Wear-toll quality 4.8 kbps speech codec

Claim 3

An apparatus as defined in claim 1, wherein said second means generates said second coded signal portion by generating a predicted value of said input speech signal and comparing said predicted value to said input speech signal, and wherein steps (3) and (4) are repeated until an amount of distortion between said predicted value and said input speech signal is minimized.

Claim 4

An apparatus as defined in claim 1, wherein said plurality of coded signal portions includes spectrum filter coefficients, and said iterative optimization means including means for first calculating an initial set of spectrum filter coefficients, then deriving said first and second optimized coded signal portions according to steps (1)-(5) in claim 1, and then deriving an optimized set of spectrum filter coefficients in accordance with at least said first and second optimized coded signal portions and said initial set of spectrum filter coefficients.

Claim 5

A speech analysis and synthesis method comprising the steps of deriving a set of predictor coefficients for each analysis time period from an original input signal having a plurality of successive analysis time periods, coding said predictor coefficients to obtain a coded representation of said coefficients, transmitting the coded representation of said predictor coefficients to a decoder and synthesizing the original input speech signal in accordance with said transmitted coded representation of said predictor coefficients, said coding step comprising: transforming said set of predictor coefficients for one analysis time period into parameters in a parameter st to form a parameter vector; subtracting from said parameter vector a mean vector determined in advance from a large speech data base to obtain an adjusted parameter vector; selecting from a codebook of 2 L entries (where L is an integer), prepared in advance from said large speech data base, a prediction matrix A such that F.sub.n =AF.sub.n-1 where n is an integer, F n is a predicted parameter vector for said one analysis time period and F n-1 is the adjusted parameter vector for an immediately preceding analysis time period; calculating a predicted parameter vector for said one analysis time period as well as a residual parameter vector comprising the difference between said predicted parameter vector and said adjusted parameter vector; quantizing said residual parameter vector in a first stage vector quantizer by selecting one of 2 M (where M is an integer) first quantization vectors to obtain an intermediate quantized vector; calculating a residual quantized vector comprising the difference between said intermediate quantized vector and said residual parameter vector; quantizing said residual quantized vector in a second stage vector quantizer by selecting one of 2 N (where N is an integer) second quantization vectors to obtain a final quantized vector; and forming said transmitted coded representation of said predictor coefficients by combining an L-bit value representing the prediction matrix A, an M-bit value representing said intermediate quantized vector and an N-bit value representing said final quantized vector.

Claim 6

A speech analysis and synthesis method comprising the steps of deriving a set of predictor coefficients for each analysis time period from an original input signal having a plurality of successive analysis time periods, coding said predictor coefficients to obtain a coded representation of said coefficients, transmitting the coded representation of said predictor coefficients to a decoder and synthesizing the original input speech signal in accordance with said transmitted coded representation of said predictor coefficients, said coding step comprising: transforming said set of predictor coefficients for one analysis time period into parameters in a parameter st to form a parameter vector; subtracting from said parameter vector a mean vector determined in advance from a large speech data base to obtain an adjusted parameter vector; selecting from a codebook of 2 L entries (where L is an integer), prepared in advance from said large speech data base, a prediction matrix A such that F.sub.n =AF.sub.n-1 where n is an integer, F n is a predicted parameter vector for said one analysis time period and F n-1 is the adjusted parameter vector for an immediately preceding analysis time period; calculating a predicted parameter vector for said one analysis time period as well as a residual parameter vector comprising the difference between said predicted parameter vector and said adjusted parameter vector; quantizing said residual parameter vector in a first stage vector quantizer by selecting one of 2 M (where M is an integer) first quantization vectors to obtain an intermediate quantized vector; calculating a residual quantized vector comprising the difference between said intermediate quantized vector and said residual parameter vector; quantizing said residual quantized vector in a second stage vector quantizer by selecting one of 2 N (where N is an integer) second quantization vectors to obtain a final quantized vector; and forming said transmitted coded representation of said predictor coefficients by combining an L-bit value representing the prediction matrix A, an M-bit value representing said intermediate quantized vector and an N-bit value representing said final quantized vector.

Claim 7

A speech analysis and synthesis method as defined in claim 4, wherein said parameters comprise line spectrum frequencies.

Claim 8

A speech analysis and synthesis method as defined in claim 4, wherein L=6, M=10 and N=10.

Claim 9

A speech analysis and synthesis method comprising the steps of deriving a set of predictor coefficients for each analysis time period from an original input signal having a plurality of successive analysis time periods, coding said predictor coefficients to obtain a coded representation of said coefficients, transmitting the coded representation of said predictor coefficients to a decoder and synthesizing the original input speech signal in accordance with said transmitted coded representation of said predictor coefficients, said coding step comprising: generating a multi-component input vector corresponding to said set of predictor coefficients for one analysis time period, with each component of said vector corresponding to a frequency; quantizing said input vector by selecting a plurality of multi-component quantization vectors from a quantization vector storage means and calculating for each selected quantization vector a distortion measure in accordance with the difference between each component of said input vector and each corresponding component of the selected quantization vector, and in accordance with a weighting factor associated with each component of said input vector, the weighting factor being determined for each component of said input vector in accordance with the frequency to which said component corresponds; selecting as a quantizer output the one of said plurality of selected quantization vectors resulting in the least distortion measure; and generating said transmitted coded representation in accordance with the selected quantizer output.

Claim 10

A speech analysis and synthesis method comprising the steps of deriving a set of predictor coefficients for each analysis time period from an original input signal having a plurality of successive analysis time periods, coding said predictor coefficients to obtain a coded representation of said coefficients, transmitting the coded representation of said predictor coefficients to a decoder and synthesizing the original input speech signal in accordance with said transmitted coded representation of said predictor coefficients, said coding step comprising: generating a multi-component input vector corresponding to said set of predictor coefficients for one analysis time period, with each component of said vector corresponding to a frequency; quantizing said input vector by selecting a plurality of multi-component quantization vectors from a quantization vector storage means and calculating for each selected quantization vector a distortion measure in accordance with the difference between each component of said input vector and each corresponding component of the selected quantization vector, and in accordance with a weighting factor associated with each component of said input vector, the weighting factor being determined for each component of said input vector in accordance with the frequency to which said component corresponds; selecting as a quantizer output the one of said plurality of selected quantization vectors resulting in the least distortion measure; and generating said transmitted coded representation in accordance with the selected quantizer output.

Claim 11

A speech analysis and synthesis method as defined in claim 7, wherein said weighting factor is given by ##EQU25## where ##EQU26## where f i denotes the frequency represented by the ith component of the input vector, D i denotes a group delay for f i in milliseconds, and D max is a maximum group delay.

Claim 12

A speech analysis and synthesis method as defined in claim 8, wherein said distortion measure is given by ##EQU27## where X i , γ i denote respectively, the components of the input vector and the corresponding components of each selected quantization vector, and ω is the corresponding weighting factor.

Claim 13

A speech analysis and synthesis system comprising: excitation signal generating means for generating for each of a plurality of analysis time periods of an input speech signal a multipulse excitation signal comprising a sequence of excitation pulses each having an amplitude and a position within said analysis time period, said excitation signal generating means comprising: means for storing a plurality of pulse amplitude codewords; means for storing a plurality of pulse position codewords; and means for reading a pulse amplitude codeword and a pulse position codeword to form said multipulse excitation pulse; and means for subsequently regenerating said speech signal in accordance with said multipulse excitation signals.

Claim 14

A speech analysis and synthesis system comprising: excitation signal generating means for generating for each of a plurality of analysis time periods of an input speech signal a multipulse excitation signal comprising a sequence of excitation pulses each having an amplitude and a position within said analysis time period, said excitation signal generating means comprising: means for storing a plurality of pulse amplitude codewords; means for storing a plurality of pulse position codewords; and means for reading a pulse amplitude codeword and a pulse position codeword to form said multipulse excitation pulse; and means for subsequently regenerating said speech signal in accordance with said multipulse excitation signals.

Claim 15

A speech analysis and synthesis method comprising the steps of: generating for each of a plurality of analysis time periods of an input speech signal a multipulse excitation vector representing a sequence of excitation pulses each having an amplitude and a position within said analysis time period, said generating step comprising: selecting a pulse position codeword from a stored plurality of pulse position codewords; selecting a pulse amplitude codeword from a stored plurality of pulse amplitude codewords; and combining said selected pulse position and pulse amplitude codewords to form said multipulse excitation vector; and subsequently regenerating said speech signal in accordance with said multipulse excitation vector.

Claim 16

A speech analysis and synthesis method comprising the steps of: generating for each of a plurality of analysis time periods of an input speech signal a multipulse excitation vector representing a sequence of excitation pulses each having an amplitude and a position within said analysis time period, said generating step comprising: selecting a pulse position codeword from a stored plurality of pulse position codewords; selecting a pulse amplitude codeword from a stored plurality of pulse amplitude codewords; and combining said selected pulse position and pulse amplitude codewords to form said multipulse excitation vector; and subsequently regenerating said speech signal in accordance with said multipulse excitation vector.

Claim 17

A speech analysis and synthesis method as defined in claim 11, wherein each multipulse excitation vector is of the form V=(m 1 , . . . , m L , g 1 , . . . , g L ), where L is the total number of excitation pulses represented by said vector, m L and g L are pulse position and pulse amplitude codewords, respectively, corresponding to the L-th excitation pulse in said vector, and wherein said step of selecting a pulse position codeword comprises determining a position m I within said analysis time period at which the absolute value of g I has a maximum value, where m I and g I are the position and amplitude of an I-th excitation pulse; and selecting a pulse position codeword m i for said I-th excitation pulse in accordance with the determined value of m I .

Claim 18

A speech analysis and synthesis method as defined in claim 12, wherein said step of selecting a pulse amplitude codeword comprises the steps of: calculating an amplitude g I for said I-th excitation pulse in accordance with said determined position M I .

Claim 19

A speech analysis and synthesis method as defined in claim 12, wherein said speech signal is regenerated using a synthesis filter, and wherein g I is given by: ##EQU28## wherein X w (n) is a weighted speech signal and h w (n) is a weighted impulse response of said synthesis filter.

Claim 20

A speech analysis and synthesis method as defined in claim 12, wherein said speech signal is regenerated using a synthesis filter, and wherein g I is given by: ##EQU29## where R hh (m) is the autocorrelation of h w (n), h w (n) is a weighted impulse response of said synthesis filter, R hx (m) is the crosscorrelation between h w (n) and X w (n), and X w (n) is a weighted speech signal.

Claim 21

A speech analysis and synthesis method as defined in claim 12, wherein said step of selecting a pulse position codeword comprises: determining a position m 1 within said analysis time period at which R hx (m) has a maximum value, where R hx (m) is the crosscorrelation between a weighted impulse response h w (n) of said synthesis filter and a weighted speech signal X w (n); and selecting a pulse position codeword in accordance with said determined position m 1 .

Claim 22

A speech analysis and synthesis method as defined in claim 16, wherein said step of selecting a pulse amplitude codeword comprises: determining a value for the amplitude g 1 of said first excitation pulse according to: ##EQU30## where R hh (0) is the autocorrelation of h w (0).

Claim 23

A speech analysis and synthesis method as defined in claim 11 wherein each said multipulse excitation vector is of the form V=(m 1 , . . . , m L , g 1 , . . . , g L ), where L is the total number of excitation pulses represented by said vector, m i and g i 1≦i≦L, are position-related and amplitude-related terms, respectively, corresponding to the i-th excitation pulse in said vector, said method further comprising coding said vectors and decoding said vectors prior to said regenerating step, said coding step comprising: generating from said vector V a position reference subvector V m and an amplitude reference subvector vector V g ; selecting from a position codebook a plurality of position codewords in accordance with said position reference subvector; selecting from an amplitude codebook a plurality of amplitude codewords in accordance with said amplitude reference subvector; generating a plurality of position codeword/amplitude codeword pairs from various combinations of said selected position and amplitude codewords; calculating a distortion measure between said multipulse excitation vector and each position codeword/amplitude codeword pair; and selecting a position codeword/amplitude codeword pair resulting in the lowest distortion measure.

Claim 24

A speech analysis and synthesis method comprising the steps of: generating for each of a plurality of analysis time periods of an input speech signal a multipulse excitation vector representing a sequence of excitation pulses each having an amplitude and a position within said analysis time period, coding said multipulse excitation vectors, wherein said coding step comprises: generating for each multipulse excitation vector a difference excitation vector which is a function of the difference between said each multipulse excitation vector and a reference multipulse excitation vector; and quantizing said difference excitation vector to obtain said coded multipulse excitation vectors; decoding the coded multipulse excitation vectors; and subsequently regenerating said speech signal in accordance with decoded multipulse excitation vectors.

Claim 25

A speech analysis and synthesis method comprising the steps of: generating for each of a plurality of analysis time periods of an input speech signal a multipulse excitation vector representing a sequence of excitation pulses each having an amplitude and a position within said analysis time period, coding said multipulse excitation vectors, wherein said coding step comprises: generating for each multipulse excitation vector a difference excitation vector which is a function of the difference between said each multipulse excitation vector and a reference multipulse excitation vector; and quantizing said difference excitation vector to obtain said coded multipulse excitation vectors; decoding the coded multipulse excitation vectors; and subsequently regenerating said speech signal in accordance with decoded multipulse excitation vectors.

Claim 26

A speech analysis and synthesis method as defined in claim 19, wherein each multipulse excitation vector is of the form V=(m 1 , . . . , m L , g 1 , . . . , g L ), where L is the total number of excitation pulses represented by said vector, m i and g i , 1≦i≦L, are pulse position and pulse amplitude codewords, respectively, corresponding to the i-th excitation pulse in said vector, and wherein said difference excitation vector is given by V=(m 1 , . . . , m L , g 1 , . . . , g L ), where m.sub.i =m.sub.i -m.sub.1 ')/m.sub.1 " and g.sub.i =g.sub.i /G where m 1 ' and m' are taken from first and second reference vectors V'=(m 1 ', . . . , m L ', g 1 ', . . . , g L ') and V"=(m 1 ", . . . , m L ", g 1 ", . . . , g L ") prepared in advance from a large speech data base, and G is a gain term given by ##EQU31##

Claim 27

A speech analysis and synthesis method as define din claim 20, wherein m 1 ' is the mean of all values of m i in said large speech data base.

Claim 28

A speech analysis and synthesis method as defined in claim 21, wherein m 1 " is the standard deviation of all values of m i in said large speech data base.

Claim 29

A speech analysis and synthesis method as defined in claim 20, wherein said coding step further comprises separating said difference vector into a position subvector (m 1 , . . . , m L ) and an amplitude subvector (g 1 , . . . , g L ), and then quantizing said position subvector in a first quantizer and quantizing said amplitude subvector in a second quantizer.

Claim 30

A speech analysis and synthesis method comprising the steps of: generating for each of a plurality of analysis time periods of an input speech signal a vector representing a sequence of excitation pulses each having an amplitude and a position within said analysis time period, each of said vectors being of the form V=(m 1 , . . . , m L , g 1 , . . . , g L ), where L is the total number of excitation pulses represented by said vector, m i and g i , 1≦i≦L, are position-related and amplitude-related terms, respectively, corresponding to the i-th excitation pulse in said vector; coding said vectors, wherein said coding step comprises separating said vector into a position subvector (m 1 , . . . , m L ) and an amplitude subvector (g 1 , . . . , g L ), and then quantizing said position subvector in a first quantizer and quantizing said amplitude subvector in a second quantizer, with the quantized position subvector and quantized amplitude subvector together comprising said coded vector; decoding the coded vectors; and subsequently regenerating said speech signal in accordance with decoded vectors.

Claim 31

A speech analysis and synthesis method comprising the steps of: generating for each of a plurality of analysis time periods of an input speech signal a vector representing a sequence of excitation pulses each having an amplitude and a position within said analysis time period, each of said vectors being of the form V=(m 1 , . . . , m L , g 1 , . . . , g L ), where L is the total number of excitation pulses represented by said vector, m i and g i , 1≦i≦L, are position-related and amplitude-related terms, respectively, corresponding to the i-th excitation pulse in said vector; coding said vectors, wherein said coding step comprises separating said vector into a position subvector (m 1 , . . . , m L ) and an amplitude subvector (g 1 , . . . , g L ), and then quantizing said position subvector in a first quantizer and quantizing said amplitude subvector in a second quantizer, with the quantized position subvector and quantized amplitude subvector together comprising said coded vector; decoding the coded vectors; and subsequently regenerating said speech signal in accordance with decoded vectors.

Claim 32

A speech analysis and synthesis method comprising the steps of: generating, for each of a plurality of analysis time periods of an input speech signal, a vector representing a sequence of excitation pulses each having an amplitude and a position within said analysis time period, each said vector being is of the form V=(m 1 , . . . , m L , g 1 , . . . , g L ), where L is the total number of excitation pulses represented by said vector, m i and g i , 1≦i≦L, are position-related and amplitude-related terms, respectively, corresponding to the i-th excitation pulse in said vector; coding said vectors, wherein said coding step comprises: generating from a given one of said vectors a position reference subvector V m and an amplitude reference subvector vector V g ; selecting from a position codebook a plurality of position codewords in accordance with said position reference subvector; selecting from an amplitude codebook a plurality of amplitude codewords in accordance with said amplitude reference subvector; generating a plurality of position codeword/amplitude codeword pairs from various combinations of said selected position and amplitude codewords; calculating a distortion measure between said given vector and each position codeword/amplitude codeword pair; and selecting a position codeword/amplitude codeword pair resulting in the lowest distortion measure as a coded version of said given vector; decoding the coded vectors; and subsequently regenerating said speech signal in accordance with decoded vectors.

Claim 33

A speech analysis and synthesis method comprising the steps of: generating, for each of a plurality of analysis time periods of an input speech signal, a vector representing a sequence of excitation pulses each having an amplitude and a position within said analysis time period, each said vector being is of the form V=(m 1 , . . . , m L , g 1 , . . . , g L ), where L is the total number of excitation pulses represented by said vector, m i and g i , 1≦i≦L, are position-related and amplitude-related terms, respectively, corresponding to the i-th excitation pulse in said vector; coding said vectors, wherein said coding step comprises: generating from a given one of said vectors a position reference subvector V m and an amplitude reference subvector vector V g ; selecting from a position codebook a plurality of position codewords in accordance with said position reference subvector; selecting from an amplitude codebook a plurality of amplitude codewords in accordance with said amplitude reference subvector; generating a plurality of position codeword/amplitude codeword pairs from various combinations of said selected position and amplitude codewords; calculating a distortion measure between said given vector and each position codeword/amplitude codeword pair; and selecting a position codeword/amplitude codeword pair resulting in the lowest distortion measure as a coded version of said given vector; decoding the coded vectors; and subsequently regenerating said speech signal in accordance with decoded vectors.

Claim 34

A speech analysis and synthesis method as defined in claim 25, wherein said distortion measure comprises a dynamically weighted distortion measure weighted in accordance with a weighting function which is a function of the amplitude of each amplitude term in each position codeword/amplitude codeword pair.

Claim 35

A speech analysis and synthesis method as defined in claim 26, wherein said dynamically weighted distortion measure D is given by, ##EQU32## where ω i is said weighting function and is given by ##EQU33## where x i denotes a component of said vector, and y i denotes a corresponding component of a position codeword/amplitude codeword pair.

Claim 36

A speech analysis and synthesis method comprising the steps of: generating a plurality of analysis signals from an input signal, said analysis signals comprising at least a pitch signal portion including a pitch value and a pitch gain value, and an excitation signal portion including an excitation codeword and an excitation gain signal; coding said analysis signals, wherein said coding step includes the steps of: classifying each of said pitch signal portions and excitation signal portions as significant or insignificant; allocating a number of coding bits to each of said pitch signal portions and excitation signal portions in accordance with results of said classifying step; and coding each of said pitch and excitation signals with the number of bits allocated to each; and decoding said analysis signals; and synthesizing said coded speech signal in accordance with the decoded analysis signals.

Claim 37

A speech analysis and synthesis method comprising the steps of: generating a plurality of analysis signals from an input signal, said analysis signals comprising at least a pitch signal portion including a pitch value and a pitch gain value, and an excitation signal portion including an excitation codeword and an excitation gain signal; coding said analysis signals, wherein said coding step includes the steps of: classifying each of said pitch signal portions and excitation signal portions as significant or insignificant; allocating a number of coding bits to each of said pitch signal portions and excitation signal portions in accordance with results of said classifying step; and coding each of said pitch and excitation signals with the number of bits allocated to each; and decoding said analysis signals; and synthesizing said coded speech signal in accordance with the decoded analysis signals.

Claim 38

A speech analysis and synthesis method as define din claim 28, wherein said allocating step comprises allocating a greater number of bits to a pitch signal portion classified as significant than to a pitch signal portion classified as insignificant, and allocating a greater number of bits to an excitation signal portion classified as significant than to an excitation signal classified as insignificant.

Claim 39

A speech analysis and synthesis method as defined in claim 29, wherein said allocating step comprises allocating zero bits to said pitch signal portion if it is classified as insignificant, and allocating zero bits to said excitation signal portion if it is classified as insignificant.

Claim 40

A speech activity detector for use in an apparatus for encoding an input signal having speech and non-speech portions, for determining the speech or non-speech character of said input signal over each of a plurality of successive intervals, said speech activity detector comprising monitoring means for monitoring an energy content of said input speech signal and discriminating means responsive to the monitored energy for discriminating between speech and non-speech input signals, said monitoring means comprising means for determining an average energy of said input signal over one of said intervals and means for determining a minimum value of said average energy over a predetermined number of said intervals; and said discriminating means comprising means for determining a threshold value in accordance with said minimum value and means for comparing said average energy of said input signal over said one interval to said threshold value to determine if said input signal during said one interval represents speech or non-speech.

Claim 41

A speech activity detector for use in an apparatus for encoding an input signal having speech and non-speech portions, for determining the speech or non-speech character of said input signal over each of a plurality of successive intervals, said speech activity detector comprising monitoring means for monitoring an energy content of said input speech signal and discriminating means responsive to the monitored energy for discriminating between speech and non-speech input signals, said monitoring means comprising means for determining an average energy of said input signal over one of said intervals and means for determining a minimum value of said average energy over a predetermined number of said intervals; and said discriminating means comprising means for determining a threshold value in accordance with said minimum value and means for comparing said average energy of said input signal over said one interval to said threshold value to determine if said input signal during said one interval represents speech or non-speech.

Claim 42

A speech activity detector as defined in claim 31, wherein said one interval is the last of said predetermined number of intervals.

Claim 43

A speech activity detector as defined in claim 31, further comprising: means responsive to the determination that said average energy in said one frame exceeds said threshold value for setting a hangover value in accordance with the number of consecutive intervals for which said threshold has been exceeded; and means responsive to a determination that said average energy for said one interval does not exceed said threshold value for determining that said input signal represents a non-speech portion if said hangover value is at a predetermined level, and otherwise decrementing said hangover value.

Claim 44

A speech detector for discriminating between speech and non-speech intervals of an input signal, said speech detector comprising monitoring means for monitoring at least one characteristic of said input signal and discriminating means responsive to said monitoring means for discriminating between speech and non-speech input signals, wherein said monitoring means comprises first means for determining if said one characteristic of said input signal for a present interval meets at least a first criterion of a signal representing speech and wherein said discriminating means comprises second means responsive to a determination of speech by said first means for setting a predetermined hangover time in accordance with a number of consecutive intervals for which said input signal has been determined to satisfy said first criterion, and third means responsive to a determination by said first means that said input signal does not satisfy said criterion for determining non-speech in accordance with a number of consecutive intervals for which said criterion has not been satisfied and in accordance with the hangover time set by said second means.

Claim 45

A speech detector for discriminating between speech and non-speech intervals of an input signal, said speech detector comprising monitoring means for monitoring at least one characteristic of said input signal and discriminating means responsive to said monitoring means for discriminating between speech and non-speech input signals, wherein said monitoring means comprises first means for determining if said one characteristic of said input signal for a present interval meets at least a first criterion of a signal representing speech and wherein said discriminating means comprises second means responsive to a determination of speech by said first means for setting a predetermined hangover time in accordance with a number of consecutive intervals for which said input signal has been determined to satisfy said first criterion, and third means responsive to a determination by said first means that said input signal does not satisfy said criterion for determining non-speech in accordance with a number of consecutive intervals for which said criterion has not been satisfied and in accordance with the hangover time set by said second means.

Claim 46

A speech analysis and synthesis method comprising the steps of: deriving a set of synthesis parameters for each frame from an original input signal having a plurality of successive frames including a current frame, a previous frame and a next frame, with each frame having first, second and third portions, said step of deriving said synthesis parameters comprising: generating a set of first parameters corresponding to each frame of said input signal, each set of first parameters for a given frame including first, second and third subsets corresponding to said first, second and third portions of the given frame; generating an interpolated first subset of parameters by interpolating between said first subsets of said current and previous frames; generating an interpolated third subset of parameters by interpolating between said third subsets of said current and next frames; combining said interpolated first subset, said second subset and said interpolated third subset of parameters to form a set of synthesis parameters for said current frame; transmitting the synthesis parameters to a decoder; and synthesizing the original input speech signal in accordance with said transmitted synthesis parameters.

Claim 47

A speech analysis and synthesis method comprising the steps of: deriving a set of synthesis parameters for each frame from an original input signal having a plurality of successive frames including a current frame, a previous frame and a next frame, with each frame having first, second and third portions, said step of deriving said synthesis parameters comprising: generating a set of first parameters corresponding to each frame of said input signal, each set of first parameters for a given frame including first, second and third subsets corresponding to said first, second and third portions of the given frame; generating an interpolated first subset of parameters by interpolating between said first subsets of said current and previous frames; generating an interpolated third subset of parameters by interpolating between said third subsets of said current and next frames; combining said interpolated first subset, said second subset and said interpolated third subset of parameters to form a set of synthesis parameters for said current frame; transmitting the synthesis parameters to a decoder; and synthesizing the original input speech signal in accordance with said transmitted synthesis parameters.

Claim 48

A speech analysis and synthesis method as define din claim 35, wherein said first set of parameters comprise line spectrum frequencies.

Claim 49

A speech analysis and synthesis method, comprising: deriving a set of spectrum filter coefficients for each frame from an original input signal representing speech and having a plurality of successive frames; converting said spectrum filter coefficients to an ordered set of n frequency parameters (f 1 , f 2 , . . . , f n ), where n is an integer; determining if any magnitude ordering has been violated, i.e., if f i <f i-1 , where i is an integer between 1 and n; if any magnitude ordering has been violated, rearranging said frequency parameters by reversing the order of the two frequencies f i and f i-1 which resulted in the violation; converting said frequency parameters, after any rearrangement if that has occurred, back to spectrum filter coefficients; and synthesizing said original input signal representing said speech in accordance with the spectrum filter coefficients resulting from said converting step.

Claim 50

A speech analysis and synthesis method, comprising: deriving a set of spectrum filter coefficients for each frame from an original input signal representing speech and having a plurality of successive frames; converting said spectrum filter coefficients to an ordered set of n frequency parameters (f 1 , f 2 , . . . , f n ), where n is an integer; determining if any magnitude ordering has been violated, i.e., if f i <f i-1 , where i is an integer between 1 and n; if any magnitude ordering has been violated, rearranging said frequency parameters by reversing the order of the two frequencies f i and f i-1 which resulted in the violation; converting said frequency parameters, after any rearrangement if that has occurred, back to spectrum filter coefficients; and synthesizing said original input signal representing said speech in accordance with the spectrum filter coefficients resulting from said converting step.

Claim 51

A speech analysis and synthesis method as defined in claim 37, wherein said frequency parameters comprise line spectrum frequencies.

Claim 52

A speech analysis and synthesis method comprising the steps of: generating a plurality of analysis signals from an input signal, said analysis signals comprising at least a pitch value, a pitch gain value, an excitation codeword and an excitation gain signal, quantizing said analysis signals, wherein said quantizing step comprises: quantizing said pitch value directly by classifying said pitch value into one of a plurality of 2 m value ranges, where m is an integer, with m quantization bits representing the classification value; and quantizing said pitch gain by selecting a corresponding codeword from a codebook of 2 n codewords, where n is an integer, with n quantization bits representing the selected codeword; providing the quantized analysis signals to a decoder, and synthesizing said speech signal in accordance with the quantized signals at the decoder.

Claim 53

A speech analysis and synthesis method comprising the steps of: generating a plurality of analysis signals from an input signal, said analysis signals comprising at least a pitch value, a pitch gain value, an excitation codeword and an excitation gain signal, quantizing said analysis signals, wherein said quantizing step comprises: quantizing said pitch value directly by classifying said pitch value into one of a plurality of 2 m value ranges, where m is an integer, with m quantization bits representing the classification value; and quantizing said pitch gain by selecting a corresponding codeword from a codebook of 2 n codewords, where n is an integer, with n quantization bits representing the selected codeword; providing the quantized analysis signals to a decoder, and synthesizing said speech signal in accordance with the quantized signals at the decoder.

Claim 54

A speech analysis and synthesis method as define din claim 39, wherein n<m.

Claim 55

A speech analysis and synthesis method as define din claim 39, wherein said quantizing step further comprises: representing said excitation codeword with k bits indicating the one of 2 k codewords from which said excitation codeword was selected; and quantizing said excitation gain by selecting a corresponding codeword from a codebook of 2 l previously computed excitation gain codewords, where l is an integer, with l quantization bits representing the selected excitation gain codeword.

Claim 56

A speech analysis and synthesis method as defined in claim 41, wherein l<k.