Celp voice encoder

Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission

Reexamination Certificate

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Details

C704S220000, C704S207000

Reexamination Certificate

active

06804639

ABSTRACT:

TECHNICAL FIELD
The present invention relates to a CELP (Code Excited Linear Prediction) type speech coding apparatus which encodes a speech signal to transmit in, for example, a mobile communication system.
BACKGROUND ART
Used in the fields of digital mobile communications and speech storage are speech coding apparatuses which compress speech information to encode with high efficiency for utilization of radio signals and recording media. Among them, the system based on a CELP (Code Excited Linear Prediction) system is carried into practice widely for the apparatuses operating at medium to low bit rates. The technology of the CELP is described in “Code-Excited Linear Prediction (CELP):High-quality Speech at Very Low Bit Rates” by M. R. Schroeder and B. S. Atal, Proc. ICASSP-85, 25.1.1., pp.937-940, 1985.
In the CELP type speech coding system, speech signals are divided into predetermined frame lengths (about 5 ms to 50 ms), linear prediction of the speech signals is performed for each frame, the prediction residual (excitation vector signal) obtained by the linear prediction for each frame is coded using an adaptive code vector and random code vector comprised of known waveforms.
The adaptive code vector is selected for use from an adaptive codebook storing previously generated excitation vectors, and the random code vector is selected for use from a random codebook storing a predetermined number of pre-prepared vectors with predetermined shapes.
In particular, used as the random code vectors stored in the random codebook are, for example, random noise sequence vectors and vectors generated by arranging a few pulses at different positions. In particular, one of representative examples of the latter is CS-ACELP (Conjugate Structure and Algebraic CELP) recommended as an international standard by ITU-T in 1996. The technology of the CS-ACELP is described in “Recommendation G.729:Coding of Speech at 8 kbit/s using Conjugate-Structure Algebraic-Code-Excited Linear-Prediction (CS-ACELP)”, March 1996.
In addition, the CS-ACELP uses an algebraic codebook as a random codebook. The random code vector generated from the algebraic codebook in the CS-ACELP is such a vector that four impulses each with an amplitude of −1 or +1 are prepared (regions other than positions with the four prepared pulses are basically all 0)in 40 samples (5 ms) per subframe basis. Since an absolute value of the amplitude is fixed to 1, it is enough to represent only a position and polarity (positive or negative) of each pulse to represent an excitation vector. Therefore it is not necessary to store a vector with 40 dimensions (subframe length) in a codebook, and a memory for codebook storage is not required. Further since four pulses with amplitudes of 1 are only present in the vector, this method has futures such that the computation amount for codebook search is largely reduced.
In the CS-ACELP, adaptive code vector information is efficiently coded by representing a pitch of a second subframe by performing quantization on a pitch differential value using a pitch of a first subframe. Further in the pitch search, a constitution is adopted in which one pitch candidate is selected by open loop pitch search for each frame, and closed loop pitch search for each subframe is performed around the pitch candidate, whereby it is designed to also reduce the computation amount required for the search.
Herein a conventional CS-ACELP coding apparatus is specifically explained with reference to FIG.
1
.
FIG. 1
illustrates a basic configuration of the conventional CS-ACELP speech coding apparatus. In
FIG. 1
, input buffer
1
performs buffering of data with a required length while updating an input digital speech signal for each frame, and outputs required data to subframe divider
2
, LPC analyzer
3
, and weighted synthesis filter
4
.
Subframe divider
2
divides a frame of the input digital signal, input from input buffer
1
, into two subframes, outputs a first subframe signal to first target calculator
5
, and further outputs a second subframe signal to second target calculator
6
. LPC analyzer
3
receives a digital speech signal required for analysis input from input buffer
1
to perform LPC analysis, and outputs linear predictive coefficients to LPC quantizer
7
and second LPC interpolator
8
. Weighted synthesis filter
4
receives as inputs the frame of the digital speech signal input from input buffer
1
and linear predictive coefficients a
1
and a
2
output from second LPC interpolator
8
, and performs perceptual weighting on the input speech signal to output to open loop pitch searcher
9
.
LPC quantizer
7
performs quantization on the linear predictive coefficients output from LPC analyzer
3
, outputs quantized LPC to first LPC interpolator
10
, and at the same time outputs coding data L of the quantized LPC to a decoder. Second LPC interpolator
8
receives as inputs the LPC output from LPC analyzer
3
, performs interpolation on LPC of the first subframe, and outputs unquantized LPC of the first and second subframes respectively as a
1
and a
2
. First LPC interpolator
10
receives as inputs the quantized LPC output from LPC quantizer
7
, performs interpolation on the quantized LPC of the first subframe, and outputs quantized LPC of the first and second subframes respectively as qa
1
and qa
2
.
First target calculator
5
receives as inputs the first subframe of the digital speech signal divided in subframe divider
2
, filter state st
1
output from second filter state updator
11
on the last second subframe, and qa
1
and a
1
that are respectively the quantized LPC and unquantized LPC of the first subframe, and calculates a target vector to output to first closed loop pitch searcher
12
, first target updator
13
, first gain codebook searcher
14
, and first filter state updator
15
. Second target calculator
6
receives as inputs the second subframe of the digital speech signal output from subframe divider
2
, filter state st
2
output from first filter state updator
15
on the first subframe of a current frame, and qa
2
and a
2
that are respectively the quantized LPC and unquantized LPC of the second subframe, and calculates a target vector to output to second closed loop pitch searcher
16
, second target updator
17
, second gain codebook searcher
18
, and second filter state updator
11
.
Open loop pitch searcher
9
receives as an input a weighted input speech signal output from weighted synthesis filter
4
to extract a pitch periodicity, and outputs an open loop pitch period to first closed loop pitch searcher
12
. First closed loop pitch searcher
12
receives a first target vector, open loop pitch, adaptive code vector candidates, and an impulse response vector respectively input from first target calculator
5
, open loop pitch searcher
9
, adaptive codebook
19
, and first impulse response calculator
20
, performs closed loop pitch search around the open loop pitch, outputs closed loop pitch P
1
to second closed loop pitch searcher
16
, first pitch period processing filter
21
and the decoder, outputs an adaptive code vector to first excitation generator
22
, and further outputs a synthetic vector obtained by performing convolution of the first impulse response and the adaptive code vector to first target updator
13
, first gain codebook searcher
14
, and first filter state updator
15
.
First target updator
13
receives the first target vector and a first adaptive code synthetic vector respectively input from first target calculator
5
and first closed loop pitch searcher
12
, and calculates a target vector for the random codebook to output to first random codebook searcher
23
. First gain codebook searcher
14
receives the first target vector, the first adaptive code synthetic vector, and a first random code synthetic vector respectively input from first target calculator
5
, first closed loop pitch searcher
12
and first random codebook searcher
23
, and selects an optimum quantized gain from gain codebook
29
to output to first excitation

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