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Patent EP0476614A2 successfully refreshed from Google Patents.
EP0476614A2
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Speech coding and decoding system
Assignee: Fujitsu Ltd
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Filing: 1991-09-18
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Granted: —
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Expires: 2011-09-18
Abstract
Abstract A CELP type speech coding system which is provided with an arithmetic processing unit (31) which transforms a perceptual weighted input speech signal vector AX to a vector <t>AAX, a sparse adaptive codebook (1) which stores a plurality of pitch prediction residual vectors P sparsed by a sparse unit, a multiplying unit (32) which multiplies the successively read out vectors P and the outcome <t>AAX from the arithmetic processing unit (31), a filter operation unit (33) which performs a filter operation on the vectors P and an evaluation unit (34) which finds the optimum vector P based on the output from the filter operation unit (33), so as to enable reduction of the amount of arithmetic operations. <IMAGE>
Inventors
- 👤 Tomohiko c/o Fujitsu Limited Taniguchi
- 👤 Mark Johnson
- 👤 Hideaki c/o Fujitsu Limited Kurihara
- 👤 Yoshinori c/o Fujitsu Limited Tanaka
- 👤 Yasuji c/o Fujitsu Limited Ohta
Bibliographic Details
Application Number
—
Jurisdiction
EP
Data Source
google_patents
Last Synchronized
2026-11-04 02:54
CPC Classifications
G; G10; G10L; G10L19/00; G10L19/04; G10L19/08; G10L19/12; G10L2019/0001; G10L2019/0002; G10L2019/0011; G10L25/00; G10L25/03; G10L25/06
Claims Analysis
Total Claims: 11
Claim 1
Independent Claim
274 words
A speech coding and decoding system which includes an adaptive codebook (1a) which stores a plurality of pitch prediction residual vectors (P) and a stochastic codebook (2) which stores a plurality of code vectors (C) comprised of white noise, whereby use is made of indexes showing an optimum pitch prediction residual vector (bP) and optimum code vector (gC) (b and g gains) closest to a perceptually weighted input speech signal vector (AX) so as to code an input speech signal, a decoder side reproducing the speech signals in accordance with the code, characterized in that the adaptive codebook (1a) is constituted as a sparse adaptive codebook (1) which stores a plurality of sparsed pitch prediction residual vectors (P) and the system comprises a first means (31) which arithmetically processes a time-reversing perceptual weighted input speech signal ( t AAX) from the perceptually weighted input speech signal vector (AX); a second means (32) which receives at a first input the time reversing perceptual weighted input speech signal output from the first means, receives at its second input the pitch prediction residual vectors (P) successively output from the sparse adaptive codebook, and multiplies the two input values so as to produce a correlation value ( t (AP)AX) of the same; a third means (33) which receives as input the pitch prediction residual vectors and finds the autocorrelation value ( t (AP)AP) of the vector (AP) after perceptual weighting reproduction; and a fourth means (34) which receives as input the correlation values from the second means (32) and third means (33), evaluates the optimum pitch prediction residual vector and optimum code vector, and decides on the same.
Claim 2
Dependent Claim
46 words
A system as set forth in claim 1, which has a fifth means (35) which is connected to the sparse adaptive codebook (1), adds the optimum pitch prediction residual vector and the optimum code vector, performs sparsing, and stores the results in the sparse adaptive codebook.
Claim 3
Dependent Claim
78 words
A system as set forth in claim 2, wherein said fifth means (35) is comprised of an adder (36) which adds in time series the optimum pitch prediction residual vector and the optimum code vector; a sparse unit (37) which receives as input the output of the adder; and a delay unit (14) which gives a delay corresponding to one frame to the output of the sparse unit and stores the result in the sparse adaptive codebook (1).
Claim 4
Dependent Claim
35 words
A system as set forth in claim 2, wherein said first means (31) is composed of a transposition matrix ( t A) obtained by transposing a finite impulse response (FIR) perceptual weighting filter matrix (A).
Claim 5
Dependent Claim
66 words
A system as set forth in claim 2, wherein the first means (31) is composed of a front processing unit (41) which rearranges time reversely the input speech signal vector (AX) along the time axis, an infinite impulse response (IIR) perceptual weighting filter (42), and a rear processing unit (43) which rearranges time reversely the output of the filter (42) once again along the time axis.
Claim 6
Dependent Claim
53 words
A system as set forth in claim 4, wherein when the FIR perceptual weighting filter matrix (A) is expressed by the following: the transposition matrix ( t A), that is, is multiplied with the input speech signal vector, that is, and the first means (31) outputs the following: (where, the asterisk means multiplication)
Claim 7
Dependent Claim
71 words
A system as set forth in claim 5, wherein when the input speech signal vector (AX) is expressed by the following: the front processing unit (41) generates the following: (where TR means time reverse) and this (AX) TR , when passing through the next IIR perceptual weighting filter (42), is converted to the following: and this A(AX) TR is output from the next rear processing unit (43) as W, that is:
Claim 8
Dependent Claim
63 words
A system as set forth in claim 3, wherein the sparse unit (37) is operative to selectively supply to the delay unit (14) only outputs of the adder (36) where the absolute value of the level of the outputs exceeds the absolute value of a fixed threshold level, transform all other outputs to zero, and exhibit a center clipping characteristic as a whole.
Claim 9
Dependent Claim
104 words
A system as set forth in claim 3, wherein the sparse unit (37) is operative, first of all, to take out the output of the adder (36) at certain intervals corresponding to a plurality of sample points, find the absolute value of the outputs of each of the sample points, then give ranking successively from the outputs with the large absolute values to the ones with the small ones, selectively supply to the delay unit (14) only the outputs corresponding to the plurality of sample points with high ranks, transform all other outputs to zero, and exhibit a center clipping characteristic as a whole.
Claim 10
Dependent Claim
99 words
A system as set forth in claim 3, wherein the sparse unit (37) is operative to selectively supply to the delay unit (14) only the outputs of the adder (36) where the absolute values of the outputs exceed the absolute value of the given threshold level and transform the other outputs to zero, where the absolute value of the threshold is made to change adaptively to become higher or lower in accordance with the degree of the average signal amplitude obtained by taking the average of the outputs over time and exhibits a center clipping characteristic as a whole.
Claim 11
Dependent Claim
91 words
A system as set forth in claim 2, wherein the decoder side is comprised of substantially the same constituent elements as the constituent elements of the coding side so as to receive the code transmitted from the coding side and reproduce the speech signal in accordance with the code and wherein it has a linear prediction code (LPC) reproducing filter (107) which receives as input a signal corresponding to the sum of the optimum pitch prediction residual vector (bP) and the optimum code vector (gC) and produces a reproduced speech signal.
References Cited (Backward Citations)
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