to amplitude values of zero, thereby generating a sparse vector with the same number of samples as the last updated vector, but with only said selected number of samples having nonzero values.

Claim 6

An improvement as defined in claim 1 comprising a two-step fast search method wherein the first step is to classify a current speech frame prior to compressing by selecting one of a plurality of classes to which the current speech frame belongs, and the seocnd step is to use a selected one of a plurality of reduced sets of codevectors to find the best match been each input vector z i and one of the codevectors of said selected reduced set of codevectors having a unique correspondence between every codevector in the set and particular vectors in said permanent indexed codebook, whereby a reduced exhaustive search is achieved for processing each input vector z i of a frame by first classifying the frame and then using a reduced codevector set selected from the permanent index codebook for every input vector of the frame.

Claim 7

An improvement as defined in claim 5 wherein classification of each frame is carried out by examining the spectral envelope parameters of the current frame and comparing said spectral envelope parameters with stored vector parameters for all classes in order to select one of said plurality of reduced sets of codevectors.

Claim 8

An improvement as defined in claim 1, wherein the step of computing said quantized value of said gain factor G j and the estimate that best matches z n is carried out by calculating the cross-correlation between the estimate z j and said vector z n , and dividing the cross-correlation product of said vector z n and said estima z j in accordance with the following equation: ##EQU11## where k is the number of samples in a vector.

Claim 9

An improvement in the method for compressing digitally encoded speech or audio signal by using a permanent indexed codebook of N predetermined excitation vectors of dimension k, each having an assigned codebook index j to find indices which identify the best match between an input speech vector s n that is to be coded and a vector c j from a codebook, where the subscript j is an index which uniquely identifies a codevector in said codebook, and the index of which is to be associated with the vector code, comprising the steps of designing said codebook to have sparse vectors by extracting vectors from an initial arbitrary codebook, one at a time, and setting to zero value all but a selected number of samples of highest amplitude values in each vector, thereby generating a sparse vector with the same number of samples as the initial vector, but with only said selected number of samples having nonzero values, buffering and grouping said vectors into frames of L samples, with L/k vectors for each frame, performing initial analyzes for each successive frame to determine a set of parameters for specifying long-term synthesis filtering, short-term synthesis filtering, and perceptual weighting, computing a zero-input response of a long-term synthesis filter, short-term synthesis fiIter, and perceptual weighting filter, perceptually weighting each input vector s n of a frame and subtracting from each input vector s n said zero input response to produce a vector z n , obtaining each codevector c j from said codebook one at a time and processing each codevector c j through a scaling unit, said unit being controlled by a gain factor G j , and further processing each scaled codevector c j through a long-term synthesis filter, short-term synthesis filter, said cascaded filters being controlled by said set of parameters to produce a set of estimates z j of said vector z n , one estimate for each codevector c j , finding the estimate z j which best matches the vector z n , computing a quantized value of said gain factor G j using said vector z n and the estimate z j which best matches z n pairing together the index j of the estimate z j which best matches z n and said quantized value of said gain factor G j for later reconstruction of said digitally encoded speech or audio signal, associating with each frame said index-gain pairs from said frame along with the quantized values of said parameters obtained by initial analysis for use in specifying long-term synthesis filtering and short-term synthesis filtering in said reconstruction of said digitally encoded speech or audio signal, and during said reconstruction, reading out of a codebook a codevector c j that is identical codevector c j used for finding said best estimate by processing said reconstruction codevector c j through said scalar and said cascaded long-term and short-term synthesis filters.

Claim 10

An improvement in the method for compressing digitally encoded speech or audio signal by using a permanent indexed codebook of N predetermined excitation vectors of dimension k, each having an assigned codebook index j to find indices which identify the best match between an input speech vector s n that is to be coded and a vector c j from a codebook, where the subscript j is an index which uniquely identifies a codevector in said codebook, and the index of which is to be associated with the vector code, comprising the steps of designing said codebook to have sparse vectors by extracting vectors from an initial arbitrary codebook, one at a time, and setting to zero value all but a selected number of samples of highest amplitude values in each vector, thereby generating a sparse vector with the same number of samples as the initial vector, but with only said selected number of samples having nonzero values, buffering and grouping said vectors into frames of L samples, with L/k vectors for each frame, performing initial analyzes for each successive frame to determine a set of parameters for specifying long-term synthesis filtering, short-term synthesis filtering, and perceptual weighting, computing a zero-input response of a long-term synthesis filter, short-term synthesis fiIter, and perceptual weighting filter, perceptually weighting each input vector s n of a frame and subtracting from each input vector s n said zero input response to produce a vector z n , obtaining each codevector c j from said codebook one at a time and processing each codevector c j through a scaling unit, said unit being controlled by a gain factor G j , and further processing each scaled codevector c j through a long-term synthesis filter, short-term synthesis filter, said cascaded filters being controlled by said set of parameters to produce a set of estimates z j of said vector z n , one estimate for each codevector c j , finding the estimate z j which best matches the vector z n , computing a quantized value of said gain factor G j using said vector z n and the estimate z j which best matches z n pairing together the index j of the estimate z j which best matches z n and said quantized value of said gain factor G j for later reconstruction of said digitally encoded speech or audio signal, associating with each frame said index-gain pairs from said frame along with the quantized values of said parameters obtained by initial analysis for use in specifying long-term synthesis filtering and short-term synthesis filtering in said reconstruction of said digitally encoded speech or audio signal, and during said reconstruction, reading out of a codebook a codevector c j that is identical codevector c j used for finding said best estimate by processing said reconstruction codevector c j through said scalar and said cascaded long-term and short-term synthesis filters.

Claim 11

An improvement in the method for compressing digitally encoded speech as defined in claim 8 by use of a codebook to store vectors c j , where the subscript j is an index for each vector stored, a method for designing an optimum codebook using an initial arbitrary codebook and a set of m speech training vectors s n by producing for each vector s n in sequence said perceptually weighted vector z n , clustering said m vectors z n , calculating N centroid vectors from said m clustered vectors, where N<m, update said codebook by replacing N vectors c j with vector s n used to produce vector z n found to be a best match with said vector z j at index location j, and testing for convergence between the updated codebook and said set of m speech training vectors s n , and if convergence has not been achieved, repeating the process using the updated codebook until convergence is achieved.

Claim 12

An improvement as defined in claim 9, including a final s of extracting the last updated vectors, one at a time, and setting to zero value all but a selected number of samples of highest amplitude values in each vector, thereby generating a sparse vector with the same number of samples as the last updated vetor, but with only said selected number of samples with nonzero values.

Claim 13

An improvement as defined in claim 8 comprising a fast search method using said codebook to select a number N c of good excitation vectors c j , where N c is much smaller than N, and using said vectors N c for an exhaustive search to find the best match between said vector z n and estimate vector z j produced from a codevector c j included in said N c codebook vectors by precomputing N vectors z j , comparing an input vector z n with vectors z j , and producing a codebook of N c codevectors for use in an exhaustive search of the best match between said input vector z n and a vector z j from a codebook of N c vectors.

Claim 14

An improvement as defined in claim 11 wherein said N c codebook is produced by making rough classification of the gain-normalized spectral shape of a current speech frame into one of M s spectral shape classes, and selecting one of M s shaped codebooks for encoding an input vector z n by comparing said input vector with the z j vectors stored in the selected one of the M s shaped codebooks, and then taking the N c condevectors which produce the N c smallest errors for use in said N c codebook.