Journal Article•
Splitting the Unit Delay - Tools for fractional delay filter design
About: This article is published in IEEE Signal Processing Magazine.The article was published on 1996-01-01 and is currently open access. It has received 686 citations till now. The article focuses on the topics: Prototype filter & Network synthesis filters.
Citations
More filters
07 Aug 2002
TL;DR: Digital calibration using adaptive signal processing corrects offset mismatch, gain mismatch, and sample-time error between time-interleaved channels in a 10b 120MSample/s pipelined ADC.
Abstract: Digital calibration using adaptive signal processing corrects for offset mismatch, gain mismatch, and sample-time error between time-interleaved channels in a 10-b 120-Msample/s pipelined analog-to-digital converter (ADC). Offset mismatch between channels is overcome with a random chopper-based offset calibration. Gain mismatch and sample-time error are overcome with correlation-based algorithms, which drive the correlation between a signal and its chopped image or its chopped and delayed image to zero. Test results show that, with a 0.99-MHz sinusoidal input, the ADC achieves a peak signal-to-noise-and-distortion ratio (SNDR) of 56.8 dB, a peak integral nonlinearity of 0.88 least significant bit (LSB), and a peak differential nonlinearity of 0.44 LSB. For a 39.9-MHz sinusoidal input, the ADC achieves a peak SNDR of 50.2 dB. The active area is 5.2 mm/sup 2/, and the power dissipation is 234 mW from a 3.3-V supply.
348 citations
Cites methods from "Splitting the Unit Delay - Tools fo..."
...To implement the fractional delay in practice, a causal FIR filter can be used [19], which introduces a delay of many sample periods in the lower signal path....
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01 Mar 1998
TL;DR: There are whole classes of algorithms that the speech community is not interested in pursuing or using in digital signal processing of sound and these algorithms and techniques are revealed in this book.
Abstract: With the advent of `multimedia', digital signal processing (DSP) of sound has emerged from the shadow of bandwidth limited speech processing to become a research field of its own. To date, most research in DSP applied to sound has been concentrated on speech, which is bandwidth limited to about 4 kilohertz. Speech processing is also limited by the low fidelity typically expected in the telephone network. Today, the main applications of audio DSP are high quality audio coding and the digital generation and manipulation of music signals. They share common research topics including perceptual measurement techniques and analysis/synthesis methods. Additional important topics are hearing aids using signal processing technology and hardware architectures for digital signal processing of audio. In all these areas the last decade has seen a significant amount of application-oriented research. The frequency range of wideband audio has an upper limit of 20 kilohertz and the resulting difference in frequency range and Signal to Noise Ratio (SNR) due to sample size must be taken into account when designing DSP algorithms. There are whole classes of algorithms that the speech community is not interested in pursuing or using. These algorithms and techniques are revealed in this book. This book is suitable for advanced level courses and serves as a valuable reference for researchers in the field. Interested and informed engineers will also find the book useful in their work.
300 citations
TL;DR: A 12-GS/s 5-bit time-interleaved flash ADC is realized in 65-nm CMOS to improve dynamic performance, and comparator offset calibration to reduce power dissipation.
Abstract: This paper presents a 12-GS/s 5-bit time-interleaved flash ADC realized in 65-nm CMOS. To improve the dynamic performance at high input frequencies, a statistics-based background calibration scheme for timing skew is employed. The timing skew is detected in the digital domain through a correlation-based algorithm and minimized by adjusting digitally controlled delay lines. In order to minimize power consumption, we employ near minimum size comparators, whose offset is reduced through foreground calibrated trim-DAC circuitry. With the timing calibration activated, the skew-related impairments are reduced by 12 dB at high input frequencies, resulting in an SNDR of 25.1 dB near Nyquist. The prototype IC consumes 81 mW from a 1.1 V supply, yielding a figure-of-merit of 0.35 pJ/conversion-step at low input frequencies, and 0.46 pJ/conversion-step for inputs near Nyquist.
241 citations
Cites background from "Splitting the Unit Delay - Tools fo..."
...The adaptive filter is a fractional delay filter [ 9 ] that interpolates between sub-ADC samples....
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01 Jan 2011
TL;DR: An overview of the state of the art in acoustic feedback control is provided, results of a comparative evaluation with a selection of existing methods are reported, and a glance at the challenges for future research is cast.
Abstract: The acoustic feedback problem has intrigued researchers over the past five decades, and a multitude of solutions has been proposed. In this survey paper, we aim to provide an overview of the state of the art in acoustic feedback control, to report results of a comparative evaluation with a selection of existing methods, and to cast a glance at the challenges for future research.
219 citations
TL;DR: An improved parametric model for a spring reverberation unit is presented, which is currently mainly used for simplified geometries or to generate reverberation impulse responses for use with a convolution method.
Abstract: The first artificial reverberation algorithms were proposed in the early 1960s, and new, improved algorithms are published regularly. These algorithms have been widely used in music production since the 1970s, and now find applications in new fields, such as game audio. This overview article provides a unified review of the various approaches to digital artificial reverberation. The three main categories have been delay networks, convolution-based algorithms, and physical room models. Delay-network and convolution techniques have been competing in popularity in the music technology field, and are often employed to produce a desired perceptual or artistic effect. In applications including virtual reality, predictive acoustic modeling, and computer-aided design of acoustic spaces, accuracy is desired, and physical models have been mainly used, although, due to their computational complexity, they are currently mainly used for simplified geometries or to generate reverberation impulse responses for use with a convolution method. With the increase of computing power, all these approaches will be available in real time. A recent trend in audio technology is the emulation of analog artificial reverberation units, such as spring reverberators, using signal processing algorithms. As a case study we present an improved parametric model for a spring reverberation unit.
218 citations
Cites methods from "Splitting the Unit Delay - Tools fo..."
...based on fourth- or higher-order Lagrange interpolation, can be applied [112], [111]....
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