Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission
Reexamination Certificate
1999-07-23
2004-11-30
Knepper, David D. (Department: 2654)
Data processing: speech signal processing, linguistics, language
Speech signal processing
For storage or transmission
Reexamination Certificate
active
06826526
ABSTRACT:
TECHNICAL FIELD
The present invention relates to coding apparatuses and methods in which a feature quantity obtained from an audio signal such as a voice signal or a music signal, especially a signal obtained by transforming an audio signal from time-domain to frequency-domain using a method like orthogonal transformation, is efficiently coded so that it is expressed with fewer coded streams as compared with the original audio signal, and to decoding apparatuses and methods having a structure capable of decoding a high-quality and broad-band audio signal using all or only a portion of the coded streams which are coded signals.
Various methods for efficiently coding and decoding audio signals have been proposed. Especially for an audio signal having a frequency band exceeding 20 kHz such as a music signal, an MPEG audio method has been proposed in recent years. In the coding method represented by the MPEG method, a digital audio signal on the time axis is transformed to data on the frequency axis using orthogonal transform such as cosine transform, and data on the frequency axis are coded from auditively important data by using the auditive sensitivity characteristic of human beings, whereas auditively unimportant data and redundant data are not coded. In order to express an audio signal with a data quantity considerably smaller than the data quantity of the original digital signal, there is a coding method using a vector quantization method, such as TC-WVQ. The MPEG audio and the TC-WVQ are described in “ISO/IEC standard IS-11172-3” and “T. Moriya, H. Suga: An 8 Kbits transform coder for noisy channels, Proc. ICASSP 89, pp. 196-199”, respectively. Hereinafter, the structure of a conventional audio coding apparatus will be explained using FIG.
37
. In
FIG. 37
, reference numeral
1601
denotes an FFT unit which frequency-transforms an input signal,
1602
denotes an adaptive bit allocation calculating unit which codes a specific band of the frequency-transformed input signal,
1603
denotes a sub-band division unit which divides the input signal into plural bands,
1604
denotes a scale factor normalization unit which normalizes the plural band components, and
1605
denotes a scalar quantization unit.
A description is given of the operation. An input signal is input to the FFT unit
1601
and the sub-band division unit
1603
. In the FFT unit
1601
, the input signal is subjected to frequency transformation, and input to the adaptive bit allocation unit
1602
; In the adaptive bit allocation unit
1602
, how much data quantity is to be given to a specific band component is calculated on the basis of the minimum audible limit, which is defined according to the auditive characteristic of human beings and the masking characteristic, and the data quantity allocation for each band is coded as an index.
On the other hand, in the sub-band division unit
1603
, the input signal is divided into, for example, 32 bands, to be output. In the scale factor normalization unit
1604
, for each band component obtained in the sub-band division unit
1603
, normalization is carried out with a representative value. The normalized value is quantized as an index. In the scalar quantization unit
1605
, on the basis of the bit allocation calculated by the adaptive bit allocation calculating unit
1602
, the output from the scale factor normalization unit
1604
is scalar-quantized, and the quantized value is coded as an index.
Meanwhile, various methods of efficiently coding an acoustic signal have been proposed. Especially in recent years, a signal having a frequency band of about 20 kHz, such as a music signal, is coded using the MPEG audio method or the like. In the methods represented by the MPEG method, a digital audio signal on the time axis is transformed to the frequency axis using orthogonal transform, and data on the frequency axis are given data quantities, with a priority to auditively important one, while considering the auditive sensitivity characteristic of human beings. In order to express a signal having a data quantity considerably smaller than the data quantity of the original digital signal, employed is a coding method using a vector quantization method, such as TCWVQ (Transform Coding for Weighted Vector Quantization). The MPEG audio and the TCWVQ are described in “ISO/IEC standard IS-11172-3” and “T. Moriya, H. Suga: An 8 Kbits transform coder for noisy channels, Proc. ICASSP 89, pp. 196-199”, respectively.
In the conventional audio signal coding apparatus constructed as described above, it is general that the MPEG audio method is used so that coding is carried out with a data quantity of 64000 bits/sec for each channel. With a data quantity smaller than this, the reproducible frequency band width and the subjective quality of decoded audio signal are sometimes degraded considerably. The reason is as follows. As in the example shown in
FIG. 37
, the coded data are roughly divided into three main parts, i.e., the bit allocation, the band representative value, and the quantized value. So, when the compression ratio is high, a sufficient data quantity is not allocated to the quantized value. Further, in the conventional audio signal coding apparatus, it is general that a coder and a decoder are constructed with the data quantity to be coded and the data quantity to be decoded being equal to each other. For example, in a method where a data quantity of 128000 bits/sec is coded, a data quantity of 128000 bits is decoded in the decoder.
However, in the conventional audio signal coding and decoding apparatuses, coding and decoding must be carried out with a fixed data quantity to obtain a good sound quality and, therefore, it is impossible to obtain a high-quality sound at a high compression ratio.
The present invention is made to solve the above-mentioned problems and has for its object to provide audio signal coding and decoding apparatuses, and audio signal coding and decoding methods, in which a high quality and a broad reproduction frequency band are obtained even when coding and decoding are carried out with a small data quantity and, further, the data quantity in the coding and decoding can be variable, not fixed.
Furthermore, in the conventional audio signal coding apparatus, quantization is carried out by outputting a code index corresponding to a code that provides a minimum auditive distance between each code possessed by a code block and an audio feature vector. However, when the number of codes possessed by the code book is large, the calculation amount significantly increases when retrieving an optimum code. Further, when the data quantity possessed by the code book is large, a large quantity of memory is required when the coding apparatus is constructed by hardware, and this is uneconomical. Further, on the receiving end, retrieval and memory quantity corresponding to the code indices are required.
The present invention is made to solve the above-mentioned problems and has for its object to provide an audio signal coding apparatus that reduces the number of times of code retrieval, and efficiently quantizes an audio signal with a code book having a lower number of codes, and an audio signal decoding apparatus that can decode the audio signal.
DISCLOSURE OF THE INVENTION
An audio signal coding method according to the present invention is a method for coding a data quantity by vector quantization using a multiple-stage quantization method comprising a first-stage vector quantization process for vector-quantizing a frequency characteristic signal sequence which is obtained by frequency transformation of an input audio signal, and second-and-onward-stages of vector quantization processes for vector-quantizing a quantization error component in the previous-stage vector quantization process: wherein, among the multiple stages of quantization processes according to the multiple-stage quantization method, at least one vector quantization process performs vector quantization using, as weighting coefficients for quantization, weighting coefficients on frequency, calculated on
Ishikawa Tomokazu
Miyasaka Shuji
Nakato Yoshihisa
Norimatsu Takeshi
Tsushima Mineo
Knepper David D.
Matsushita Electric - Industrial Co., Ltd.
Wenderoth , Lind & Ponack, L.L.P.
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