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Showing papers on "Quadrature mirror filter published in 1980"


Proceedings ArticleDOI
James D. Johnston1
09 Apr 1980
TL;DR: This paper discusses a family of filters that have been designed for Quadrature Mirror Filter (QMF) Banks that provide a significant improvement over conventional optimal equiripple and window designs when used in QMF banks.
Abstract: This paper discusses a family of filters that have been designed for Quadrature Mirror Filter (QMF) Banks. These filters provide a significant improvement over conventional optimal equiripple and window designs when used in QMF banks. The performance criterion for these filters differ from those usually used for filter design in a way which makes the usual filter design techniques difficult to apply. Two filters are actually designed simultaneously, with constraints on the stop band rejection, transition band width, and pass and transition band performance of the QMF filter structure made from those filters. Unlike most filter design problems, the behavior of the transition band is constrained, which places unusual requirements on the design algorithm. The requirement that the overall passband behavior of the QMF bank be constrained (which is a function of the passband and stop band behavior of the filter) also places very unusual requirements on the filter design. The filters were designed using a Hooke and Jeaves optimization routine with a Hanning window prototype. Theoretical results suggest that exactly flat frequency designs cannot be created for filter lengths greater than 2, however, using the discussed procedure, one can obtain QMF banks with as little as ±.0015dB ripple in their frequency response. Due to the nature of QMF filter applications, a small set of filters can be derived which will fit most applications.

724 citations


Proceedings ArticleDOI
01 Apr 1980
TL;DR: The development of a digital encoding system designed to exploit the limited detection ability of the auditory system is described, dynamically shaping the encoding error spectrum as a function of the input speech signal, the error is masked by the speech.
Abstract: The development of a digital encoding system designed to exploit the limited detection ability of the auditory system is described. By dynamically shaping the encoding error spectrum as a function of the input speech signal, the error is masked by the speech. Psychoacoustic experiments and results from the literature provide a basis for determining the system parameters that ensure that the error is inaudible. The encoder is a multi-channel system, each channel approximately of critical bandwidth. The input signal is filtered into 17 frequency channels via the quadrature mirror filter technique. Each channel is then coded using block-companding adaptive PCM. For 4.1 kHz bandwidth speech, the differential threshold of the encoding degradation occurs at a bit rate of 34.4 kbps. At 16 kbps, the encoder produces toll quality speech output.

103 citations


Journal ArticleDOI
D. Behar1, H. Olaisen1, Gordon S. Kino1, D. Corl1, Peter Grant1 
TL;DR: A real-time deconvolution or inverse filter, operating at signal frequencies up to 5 MHz, is reported, which can be clearly discriminated after passing through a distorting medium.
Abstract: A real-time deconvolution or inverse filter, operating at signal frequencies up to 5 MHz, is reported. The programmable digital filter is controlled by a computer which calculates the Wiener-filter solution using f.f.t. techniques. Deconvolved signals can be clearly discriminated after passing through a distorting medium.

9 citations