Data processing: speech signal processing – linguistics – language – Audio signal bandwidth compression or expansion
Patent
1995-11-20
1999-02-16
Hudspeth, David R.
Data processing: speech signal processing, linguistics, language
Audio signal bandwidth compression or expansion
381 2, 704227, 704229, H04S 300, H04B 166, H04H 500
Patent
active
058730656
DESCRIPTION:
BRIEF SUMMARY
FIELD OF THE INVENTION
This invention relates to a method and apparatus for compressing the sound signals of the multi-channel sound system of, for example, a motion picture theater, a video tape recorder, or a video disc player, a method for transmitting the compressed sound signals of a multi-channel sound system, a method and apparatus for expanding the compressed sound signals of a multi-channel sound system, and a recording medium on which the compressed sound signals of a multi-channel sound system are recorded.
BACKGROUND OF THE INVENTION
Many techniques for compressing digital audio or speech signals are known. For example, in sub-band coding, a non block-forming frequency band dividing system, in which the input audio signal is not divided in time into blocks, but is divided in frequency by a filter into plural frequency bands for quantizing. In a block-forming frequency band dividing system, such as a transform coding system, the input audio signal in the time domain is converted into spectral coefficients in the frequency domain by an orthogonal transform. The resulting spectral coefficients are divided by frequency into plural frequency bands, and the spectral coefficients in each band are quantized.
A technique consisting of a combination of sub-band coding and transform coding is also known. In this, frequency range signals produced by dividing the input audio signal in frequency without dividing it into blocks are individually orthogonally transformed into spectral coefficients. The spectral coefficients are then divided by frequency into plural frequency bands, and the spectral coefficients in each band are then quantized.
Among the filters useful for dividing a digital audio input signal into frequency ranges without dividing it into blocks is the quadrature mirror (QMF) filter, which is described, for example, in R. E. Crochiere, Digital Coding of Speech in Sub-bands, 55 BELL SYST. TECH. J. No.8, (1976). A technique of dividing the audio input signal in frequency into frequency bands of an equal width is discussed in Joseph H. Rothweiler, Polyphase Quadrature Filers-a New Sub-band Coding Technique, ICASSP 83, BOSTON (1983).
Known techniques for orthogonally transforming an input signal include the technique of dividing the digital input audio signal in time into blocks having a predetermined duration, and processing the resulting blocks using a fast Fourier transform (FFT), a discrete cosine transform (DCT), or a modified DCT (MDCT) to convert each block of the digital audio signal in the time domain into a set of spectral coefficients in the frequency domain. A modified DCT is discussed in J. P. Princen and A. B. Bradley, Subband/Transform Coding Using Filter Bank Based on Time Domain Aliasing Cancellation, ICASSP 1987.
As a technique for quantizing the spectral coefficients obtained by frequency division, it is known to divide the spectral coefficients by frequency into bands to take account of the frequency resolution characteristics of the human sense of hearing. The audio frequency range of 0 Hz to 20 or 22 kHz may be divided in frequency into bands, such as 25 critical bands, which have a bandwidth that increases with increasing frequency. The spectral coefficients in each of the bands are quantized by adaptive bit allocation applied to each band. For example, the spectral coefficients resulting from a modified discrete cosine transform (MDCT) are divided by frequency into bands, and the spectral coefficients in each band are quantized using an adaptively-determined number of bits.
Two known adaptive bit allocation techniques will be now be described. First, in the technique described in ASSP-25, IEEE TRANSACTIONS OF ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, No.4, August 1977, bit allocation is carried out on the basis of the magnitude of the signals of the respective bands. Although this system provides a flat quantizing noise spectrum, and minimizes noise energy, noise perceived by the listener is not minimized because this technique does not exploit the masking characteristic
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Akagiri Kenzo
Davis Mark Franklin
Dolby Ray Milton
Todd Craig Campbell
Hudspeth David R.
Smits Talivaldis Ivars
Sony Corporation
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