Coded data generation or conversion – Digital code to digital code converters – To or from particular bit symbol
Patent
1995-09-27
1998-05-19
Gaffin, Jeffrey A.
Coded data generation or conversion
Digital code to digital code converters
To or from particular bit symbol
341 50, 395 2, 395 38, H03M 522
Patent
active
057541272
DESCRIPTION:
BRIEF SUMMARY
TECHNICAL FIELD
This invention relates to an information encoding method and an information encoding apparatus, and an information decoding method and an information decoding apparatus using so called efficient encoding/decoding, and more particularly to operation of digital filter of so called QMF (Quadrature Mirror Filter) for halving the band with respect to an input signal and IQMF (Inverse QMF) for synthesizing the bands halved by such QMF, and a processing of so called MDCT and IMDCT for synthesizing signals transformed by such MDCT.
BACKGROUND ART
Hitherto, there have been proposed various efficient encoding techniques for audio signals or speech signals, etc. For example, there can be enumerated transform coding which is the blocking frequency band division system of carrying out blocking of a signal on the time base in predetermined time units to transform (spectrum-transform) the signal on the time base every block into a signal on the frequency base to divide it into signal components in a plurality of frequency bands to encode them every respective bands, and Sub Band Coding (SBC) which is the non-blocking frequency band division system of dividing, an audio signal, etc. on the time base into signal components in a plurality of frequency bands without carrying out blocking to encode them, etc. Moreover, there is also proposed a technique in which the Sub Band Coding (SBC) and the transform coding described-above are combined. In this case, for example, an approach is employed to carry out band division by the above-mentioned Sub Band Coding thereafter to spectrum-transform signals every respective bands into signals on the frequency base to encode the spectrum-transformed signals every respective bands.
Here, as a filter for band division used in the above-described Sub Band Coding (band division encoding), there is a filter, e.g., QMF, etc. This filter of QMF is described in the literature "Digital coding of speech in subbands" R. E. Crochiere, Bell Syst. Tech. J., Vol. 55, No. 8 1976.
Moreover, in the literature "Polyphase Quadrature filters-A new subband coding technique", Joseph H. Rothweiler ICASSP 83, BOSTON, filter division technique of equal bandwidth is described.
As an algorithm of calculation provides a more practical sense of the QMF, there is, e.g., an algorithm as described below.
When coefficients of QMF having the number of taps of L are assumed to be h(0), h(1), . . . , h(L-1), h(j)=h(L-j-1) holds with respect to j in the range expressed as L/2.ltoreq.j<L from the property of QMF. It is now assumed that sampling in a lower frequency band to which QMF is implemented to an input signal x(n) is z.sub.1 (n), and sampling in a higher frequency band is Z.sub.2 (n). When the following formula (1) is calculated in the case where n is an even number, and the following formula (2) is calculated in the case where n is an odd number, z.sub.1 (n) and Z.sub.2 (n) are respectively expressed as the following formulas (3) and (4): ##EQU1## Since bands are respectively limited to one half with respect to z.sub.1 (n), z.sub.2 (n), odd numbered samples can be subjected to thinning, and it is therefore unnecessary to calculate them.
Moreover, in the case of IQMF (Inverse QMF) which is inverse filter of QMF, when it is assumed that an input signal in a lower frequency band is y.sub.1 (m) and an input signal in a higher frequency band is y.sub.2 (m), output s(n) (n=2m and n=2m+1) obtained by synthesizing these bands is expressed as the following formulas (5) and (6) when n is equal to 2m, and is expressed as the following formulas (7), (8) when n is equal to 2m+1. ##EQU2##
Further, as the above-described spectrum transform processing, there is, e.g., such a spectrum transform processing to carry out blocking of an input audio signal every predetermined unit of time (frame) to carry out, for every respective blocks, Discrete Fourier Transform (DFT), Discrete Cosine Transform (DCT), or Modified DCT (MDCT), etc. to thereby transform signals on the time base into signals on the frequency base. The
REFERENCES:
patent: 5375189 (1994-12-01), Tsutsui
Sonohara Mito
Tsutsui Kyoya
Gaffin Jeffrey A.
Kost Jason L. W.
Sony Corporation
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