Showing papers on "Sampling (signal processing) published in 2004"

Coherent detection method using DSP for demodulation of signal and subsequent equalization of propagation impairments

Abstract: A new approach to coherent detection is demonstrated which achieves the same high sensitivity as homodyne detection but without the need to phase lock the local oscillator laser. In addition, 1470 ps/nm of chromatic dispersion is compensated with zero net penalty by electronic domain equalization, a result which has not been achieved before because zero-penalty equalization is not possible after direct detection. The method proposes the use of high-speed digital signal processing technology, and the experimental results are obtained using burst-mode sampling followed by offline signal processing.

Multirate Signal Processing for Communication Systems

Abstract: Multirate Signal Processing for Communication Systems: Current Practice and Next Generation Techniques fredric j harrisMultirate signal processing can reduce costs and improve performance in applications ranging from laboratory instruments to cable modems, wireless systems, and consumer entertainment products. This book offers the first systematic, clear, and intuitive introduction to multirate signal processing for working engineers and system designers.The author uses extensive examples and figures to illuminate a wide range of multirate techniques, from basic resampling to leading-edge cascade and multiple-stage filter structures. Along the way, he draws on extensive research and consulting experience to introduce processing itricksi shown to maximize performance and efficiency.Coverage includes: Effective sampling and resampling in time and frequency domains Relationships between IIR Filter specifications and filter length (taps) Window design and equal-ripple (Remez) design techniques Square-Root Nyquist and Half Band Filters, including new design enhancements Polyphase IIR Filters: up-sampling, down-sampling, and cascade up-down sampling Polyphase interpolators and filters that perform arbitrary sample rate change Dyadic Half Band Filters, including quadrature mirror and IIR Filters Polyphase Channelizers, including M-path modulators, demodulator channel banks, simultaneous interpolation, and channel bank formation Comprehensive coverage of recursive all-pass filtersoa topic never before covered in this detail Comparisons with traditional DSP design techniques Extensive applications coverage throughout

A digitally enhanced 1.8-V 15-bit 40-MSample/s CMOS pipelined ADC

Abstract: A 1.8-V 15-bit 40-MSample/s CMOS pipelined analog-to-digital converter with 90-dB spurious-free dynamic range (SFDR) and 72-dB peak signal-to-noise ratio (SNR) over the full Nyquist band is presented. Its differential and integral nonlinearities are 0.25 LSB and 1.5 LSB, respectively, and its power consumption is 400 mW. This performance is enabled by digital background calibration of internal digital-to-analog converter (DAC) noise and interstage gain errors. The calibration achieves improvements of better than 12 dB in signal-to-noise plus distortion ratio and 20 dB in SFDR relative to the case where calibration is disabled. Other enabling features of the prototype integrated circuit (IC) include a low-latency, segmented, dynamic element-matching DAC, distributed passive input signal sampling, and asymmetric clocking to maximize the time available for the first-stage residue amplifier to settle. The IC is realized in a 0.18-/spl mu/m mixed-signal CMOS process and has a die size of 4mm/spl times/5 mm.

Adaptive sampling for sensor networks

Abstract: A distributed data-stream architecture finds application in sensor networks for monitoring environment and activities. In such a network, large numbers of sensors deliver continuous data to a central server. The rate at which the data is sampled at each sensor affects the communication resource and the computational load at the central server. In this paper, we propose a novel adaptive sampling technique where the sampling rate at each sensor adapts to the streaming-data characteristics. Our approach employs a Kalman-Filter (KF)-based estimation technique wherein the sensor can use the KF estimation error to adaptively adjust its sampling rate within a given range, autonomously. When the desired sampling rate violates the range, a new sampling rate is requested from the server. The server allocates new sampling rates under the constraint of available resources such that KF estimation error over all the active streaming sensors is minimized. Through empirical studies, we demonstrate the flexibility and effectiveness of our model.

Phase interpolator device and method

Abstract: A high-speed serial data transceiver includes multiple receivers and transmitters for receiving and transmitting multiple analog, serial data signals at multi-gigabit-per-second data rates Each receiver includes a timing recovery system for tracking a phase and a frequency of the serial data signal associated with the receiver The timing recovery system includes a phase interpolator responsive to phase control signals and a set of reference signals having different predetermined phases The phase interpolator derives a sampling signal, having an interpolated phase, to sample the serial data signal The timing recovery system in each receiver independently phase-aligns and frequency synchronizes the sampling signal to the serial data signal associated with the receiver A receiver can include multiple paths for sampling a received, serial data signal in accordance with multiple time-staggered sampling signals, each having an interpolated phase

A discrete-time Bluetooth receiver in a 0.13/spl mu/m digital CMOS process

Abstract: A discrete-time receiver architecture for a wireless application is presented. Analog signal processing concepts are used to directly sample the RF input at Nyquist rate. Maximum receiver sensitivity is -83dBm and the chip consumes a total of 41mA from a 1.575V internally regulated supply. The receiver is implemented in a 0.13/spl mu/m digital CMOS process.

Interference cancellation in a signal

Abstract: According to a first broad aspect of the present invention, there is provided a novel communication system having a forward link comprising: at least one base station which transmits multiple radio frequency (RF) signals; and a mobile station, the mobile station including: a receiver for receiving the RF signals; means for converting the RF signal to an intermediate frequency (IF) signal; means (704) for sampling the IF signal to generate a digital signal, the digital signal having a data component; means (712) for canceling co-channel and or cross-channel interference in the digital signal by projecting the IF signal into a subspace orthogonal to a subspace of the interference component and multiplying this projection with the digital signal; and means (710) for acquisition and tracking of the digital signal.

A practical, precise method for frequency tracking and phasor estimation

Abstract: Comprehensive analysis of discrete Fourier transform (DFT) error is given in this paper, including why it is accurate when used in the case of synchronous sampling and how error rises when sampling frequency does not synchronized to signal frequency. Simple but precise expressions of phase angle error and amplitude error are given. Practical formulas to calculate the true phase angle and amplitude are presented. The formulas are very simple and precise. Based on the formula to calculate true phase angle, a new frequency tracking method is developed. The proposed method can be calculated recursively. And, with notable accuracy improvement, the calculation burden is little more than the traditional DFT method. Also, an adaptive method to suppress the effect of harmonics is presented, which adds very little calculation burden with satisfying performance. The most distinguished feature of the proposed method is that it is not only precise, but also simple. Some examples are given to demonstrate the feasibility, precision, simpleness and robustness of the proposed method.

Reconstruction of band-limited periodic nonuniformly sampled signals through multirate filter banks

Abstract: A band-limited signal can be recovered from its periodic nonuniformly spaced samples provided the average sampling rate is at least the Nyquist rate. A multirate filter bank structure is used to both model this nonuniform sampling (through the analysis bank) and reconstruct a uniformly sampled sequence (through the synthesis bank). Several techniques for modeling the nonuniform sampling are presented for various cases of sampling. Conditions on the filter bank structure are used to accurately reconstruct uniform samples of the input signal at the Nyquist rate. Several examples and simulation results are presented, with emphasis on forms of nonuniform sampling that may be useful in mixed-signal integrated circuits.

Acoustic wave touch detection circuit and method

Abstract: A circuit for an acoustic wave switch or sensor having a resonant acoustic wave cavity detects a touch or sensed event using a time domain approach. The circuit includes a controller that drives an acoustic wave transducer to generate a resonant acoustic wave in the acoustic wave cavity during a first portion of a sampling cycle. In a second portion of the sampling cycle, the controller monitors the time that it takes for the acoustic wave signal from the transducer to decay to a predetermined level. Based on the decay time, the controller detects a sensed event, such as a touch on the acoustic wave switch/sensor.

Design techniques for a pipelined ADC without using a front-end sample-and-hold amplifier

Abstract: Design techniques for a low-power pipelined analog-to-digital converters (ADC) without using a front-end sample-and-hold amplifier are presented. Two sampling topologies are compared that minimize aperture error by matching the time constant between signal paths. A digital correction expansion technique is also presented for multibit ADCs, which further increases tolerance to aperture error. Elimination of the front-end SHA can save more than half of the ADCs static power dissipation.

A sampling algorithm for digitally controlled boost PFC converters

Abstract: Digital control of a boost power factor correction (PFC) converter requires sampling of the input current. As the input current contains a considerable amount of switching ripple and high frequency switching noise, the choice of the sampling instant is very important. To avoid aliasing without employing a (very) high sampling frequency, the sampling is synchronized with the pulse width modulation (PWM). Sampling algorithms employing this technique successfully reject the input current ripple but are not immune to the high frequency switching noise present on all sampled signals. Therefore, a new sampling algorithm, called alternating-edge-sampling and intended for center-based or symmetric PWM, is deduced with as most important features: switching noise immunity, straightforwardness, accurate measurement of the averaged input current and the need for only few processor cycles. The operating principle, design issues and a theoretical study of the input current error induced by the sampling algorithm due to sampling instant timing errors are derived. All theoretical results are validated experimentally for a digitally controlled boost PFC converter switching at 50 kHz.

Data-level clock recovery

Abstract: A circuit for adjusting the phase of a clock signal. A first sampling circuit generates a sequence of data samples in response to transitions of the clock signal, each of the data samples having either a first state or a second state according to whether an incoming signal exceeds a first threshold. An second sampling circuit generates an error sample in response to one of the transitions of the clock signal, the error sample having either the first state or the second state according to whether the incoming signal exceeds a second threshold. A phase adjust circuit adjusts the phase of the clock signal if the sequence of data samples matches a predetermined pattern and based, at least in part, on whether the error sample has the first state or the second state.

Display and control method thereof

Abstract: The present invention relates to an apparatus and a method of determining a display mode from a plurality of display modes for a display device that displays an image, each display mode relates to an inputted signal from a signal source that includes an analog signal and an H/V synchronization signal, the method including determining at least one display mode corresponding to the inputted H/V synchronization signal; converting the inputted analog signal to a digital signal according to a sampling signal corresponding to each of the at least one determined display mode; and displaying the image using a display mode from the at least one determined display mode that most closely relates to the signal source.

Adaptive compression and decompression of bandlimited signals

Abstract: An efficient method for compressing sampled analog signals in real time, without loss, or at a user-specified rate or distortion level, is described. The present invention is particularly effective for compressing and decompressing high-speed, bandlimited analog signals that are not appropriately or effectively compressed by prior art speech, audio, image, and video compression algorithms due to various limitations of such prior art compression solutions. The present invention's preprocessor apparatus measures one or more signal parameters and, under program control, appropriately modifies the preprocessor input signal to create one or more preprocessor output signals that are more effectively compressed by a follow-on compressor. In many instances, the follow-on compressor operates most effectively when its input signal is at baseband. The compressor creates a stream of compressed data tokens and compression control parameters that represent the original sampled input signal using fewer bits. The decompression subsystem uses a decompressor to decompress the stream of compressed data tokens and compression control parameters. After decompression, the decompressor output signal is processed by a post-processor, which reverses the operations of the preprocessor during compression, generating a postprocessed signal that exactly matches (during lossless compression) or approximates (during lossy compression) the original sampled input signal. Parallel processing implementations of both the compression and decompression subsystems are described that can operate at higher sampling rates when compared to the sampling rates of a single compression or decompression subsystem. In addition to providing the benefits of real-time compression and decompression to a new, general class of sampled data users who previously could not obtain benefits from compression, the present invention also enhances the performance of test and measurement equipment (oscilloscopes, signal generators, spectrum analyzers, logic analyzers, etc.), busses and networks carrying sampled data, and data converters (A/D and D/A converters).

A 5-b 10-GSample/s a/D converter for 10-gb/s optical receivers

Abstract: A 5-b flash A/D converter (ADC) is developed in an 0.18-/spl mu/m SiGe BiCMOS that supports sampling rates of 10 Gsample/s. The ADC is optimized to operate in digital equalizers for 10-Gb/s optical receivers, where the ADC has to deliver over three effective number of bits (ENOBs) at Nyquist. A fully differential flash ADC incorporating a wide-band track-and-hold amplifier (THA), a differential resistive ladder, an interpolation technique, and a high-speed comparator design is derived to resolve the aperture jitter and metastability error. The ADC achieves better than 4.1 effective bits for lower input frequencies and three effective bits for Nyquist input at 10 GS/s. The ADC dissipates about 3.6 W at the maximum clock rate of 10 GS/s while operating from dual -3.7/-3V supplies and occupies 3/spl times/3mm/sup 2/ of chip area.

Digital signal demodulation of an OFDM signal with error correction

Abstract: A digital signal demodulator digitizes an OFDM signal at a sampling frequency from a sampling oscillator to produce a digital OFDM signal. The digital OFDM signal is converted into I and Q components using a carrier frequency from a carrier oscillator. The IQ components are transformed into digital complex symbols, and pilot signals are extracted from the complex symbols. A processor calculates an inter-symbol difference of phase differences between pilot signals to control the sampling oscillator to correct the sampling frequency; calculates an inter-symbol difference for one of the pilot signals to control the carrier oscillator to correct the carrier frequency; and calculates a phase angle for one of the subcarriers at a frequency in the middle of the plurality of subcarriers for the OFDM signal to control the carrier oscillator to correct the carrier frequency phase.

Acoustic tomography on the basis of travel-time measurement

Abstract: Acoustic images of variable parameters of objects can be reconstructed by means of tomographic techniques which utilize the propagation of sound waves in the investigated medium. The technique described here utilizes sound frequencies in the audio range for acoustic imaging. The temperature-dependent sound speed as well as the flow field can be estimated by measuring the travel time of a defined acoustic signal between a sound source and a receiver when the distance between them is known exactly. The properties of the flow field are reconstructed using reciprocal sound rays to separate the direction-independent Laplace sound speed from the effective sound velocity. The temperatures in the flow field are then calculated by a combined inversion of all travel-time information resulting from the Laplace sound speed using an algebraic reconstruction technique. This reconstruction technique provides a cross section of the temperature distribution throughout the investigated area or volume. The tomographic system has been generalized to allow flow phenomena and temperature fields to be investigated with adapted sampling rates. The technique and procedures are exemplified by means of a scalable (from model-sized up to large-scale outdoor tomography) commercially oriented prototype of a tomograph which can utilize the whole audio and near-audio ultrasonic range. The software technology approach forms an inherent part of the realization.

Demodulator for a multi-pair gigabit transceiver

Abstract: A feedforward equalizer for equalizing a sequence of signal samples received by a receiver from a remote transmitter. The feedforward equalizer has a gain and is included in the receiver which includes a timing recovery module for setting a sampling phase and a decoder. The feedforward equalizer comprises a non-adaptive filter and a gain stage. The non-adaptive filter receives the signal samples and produces a filtered signal. The gain stage adjusts the gain of the feedforward equalizer by adjusting the amplitude of the filtered signal. The amplitude of the filtered signal is adjusted so that it fits in the operational range of the decoder. The feedforward equalizer does not affect the sampling phase setting of the timing recovery module of the receiver.

Analysis of practical sampling and reconstruction from Fresnel fields

Abstract: We present an analysis of the different aspects involved with the sampling and reconstruction of Fresnel field distribution. Fresnel fields, describing a propagating optical wave, are digitally recorded in many optical applications. The recording process involves discretization of the continuous Fresnel field using a sampling sensor. Typical nonideal sensors induce degradation of the optical information due to finite spatial sampling rate, finite aperture size, and finite detector element size (finite fill factor). In this work, we investigate the condition and limitation of the reconstruction of Fresnel fields sampled with nonideal sampling sensors. We also analyze the propagation of measurement and reconstruction noise through the reconstruction process. In our analysis, we distinguish between continuous (optical) and numerical (computational) reconstructions. We focus on the different reconstruction conditions and limitations, depending on whether the reconstruction is performed in a continuous space or numerically.

Enhanced perception lighting

Abstract: The present invention relates to an enhanced perception lighting (EPL) system for providing enhanced perception of a user's physical environment. The EPL system comprises a sensor module for detecting and sampling a physical aspect from at least one point in a physical environment and for generating an observation signal based on the physical aspect; a processor module coupled with the sensor module for receiving the observation signal, processing the observation signal, and generating an output signal based on the observation signal; and a projection display module located proximate the sensor module and communicatively connected with the processor module for projecting a display onto the at least one point in the physical environment based upon the output signal. The system allows a user to gather information from the physical environment and project that information onto the physical environment to provide the user with an enhanced perception of the physical environment.

Array speaker system

Abstract: An array speaker system, in which a plurality of speaker units are arranged in an array and are supplied with signals having prescribed time differences so as to perform directivity control on audio signal beams emitted therefrom, includes a delay memory (e.g., a shift register) having plural delay taps for outputting an input signal thereof with different delay times, which are set in units of the sampling period, and an interpolation processing means for performing interpolation processing on the output of the delay memory. A control means calculates distances between a focal point of audio signal beams and the speaker units so as to produce delay times, and it also sets interpolation coefficients with respect to the speaker units respectively. The interpolation processing means performs linear interpolation on the outputs of the delay memory. Alternatively, an FIR low-pass filter is formed using the delay memory and interpolation processing means, thus performing delay and interpolation processing. Delayed and interpolated signals are supplied to the speaker units, thus performing directivity control on audio signal beams with high precision.

An FFT-based method to evaluate and compensate gain and offset errors of interleaved ADC systems

Abstract: Interleaved analog-digital converter (ADC) systems can be used to increase the sampling rate for a given ADC implementation technique. In theory, the maximum sampling rate that can be achieved is limited only by the bandwidth and the practical limits related to the power and space of integrated circuits. In this paper, a solution to increase the sampling rate of a digitizing system based on interleaved ADCs is presented. An error analysis, which takes into consideration offset and gain errors of the different ADC channels, is performed in order to quantify the effect of such errors in the system's performance. A software method based on the fast Fourier transform is presented for offset and gain error compensation of interleaved ADC associations. Numerical simulations and experimental results are used to validate the theory and the proposed compensation algorithm.

A cascaded continuous-time ΣΔ modulator with 67-dB dynamic range in 10-MHz bandwidth

Abstract: This paper presents the design of a 2-2 cascaded continuous-time sigma-delta modulator. The cascaded modulator comprises two stages with second-order continuous-time resonator loopfilters, 4-bit quantizers, and feedback digital-to-analog converters. The digital noise cancellation filter design is determined using continuous-time to discrete-time transformation of the sigma-delta loopfilter transfer functions. The required matching between the analog and digital filter coefficients is achieved by means of simple digital calibration of the noise cancellation filter. Measurement results of a 0.18-μm CMOS prototype chip demonstrate 67-dB dynamic range in a 10-MHz bandwidth at 8 times oversampling for a single continuous-time cascaded modulator. Two cascaded modulators in quadrature configuration provide 20-MHz aggregate bandwidth. Measured anti-alias suppression is over 50 dB for input signals in the band from 150 to 170 MHz around the sampling frequency of 160 MHz.

Circuit and method for measuring jitter of high speed signals

Abstract: A method and circuit for measuring a statistical value of jitter for a data signal having a data rate fD, comprises digitally sampling the data signal at a sampling rate, fS, to produce sampled logic values, where fD/fS is a predetermined non-integer ratio; and analyzing the sampled values to deduce a statistical value of the jitter.

4/spl times/2.5 GHz repetitive photonic sampler for high-speed analog-to-digital signal conversion

Abstract: This letter presents a 10-gigasample/s (GS/s) photonic analog-to-digital converter (ADC) system constructed using a four-wavelength picosecond pulsed source. The lasing-to-nonlasing modes suppression ratio of the optical source is over 24 dB. By using the 10-GHz optical source, a 10-GS/s photonic ADC has been demonstrated and was used to sample an arbitrary radio-frequency signal. The system was further investigated by sampling a 2.4-GHz sinusoidal signal. Important parameters including the signal-to-noise and distortion ratio and the spurious-free dynamic range have been determined.

Position-sensitive Si pad detectors for electron emission channeling experiments

Abstract: Position-sensitive detector systems, initially developed for the detection of X-rays, have been adapted for their use in electron emission channeling experiments. Each detection system consists of a 30.8×30.8 mm 2 22×22-pad Si detector, either of 0.3, 0.5 or 1 mm thickness, four 128-channel preamplifier chips, a backplane trigger circuit, a sampling analog to digital converter, a digital signal processor, and a personal computer for data display and storage. The operational principle of these detection systems is described, and characteristic features such as energy and position resolution and maximum count rate, which have been determined from tests with conversion electrons and β − particles in the energy range 40–600 keV, are presented.