Showing papers by "David A. Johns published in 1993"
TL;DR: It is shown that latency and computational complexity can be reduced in some VLSI applications where digital representations of analog signals exist using oversampled Delta Sigma converters.
Abstract: Design techniques for IIR filters operating on oversampled delta-sigma ( Delta Sigma ) modulated signals are presented. It is shown that Delta Sigma -based IIR filters can be efficiently realized by eliminating all multibit multipliers through the use of remodulating internal filter states. Noise results are presented showing that linear noise analysis gives excellent predictions of the noise performance over the frequency band of interest. It is also shown that latency and computational complexity can be reduced in some VLSI applications where digital representations of analog signals exist using oversampled Delta Sigma converters. >
101 citations
TL;DR: Although a high speed sample-and-hold circuit is still required for an analogue-to-digital convertor, speed constraints are significantly reduced for the majority of analogue parts such as loop filters, A/D and D/A blocks.
Abstract: A new architecture is proposed which exploits the time-interleaving concept to increase the oversampling ratio in delta-sigma modulators. It is shown that the effective oversampling ratio is increased by a factor M through the use of M interconnected modulators. Although a high speed sample-and-hold circuit is still required for an analogue-to-digital convertor, speed constraints are significantly reduced for the majority of analogue parts such as loop filters, A/D and D/A blocks.
60 citations
03 May 1993
TL;DR: Experimental results indicate an effective dynamic range of approximately 70 dB, equaling the behavior of a Hewlett-Packard HP3254A high-performance signal generator.
Abstract: A high-quality analog oscillator designed specifically for audio applications is described. Except for a rather imprecise low-pass filter and a 1-b digital-analog converter, the proposed circuit is entirely digital, providing accurate control over its oscillation frequency and amplitude. Experimental results indicate an effective dynamic range of approximately 70 dB, equaling the behavior of a Hewlett-Packard HP3254A high-performance signal generator. >
26 citations
TL;DR: In this article, the performance of an adaptive tuning system on tuning a continuous-time filter with non-ideal integrators is compared to that of the popular approach of tuning only the poles of a transfer function.
Abstract: The parasitic effects and tuning limitations that nonideal integrators have on continuous-time filters are discussed. The nonideal integrator is modeled to have finite DC gain and a single nondominant pole. The performance of an adaptive tuning system on tuning a continuous-time filter with nonideal integrators is compared to that of the popular approach of tuning only the poles of a transfer function. Simulation results are presented showing that the adaptive tuning system significantly reduces parasitic effects. >
11 citations
03 May 1993
TL;DR: In this article, a theoretical framework based on the describing function method is derived which enables estimation of the maximum and the minimum level at the input of the comparator, as well as the period of limit cycles with respect to the input.
Abstract: Comparator hysteresis of a 1-bit quantizer in /spl Delta/-/spl Sigma/ interpolative modulators is considered. A theoretical framework based on the describing function method is derived which enables estimation of the maximum and the minimum level at the input of the comparator, as well as the period of limit cycles with respect to the input. The results are in good agreement with simulation results, showing that hysteresis has a twofold worsening effect, i.e., increasing noise power and shifting the noise spectrum toward the signal band. >
8 citations
Proceedings Article•
01 Jan 1993
8 citations
TL;DR: In this article, a fully differential BiCMOS operational transconductance amplifier (OTA) was proposed for high-frequency data communication applications where moderate SNR is acceptable, and the tuning scheme was based on current steering resulting in the maximum input signal level being independent of tuning.
5 citations
27 Apr 1993
TL;DR: These adaptive filters-based on a variant of the familiar LMS (least mean square) algorithm-utilize only simple hardware components such as adders and registers, making the scheme hardware-efficient and a viable alternative to conventional, Nyquist-rate LMS adaptive filters.
Abstract: A novel method for constructing adaptive recursive digital filters is presented. The proposed filters operate at the oversampled rate directly on the bitstream produced by delta-sigma analog-to-digital converters. Thus, these filters obviate the need for the decimation and interpolation circuitry required by Nyquist-rate filters. In addition, these adaptive filters-based on a variant of the familiar LMS (least mean square) algorithm-utilize only simple hardware components such as adders and registers, making the scheme hardware-efficient and a viable alternative to conventional, Nyquist-rate LMS adaptive filters. >
3 citations
03 May 1993
TL;DR: Design equations and experimental results for a second-order delta-sigma (/spl Sigma/-/spl Delta/) based infinite impulse response (IIR) filter are presented and it is shown experimentally that such a filtering technique can realize filter pole-Q's of at least 2600 in the audio frequency range.
Abstract: Design equations and experimental results for a second-order delta-sigma (/spl Sigma/-/spl Delta/) based infinite impulse response (IIR) filter are presented. Although input and output signals are analog, filtering is performed directly on an oversampled /spl Delta/-/spl Sigma/ modulated signal, thus eliminating the need for any decimation or interpolation filters. Circuit complexity consists of only a single /spl Delta/-/spl Sigma/ analog-digital (A/D) modulator, approximately ten 26-b adders, a 1-b digital-analog converter (D/A) and a low-pass filter. It is shown experimentally that such a filtering technique can realize filter pole-Q's of at least 2600 in the audio frequency range. >
3 citations
03 May 1993
TL;DR: The authors present a method for equalizing the frequency response of a weakly nonlinear system while simultaneously reducing the amount of nonlinear distortion by introducing a linear equalizer in a feedback loop with the system.
Abstract: The authors present a method for equalizing the frequency response of a weakly nonlinear system while simultaneously reducing the amount of nonlinear distortion. This is achieved by introducing a linear equalizer in a feedback loop with the system. A dynamic loudspeaker is used to illustrate the theory and its mechanics. Simulation results are presented which show an improvement in frequency response and a reduction in nonlinear distortion. >