The basic concepts and building blocks in multirate digital signal processing (DSP), including the digital polyphase representation, are reviewed. Recent progress, as reported by several authors in this area, is discussed. Several applications are described, including subband coding of waveforms, voice privacy systems, integral and fractional sampling rate conversion (such as in digital audio), digital crossover networks, and multirate coding of narrowband filter coefficients. The M-band quadrature mirror filter (QMF) bank is discussed in considerable detail, including an analysis of various errors and imperfections. Recent techniques for perfect signal reconstruction in such systems are reviewed. The connection between QMF banks and other related topics, such as block digital filtering and periodically time-varying systems, is examined in a pseudo-circulant-matrix framework. Unconventional applications of the polyphase concept are discussed. >

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Multirate digital filters, filter banks, polyphase networks, and applications: a tutorial

A 4-b/sample transform coder is designed using a psychoacoustically derived noise-making threshold that is based on the short-term spectrum of the signal. The coder has been tested in a formal subjective test involving a wide selection of monophonic audio inputs. The signals used in the test were of 15-kHz bandwidth, sampled at 32 kHz. The bit rate of the resulting coder was 128 kb/s. The subjective test shows that the coded signal could not be distinguished from the original at that bit rate. Subsequent informal work suggests that a bit rate of 96 kb/s may maintain transparency for the set of inputs used in the test. >

/pdf/transform-coding-of-audio-signals-using-perceptual-noise-2yjpl6gajw.pdf

Transform coding of audio signals using perceptual noise criteria

In this paper, quadrature mirror filters (QMF) are reviewed. After a brief introduction to multirate building blocks, the two-band QMF bank is discussed. Various distortions caused by the structure, and methods to eliminate these distortions are outlined. Perfect-reconstruction structures for the two-band case are reviewed, and the results are extended to the case of arbitrary number of channels. The relation between perfect-reconstruction QMF banks and the concept of losslessness in transfers-matrices is indicated. New lattice structures are presented, which perform the perfect reconstruction, sometimes even under coefficient quantization.

/pdf/quadrature-mirror-filter-banks-m-band-extensions-and-perfect-2551h0jccp.pdf

Quadrature mirror filter banks, M-band extensions and perfect-reconstruction techniques

A method and modem for communicating serial input data over a transmission link. Serial input data is partitioned into parallel data elements prior to rotation by an invertible linear mapping. Resulting frames of parallel signal elements sequentially modulate a carrier, which is then transmitted over the link. After receipt of the modulated carrier from the link, the signal is demodulated and assembled into frames of parallel signal elements which are de-rotated by an inverse linear mapping. Thresholding the result of the inverse mapping recovers the parallel data elements, which are then re-assembled into serial output data. The linear mapping employs: 1) commuting rotation matrices for convolutionally rotating data vectors into signal vectors and vice-versa; 2) filter bank polyphase rotation matrices; or 3) computationally efficient multi-rate wavelet filter banks. Transmitter pre-emphasis places most of the information in lower baseband frequencies; complimentary de-emphasis occurs in the receiver. Logarithmic amplification of the baseband signal prior to carrier modulation improves modulation gain and transmit channel noise attenuation. Coefficients of the rotation matrix of the receiver are adaptively equalized to correct for transmission path distortion. FM double-side band is employed in systems requiring minimized cost and complexity. FM single-side band is employed in systems in which bandwidth reduction is desirable. AM is also employable.

Method and apparatus for signal transmission and reception

This paper presents design algorithms for hybrid filter banks (HFBs) for high-speed, high-resolution conversion between analog and digital signals. The HFB is an unconventional class of filter bank that employs both analog and digital filters. When used in conjunction with an array of slower speed converters, the HFB improves the speed and resolution of the conversion compared with the standard time-interleaved array conversion technique. The analog and digital filters in the HFB must be designed so that they adequately isolate the channels and do not introduce reconstruction errors that limit the resolution of the system. To design continuous-time analog filters for HFBs, a discrete-time-to-continuous-time ("Z-to-S") transform is developed to convert a perfect reconstruction (PR) discrete-time filter bank into a near-PR HFB; a computationally efficient algorithm based on the fast Fourier transform (FFT) is developed to design the digital filters for HFBs. A two-channel HFB is designed with sixth-order continuous-time analog filters and length 64 FIR digital filters that yield -86 dB average aliasing error. To design discrete-time analog filters (e.g., switched-capacitors or charge-coupled devices) for HFBs, a lossless factorization of a PR discrete-time filter bank is used so that the reconstruction error is not affected by filter coefficient quantization. A gain normalization technique is developed to maximize the dynamic range in the finite-precision implementation. A four-channel HFB is designed with 9-bit (integer) filter coefficients. With internal precision limited to the equivalent of 15 bits, the maximum aliasing error is -70 dB, and with the equivalent of 20 bits internal precision, maximum aliasing is -100 dB. The 9-bit filter coefficients degrade the stopband attenuation (compared with unquantized coefficients) by less than 3 dB.

http://circuit.ucsd.edu/~nguyent/Nguyen/00668549.pdf

Design of hybrid filter banks for analog/digital conversion

The Analog Voice Privacy System is based on individual sample permutation of the output samples of a sub-band coder analysis filterbank. The system has a large number of digital keys, giving it the strength of a digital encryption system, but also retains the good quality characteristics of analog scramblers. It has been implemented in a real-time hardware prototype designed for evaluation in the field. The units work with any modular telephone and standard 120 volts AC electricity. The device contains two circuitry boards, one for analog and one for digital processing which contain four digital signal processors. There are 125! possible permutation keys. These prototypes were designed to be tested in real telephone environments. To date, the device has been successfully tested over long distance telephone connections, several different analog and digital PBXs and telephone switches, and a channel simulator. The quality of the decrypted speech is considered very natural, and in particular, speaker recognition is retained. This is a significant advantage over digital vocoders. This paper describes the underlying principles of the algorithm, the details of its implementation and laboratory test results.

The analog voice privacy system

Development of Circuit Switched Digital Capability (CSDC) in the United States and the general evolution toward an integrated digital services network worldwide has created interest in the development of 56 or 64 kbps digital coders for audio signals with 6 or 7 kHz bandwidth. To date there have been a number of coders proposed but no formal testing of them. Furthermore, a method for testing them has not been proposed. In this paper we discuss the problem of testing such coders and describe a series of both objective and subjective tests which can be performed using a real-time coding system. We discuss the results of these tests when performed on a selection of coders which we have implemented.

D. Bock

Papers

The analog voice privacy system

Testing of wideband digital coders