Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission
Reexamination Certificate
1998-02-03
2002-02-05
Tsang, Fan (Department: 2648)
Data processing: speech signal processing, linguistics, language
Speech signal processing
For storage or transmission
C704S500000
Reexamination Certificate
active
06345246
ABSTRACT:
BACKGROUND OF THE INVENTION
The present invention relates to a coding method that permits efficient coding of plural channels of an acoustic signal, such as speech or music, and is particularly suitable for its transmission at low bit rates, a method for decoding such a coded signal and encoder and decoder using the coding and decoding methods, respectively.
It is well-known in the art to quantize a speech, music or similar acoustic signal in the frequency domain with a view to reducing the number of bits for coding the signal. The transformation from the time to frequency domain is usually performed by DFT (Discrete Fourier Transform), DCT (Discrete Cosine Transform) and MDCT (Modified Discrete Cosine Transform) that is a kind of Lapped Orthogonal Transform (LOT). It is also well-known that a linear predictive coding (LPC) analysis is effective in flattening frequency-domain coefficients (i.e. spectrum samples) prior to the quantization. As an example of a method for high-quality coding of a wide variety of acoustic signals through the combined use of these techniques, there are disclosed acoustic signal transform coding and decoding methods, for example, in Japanese Patent Application Laid-Open Gazette No. 44399/96 (corresponding U.S. Pat. No. 5,684,920). In
FIG. 1
there is depicted in a simplified form the configuration of a coding device that utilizes the disclosed method.
In
FIG. 1
, an acoustic signal from an input terminal
11
is applied to an orthogonal transform part
12
, wherein it is transformed to coefficients in the frequency domain through the use of the above-mentioned scheme. The frequency-domain coefficients will hereinafter be referred to as spectrum samples. The input acoustic signal also undergoes linear predictive coding (LPC) analysis in a spectral envelope estimating part
13
. By this, the spectral envelope of the input acoustic signal is detected. That is, in the orthogonal transform part
12
the acoustic digital signal from the input terminal
11
is transformed to spectrum sample values through Nth-order lapped orthogonal transform (MDCT, for instance) by extracting an input sequence of the past 2N samples from the acoustic signal every N samples. In an LPC analysis part
13
A of a spectral envelope estimating part
13
, too, a sequence of 2N samples are similarly extracted from the input acoustic digital signal every N samples. From the thus extracted samples d are derived Pth-order predictive coefficients &agr;
0
, . . . , &agr;
P
. These predictive coefficients &agr;
0
, . . . , &agr;
P
are transformed, for example, to LSP parameters or k parameters and then quantized in a quantization part
13
B, by which is obtained an index In
1
indicating the spectral envelope of the predictive coefficients. In an LPC spectrum calculating part
13
C the spectral envelope of the input signal is calculated from the quantized predictive coefficients. The spectral envelope thus obtained is provided to a spectrum flattening or normalizing part
14
and a weighting factor calculating part
15
D.
In the spectrum normalizing part
14
the spectrum sample values from the orthogonal transform part
12
are each divided by the corresponding sample of the spectral envelope from the spectral envelope estimating part
13
(flattening or normalization), by which spectrum residual coefficients are provided. A residual-coefficient envelope estimating part
15
A further calculates a spectral residual-coefficient envelope of the spectrum residual coefficients and provides it to a residual-coefficient flattening or normalizing part
15
B and the weighting factor calculating part
15
D. At the same time, the residual-coefficient envelope estimating part
15
A calculates and outputs a vector quantization index In
2
of the spectrum residual-coefficient envelope. In the residual-coefficient normalizing part
15
B the spectrum residual coefficients from the spectrum normalizing part
14
are divided by the spectral residual-coefficient envelope to obtain spectral fine structure coefficients, which are provided to a weighted vector quantization part
15
C. In the weighting factor calculating part
15
D the spectral residual-coefficient envelope from the residual-coefficient envelope estimating part
15
A and the LPC spectral envelope from the spectral envelope estimating part
13
are multiplied for each corresponding spectrum sample to obtain weighting factors W=w
1
, . . . , w
N
, which are provided to the weighted vector quantization part
15
C. It is also possible to use, as the weighting factors W, coefficients obtained by multiplying the multiplied results by psychoacoustic or perceptual coefficients based on psychoacoustic or perceptual models. In the weighted vector quantization part
15
C the weighted factors W are used to perform weighted vector quantization of the fine structure coefficients from the residual coefficient normalizing part
15
B. And the weighted vector quantization part
15
C outputs an index In
3
of this weighted vector quantization. A set of thus obtained indexes In
1
, In
2
and In
3
is provided as the result of coding of one frame of the input acoustic signal
At the decoding side depicted in
FIG. 1B
, the spectral fine structure coefficients are decoded from the index In
3
in a vector quantization decoding part
21
A. In decoding parts
22
and
21
B the LPC spectral envelope and the spectral residual-coefficient envelope are decoded from the indexes In
1
and In
2
, respectively. A residual coefficient de-flattening or de-normalizing (inverse flattening or inverse normalizing) part
21
C multiplies the spectral residual coefficient envelope and the spectral fine structure coefficients for each corresponding spectrum sample to restore the spectral residual coefficients. A spectrum de-flattening or de-normalizing (inverse flattening or inverse normalizing) part
25
multiplies the thus restored spectrum residual coefficients by the decoded LPC spectral envelope to restore the spectrum sample values of the acoustic signal. In an orthogonal inverse transform part
26
the spectrum sample values undergo orthogonal inverse transform into time-domain signals, which are provided as decoded acoustic signals of one frame at a terminal
27
.
In the case of coding input signals of plural channels through the use of such coding and decoding methods described in the afore-mentioned Japanese patent application laid-open gazette, the input signal of each channel is coded into the set of indexes In
1
, In
2
and In
3
as referred to above. It is possible to reduce combined distortion by controlling the bit allocation for coding in accordance with unbalanced power distribution among channels. In the case of stereo signals, there has already come into use, under the name of MS stereo, a scheme that utilizes the imbalance in power between right and left signals by transforming them into sum and difference signals.
The MS stereo scheme is effective when the right and left signals are closely analogous to each other, but it does not sufficiently reduce the quantization distortion when they are out of phase with each other. Thus the conventional method cannot adaptively utilize correlation characteristics of the right and left signals. Furthermore, there has not been proposed an idea of multichannel signal coding through utilization of the correlation between multichannel signals when they are unrelated to each other.
SUMMARY OF THE INVENTION
It is therefore an object of the present invention to provide a coding method that provides improved signal quality through reduction of the quantization distortion in the coding of multichannel input signals such as stereo signals, a decoding method therefor and coding and decoding devices using the methods.
The multichannel acoustic signal coding method according to the present invention comprises the steps of:
(a) interleaving acoustic signal sample sequences of plural channels under certain rules into a one-dimensional signal sequence; and
(b) coding the one-dimensional signal sequence through utilization of the c
Ikeda Kazunaga
Iwakami Naoki
Mori Takeshi
Moriya Takehiro
Nippon Telegraph and Telephone Corporation
Opsasnick Michael N.
Tsang Fan
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