Madihally J. Narasimha

On the Computation of the Discrete Cosine Transform

An N -point discrete Fourier transform (DFT) algorithm can be used to evaluate a discrete cosine transform by a simple rearrangement of the input data. This method is about two times faster compared to the conventional method which uses a 2N -point DFT.

An improved single frequency estimator

A new method for estimating the frequency of a complex sinusoid in complex white Gaussian noise is proposed. Its computational complexity is comparable to Kays method [1989], but it attains the Cramer-Rao bound (CRB) down to lower signal-to-noise ratio (SNR) values. Simulation results are included to demonstrate the performance of the proposed method.

The Batcher-banyan self-routing network: universality and simplification

It is shown that the Batcher-banyan network performs as a universal self-routing switch when inputs with unassigned destinations are present. This is demonstrated by first proving that banyan networks can realize permutations represented by bitonic sequences, and then noting that the sorted output of the Batcher network can be viewed as a bitonic sequence. Two methods are proposed for reducing the complexity of the Batcher-banyan network. In the first method, one stage of the banyan network is eliminated by assigning proper destination tags to the unassigned inputs. In the second, a self-routing switch based on the binary-radix sorting scheme is shown to be more economical for a small number of lines. >

A recursive concentrator structure with applications to self-routing switching networks

A technique for constructing an N-input concentrator from two (N/2)-input concentrators and a stage of 2/spl times/2 switching elements is described. Recursive decomposition of each of the (N/2)-input concentrators yields a log/sub 2/N-stage switching network similar in topology to the inverse omega network. The control algorithm for setting the switching elements is shown to be a simple logic function of the activity bits of the inputs. An iterative cell that combines the functions of the control logic and the 2/spl times/2 switching element can be used as a building block to realize the entire concentrator. Applications of the concentrator structure as an interconnection fabric in fast packet switching is explored. This leads to an efficient self-routing network, similar in topology to the transposed Batcher-banyan network. >

Modified Overlap-Add and Overlap-Save Convolution Algorithms for Real Signals

Modified overlap-save and overlap-add algorithms for filtering a real signal by an N-tap finite impulse response filter are delineated. These algorithms compute two N-sample output blocks simultaneously at every processing step using 2N-point transforms, while the traditional methods compute only one N-sample block with the same length transforms. They achieve streamline processing for real signals but incur an additional delay of N samples

Chromatic derivative filter banks

A new contribution to generalized sampling of bandlimited signals based on the so-called chromatic derivative operators was recently introduced by Ignjatovic (see Kromos Technology, Los Altos, CA. [Online] Tech. Rep. 1, 2001). Chromatic derivatives are linear combinations of the ordinary derivatives, where the coefficients of the combination are derived from orthogonal polynomial theory. This article describes the connection between these operators and the well-established ideas of perfect reconstruction and biorthogonality in analog filter banks.

Speech over VoIP Networks: Advanced Signal Processing and System Implementation

Speech communication using the Voice over Internet Protocol (VoIP) is very common today. The underlying network channel may be the public switched telephone network (PSTN channel), satellite channels or cellular wireless channels to name a few. The packetization of speech and its transmission through packet switched networks, however, introduce numerous impairments such as delay, jitter, packet loss and decoder clock offset, which degrade the quality of the speech. We present an overview of the challenges and a description of the advanced signal processing algorithms used to combat these impairments and render the perceived quality of a VoIP conversation to be as good as that of the existing telephone system. We also present an example of a speech coder designed for packet-switched networks and discuss the possibilities for hardware implementations.

Design and applications of uniform digital bandpass filter banks

A digital bandpass filter-bank for demodulating a wideband frequency multiplexed signal into a specified number of uniform narrow band channel outputs, and conversely for modulating a set of channel inputs into a composite frequency multiplexed signal, can be realized efficiently by the combination of a suitable transform processor and a weighting network. The design and implementation of these filter-banks for different types of input-ouput configurations such as complex signals at both the wideband and narrow band sides, real signal at the wideband side and complex signals at the narrow band side, and real signals at both ends are discussed. Applications in telecommunications (TDM-FDM translators), speech processing (phase vocoders), signal analysis and detection (spectrum analyzers) and other signal processing systems are also described.

Unbiased timing-error estimation in the presence of nonideal interpolation

We propose a method to eliminate the bias term present in the timing-error estimator employed in digital receivers where the input signal is sampled by a fixed clock which is not synchronized to the transmitter clock. This bias error results from the nonideal interpolation that precedes the timing-error estimator. We show that it can be derived as a function of the previously estimated symbol timings. An unbiased timing-error estimate can then be obtained by subtracting this bias term from the output of the timing-error detector. Simulation results are included to show the performance improvement realizable by employing this method.

Design of FIR Filter Banks for a 24-Channel Transmultiplexer

The design of a bank of FIR bandpass filters for channel filtering and sample rate alteration in a 24-channel transmultiplexer is considered in this paper. It is shown that such a filter bank can be realized by the combination of a weighting network and a discrete cosine transform. A minimum phase design is suggested for the lowpass prototype filter in order to solve the long absolute delay Problem inherent in linear phase filters. Two approaches to the design of long minimum phase filters are disclosed; one employs frequency sampling design techniques, and the other makes use of the properties of the complex cepstrum of a minimum phase sequence. Filter specifications for the low-pass prototype are discussed and a design example is included. The hardware realization of the filter bank using a multiplier-accumulator as the arithmetic element is also discussed.