Showing papers on "Quadrature mirror filter published in 1985"

Filters for distortion-free two-band multirate filter banks

Abstract: In this paper, conditions are given for a two-band multi-rate filter bank to be alias free and to have a unity frequency response. It is shown that the class of quadrature mirror filters (QMF's) that satisfies these conditions is quite limited. A class of filters which does satisfy these conditions is given, and a simple procedure for designing filters from this class is presented with an example.

Quadrature mirror filter design for an arbitrary number of equal bandwidth channels

Abstract: The two-channel quadrature mirror filter structure of Croisier and Esteban may be extended to an arbitrary number of equal bandwidth channels, given certain restrictions on the bandpass filters. The most serious restriction is that the stopband attenuation of eacli band-pass filter must be high for all frequencies outside twice the nominal 3 dB bandwidth of the filter. This restriction is not really a limiting factor for speech subband waveform coding since high adjacent channel attenuation is a necessity for the confinement of quantization noise. A key feature of our filter structure is that the number of multiplies, adds, and stored coefficients required for implementation is significantly less than those needed for the conventional QMF structure, given the same number of channels. Fortran code for a 16-channel filter structure is listed as an example of efficient implementation.

Flexible design of computationaly efficient nearly perfect QMF filter banks

Abstract: This paper introduces a flexible design method of computationally efficient uniform filter bank where each filter channel is obtained by frequency translation of a prototype base-band filter. It is shown that the original signal decomposed by the analysis filter bank into N adjacent uniform subbands subsampled by N, can be reconstructed with negligible distortion. We give a practical implementation which allows an important reduction in computational complexity. It can be therefore applied to such signal processing tasks as speech coding or spectral parametrization of signals.

Recursive quadrature mirror filters--Criteria specification and design method

Abstract: Criteria are found for a pair of recursive infinite impluse response (IIR) filters which have mirror image amplitude responses and whose outputs are in phase quadrature at all frequencies. A method is suggested for designing filters with these characteristics, and their performance and processing requirements are compared with a typical nonrecursive finite impluse response (FIR) realization. A particular filter is described in more detail, and it is shown how it could be incorporated in a multiband processing scheme. It is concluded that recursive quadrature mirror filters could offer considerable savings in terms of signal processing compared with nonrecursive filters with similar performances. Additionally, it is suggested that, because they can be designed to have relatively short absolute delays, the performance of the IIR filters may prove superior for certain applications.

Subband coding of speech using backward adaptive prediction and bit allocation

Abstract: In this study it is our goal to improve the performance of ADPCM and subband speech coders at medium bit rates (9.6∼16 kb/s) without increasing the coder complexity substantially. Various major building blocks including the predictors (both fixed and adaptive), subband quadrature mirror filter (QMF) and bit assignment strategy (both static and dynamic) are investigated in detail. We have found that (1) the Least-Squares (LS) adaptive lattice predictor outperforms both the pole-zero adaptive predictor recommended by CCITT and a first-order fixed predictor. (2) more subbands can improve the coder performance (3) longer QMF can reduce the interband aliasing and improve the subjective performance of a subband coder (4) an optimal dynamic bit allocation scheme with an improvement of SNR as high as 5 dB is much more favorable than a fixed bit allocation. With all of the above finding we propose a 4-band hybrid subband coder with an LS adaptive lattice predictor and an optimal dynamic bit allocation strategy.

Quadrature mirror filters with perfect reconstruction and reduced computational complexity

Abstract: This paper introduces a new quadrature mirror filter technique allowing to split a signal in sub-bands, with subsequent perfect reconstruction of this signal The analysis is performed by a tree structure based on a set of non-linear phase FIR filters that split the signal into N equally spaced sub-band signals sub-sampled by 1/N The synthesis makes use of a set of non-linear phase filters that are linear phase complementary to the analysis filters This new scheme replaces the shifted sampling technique used in an earlier approach, by a synchronous sampling technique, in such a way that the perfect reconstruction property is retained, with a reduced computational complexity

A novel approach to the design of filters for filter banks

Abstract: In this paper a design technique is given for filters with certain constraints on their transition bands. Filters designed by the proposed procedure can be used in conventional polyphase filter banks as well as in quadrature mirror filter (QMF) banks for a unity transfer function (besides a linear phase term) of a cascaded analysis/synthesis system. Additional design criteria can be met approximately as the given technique leads to a usually large number of solutions with different phase characteristics for the subsystems. The described method is an analytical one, whereas up to now filter designs for these purposes were mostly based on optimization procedures or iterative algorithms.

A Framework For Scaling Arbitrary Fixed Point Digital Filter Structures

Abstract: The precise modeling of a digital filter structure through expanded state equations is used as a basis for scaling the filter states and summers. The scaling process is structured to allow each state and summer full utilization of its dynamic range. The scaling process is user defined and includes, but is not limited to, 1 norms, peak magnitude filter operation, and ran8om noise. The measured dynamic range is then used to "normalize" the summer and state dynamic ranges by changing or adding filter multipliers. This scaling process does not change the basic filter structure. Ihe scaling gives an improved signal-to-noise ratio for the same filter dynamic range. The scaling process has been computer automated for an arbitrary filter.