Electrical computers: arithmetic processing and calculating – Electrical digital calculating computer – Particular function performed
Reexamination Certificate
1998-01-26
2001-10-02
Malzahn, David H. (Department: 2121)
Electrical computers: arithmetic processing and calculating
Electrical digital calculating computer
Particular function performed
C708S300000
Reexamination Certificate
active
06298361
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Field of the Invention
This invention relates to a band synthesis filter bank and a band synthesis filtering for band-synthesizing band-split signals for restoration of the original signals, a band splitting filter bank and a band splitting filtering for band-splitting signals, an encoder and an encoding method, a decoder and a decoding method, and a recording medium.
2. Description of the Related Art
There are a wide variety of methods for encoding signals, such as audio or acoustic signals. Examples of these are a sub-band encoding method, which is a non-blocking frequency spectrum splitting method of splitting time-domain audio signals into plural frequency bands without blocking, using a spectrum-splitting filter, and encoding the resulting signals of the frequency bands, and a so-called transform coding which is a blocking frequency spectrum splitting method of transforming time-domain signals into frequency domain signals by orthogonal transform and encoding the resulting spectral components from one frequency band to another.
There is also known a high-efficiency encoding technique which is a combination of the sub-band coding and transform coding, in which case time domain signals are split into plural frequency bands by sub-band coding and the resulting band signals are orthogonal-transformed into spectral components which are encoded from band to band.
Among the above-mentioned filters is a so-called QMF (quadrature mirror filter) as discussed in 1976, R. E. Crochiere, Digital Coding of Speech in Subbands, Bell Syst. Tech. J. Vol. 55, No. 8, 1976. However, if the frequency spectrum is split into a number of bands, a corresponding number of QMF stages is required, thus increasing the processing volume and consequent signal delay.
Thus, attention is being directed to PQF (polyphase quadrature filter) suffering from only small processing volume and delay and which can split the frequency spectrum into M equal bands such that the number of samples of orthogonal transform associated with the same time point become equal in all bands to simplify the filter bank structure to enable efficient processing. This PQF is discussed in Joseph H. Rothweiler, Polyphase Quadrature Filters—A New Subband Coding Technique, ICASSP 83 BOSTON.
Among the above-mentioned techniques for orthogonal transform is a technique in which an input audio signal is blocked every pre-set unit time, such as every frame, and discrete Fourier transform (DFT), discrete cosine transform (DCT) or modified DCT (MDCT) is applied to each block for converting the signals from those on the time axis to those on the frequency axis. If the above-mentioned DFT or DCT is applied on the basis of a time block composed of N samples, N independent real-number data are obtained. It is noted that, for reducing junction distortions between time blocks, a given time bock is usually overlapped with N
1
samples with both neighboring blocks, and N real-number data on an average are quantized and encoded in DFT or DCT for (N-N
1
) samples.
On the other hand, if the above-mentioned MDCT is used as a method for orthogonal transform, N independent real-number data are obtained from 2N samples overlapped with N samples of both neighboring time blocks. Thus, in MDCT, N real-number data on an average are quantized and encoded for N samples. A decoding device adds waveform elements obtained on inverse transform in each block from the codes obtained by MDCT with interference for reconstructing the waveform signals. Discussions of the MDCT are found in J. P. Princen and A. B. Bradley, Subband/Transform Coding Using Filter Bank Based on Time Domain Aliasing Cancellation, ICASSP 1987. By quantizing signals split into plural frequency bands by a filter or orthogonal transform, the frequency band in which occurs the quantization noise can be controlled so that encoding can be achieved with psychoacoustic higher efficiency by exploiting acoustic characteristics such as masking effects or equal loudness effects.
In high-efficiency encoding/decoding acoustic signals, the frequency spectrum is split in many cases into several bands. In such case, band-splitting/band synthesis of high precision is desirable to remove the effect from the neighboring bands. To this end, a band splitting/band synthesizing filter with as long a tap length as possible and with acute frequency response is typically used. This, however, increases the processing volume to render real-time processing difficult.
In particular, it is necessary to optimize the bit allocation for the spectra components, thus significantly increasing the processing volume. Notwithstanding, an encoder of a larger size and a larger processing capability can be used in many cases, so that the tap length of the frequency-splitting filter bank used for an encoder can be selected to a proper length matched to the desired characteristics. On the other hand, it is not mandatory with an encoder to perform real-time processing.
However, if desired to design an encoder of a small size and a low processing capability, it becomes necessary to suppress the processing volume to a lower value, even although the encoder can use a device of a larger size and a higher processing capability.
In a decoder not in need of an optimizing processing as required for an encoder, a device of a smaller size is desired with a smaller processing volume than that required in an encoder, so that a lower processing volume is desirable. Recently, a decoder is produced by software in many cases. Since the software is lower in processing capability than the hardware, it becomes more crucial to suppress the processing volume to a smaller value.
SUMMARY OF THE INVENTION
It is therefore an object of the present invention to provide a band synthesizing filter bank and a band synthesizing filtering method capable of reducing the processing volume to realize decoding with a low processing volume with priority placed on real-time processing in case it is necessary to use a decoder of low processing capability in carrying out band synthesis processing.
It is another object of the present invention to provide a band-splitting filter bank and a band-splitting filtering method capable of reducing the processing volume to realize encoding with priority placed on real-time processing if, in performing band-splitting, an encoder with a low processing capability needs to be used.
It is still another object of the present invention to provide an encoder in which the processing volume can be controlled to place a priority on the real-time processing if, in performing encoding accompanied by band splitting, the processing capability is low or if the processing capability is changed with time.
It is yet another object of the present invention to provide a decoder in which the processing volume can be controlled to place a priority on the real-time processing if, in performing encoding accompanying band-synthesis processing, the processing capability is low or if the processing capability is changed with time.
In one aspect, the present invention provides a band-splitting filter bank for processing an input signal with band-splitting, including tap length converting means for outputting filter coefficients with a contracted tap length obtained on contracting the filter coefficients of a standard tap length based on the tap length information and processing means for convolving the filter coefficients of the contracted tap length outputted by the tap length converting means with the input signal.
In another aspect, the present invention provides a band-splitting filtering method for band-splitting an input signal including a tap length converting step for outputting filter coefficients with a contracted tap length obtained on contracting the filter coefficients of a standard tap length based on the tap length information and a processing step for convolving the filter coefficients of the contracted tap length outputted by the tap length converting step with the input signal.
In a further aspect, the present invention pro
Malzahn David H.
Sonnenschein Nath & Rosenthal
Sony Corporation
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