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Showing papers on "Sampling (signal processing) published in 1999"


Journal ArticleDOI
R.H. Walden1
TL;DR: The state-of-the-art of ADCs is surveyed, including experimental converters and commercially available parts, and the distribution of resolution versus sampling rate provides insight into ADC performance limitations.
Abstract: Analog-to-digital converters (ADCs) are ubiquitous, critical components of software radio and other signal processing systems. This paper surveys the state-of-the-art of ADCs, including experimental converters and commercially available parts. The distribution of resolution versus sampling rate provides insight into ADC performance limitations. At sampling rates below 2 million samples per second (Gs/s), resolution appears to be limited by thermal noise. At sampling rates ranging from /spl sim/2 Ms/s to /spl sim/4 giga samples per second (Gs/s), resolution falls off by /spl sim/1 bit for every doubling of the sampling rate. This behavior may be attributed to uncertainty in the sampling instant due to aperture jitter. For ADCs operating at multi-Gs/s rates, the speed of the device technology is also a limiting factor due to comparator ambiguity. Many ADC architectures and integrated circuit technologies have been proposed and implemented to push back these limits. The trend toward single-chip ADCs brings lower power dissipation. However, technological progress as measured by the product of the ADC resolution (bits) times the sampling rate is slow. Average improvement is only /spl sim/1.5 bits for any given sampling frequency over the last six-eight years.

2,220 citations


Journal ArticleDOI
TL;DR: This paper investigates some simple first order systems with event based sampling and compares achieved closed loop variance and sampling rate with results from periodic sampling and shows that event based sampled gives better performance than periodic sampling.

684 citations


Journal ArticleDOI
TL;DR: A closed-form weighted-equation-error method is derived that computes the optimal mapping coefficient as a function of sampling rate, and the solution is shown to be generally indistinguishable from the optimal least-squares solution.
Abstract: Use of a bilinear conformal map to achieve a frequency warping nearly identical to that of the Bark frequency scale is described Because the map takes the unit circle to itself, its form is that of the transfer function of a first-order allpass filter Since it is a first-order map, it preserves the model order of rational systems, making it a valuable frequency warping technique for use in audio filter design A closed-form weighted-equation-error method is derived that computes the optimal mapping coefficient as a function of sampling rate, and the solution is shown to be generally indistinguishable from the optimal least-squares solution The optimal Chebyshev mapping is also found to be essentially identical to the optimal least-squares solution The expression 08517[arctan(006583fs)]/sup 1/2/-0916 is shown to accurately approximate the optimal allpass coefficient as a function of sampling rate f/sub s/ in kHz for sampling rates greater than 1 kHz A filter design example is included that illustrates improvements due to carrying out the design over a Bark scale Corresponding results are also given and compared for approximating the related "equivalent rectangular bandwidth (ERB) scale" of Moore and Glasberg (ACTA Acustica, vo82, p335-45, 1996) using a first-order allpass transformation Due to the higher frequency resolution called for by the ERB scale, particularly at low frequencies, the first-order conformal map is less able to follow the desired mapping, and the error is two to three times greater than the Bark-scale case, depending on the sampling rate

432 citations


Journal ArticleDOI
15 Feb 1999
TL;DR: A 640/spl times/512 image sensor with Nyquist rate pixel level ADC implemented in a 0.35 /spl mu/m CMOS technology shows how a pixellevel ADC enables flexible efficient implementation of multiple sampling.
Abstract: Analysis results demonstrate that multiple sampling can achieve consistently higher signal-to-noise ratio at equal or higher dynamic range than using other image sensor dynamic range enhancement schemes such as well capacity adjusting. Implementing multiple sampling, however, requires much higher readout speeds than can be achieved using typical CMOS active pixel sensor (APS). This paper demonstrates, using a 640/spl times/512 CMOS image sensor with 8-b bit-serial Nyquist rate analog-to-digital converter (ADC) per 4 pixels, that pixel-level ADC enables a highly flexible and efficient implementation of multiple sampling to enhance dynamic range. Since pixel values are available to the ADC's at all times, the number and timing of the samples as well as the number of bits obtained from each sample can be freely selected and read out at fast SRAM speeds. By sampling at exponentially increasing exposure times, pixel values with binary floating-point resolution can be obtained. The 640/spl times/512 sensor is implemented in 0.35-/spl mu/m CMOS technology and achieves 10.5/spl times/10.5 /spl mu/m pixel size at 29% fill factor. Characterization techniques and measured quantum efficiency, sensitivity, ADC transfer curve, and fixed pattern noise are presented. A scene with measured dynamic range exceeding 10000:1 is sampled nine times to obtain an image with dynamic range of 65536:1. Limits on achievable dynamic range using multiple sampling are presented.

345 citations


Journal ArticleDOI
16 May 1999
TL;DR: In this paper, an 8-bit 5-stage pipelined and interleaved analog-to-digital converter that performs analog processing only by means of open-loop circuits such as differential pairs and source followers is described.
Abstract: This paper describes an 8-bit 5-stage pipelined and interleaved analog-to-digital converter that performs analog processing only by means of open-loop circuits such as differential pairs and source followers to achieve a high conversion rate. The concept of sliding interpolation is proposed to obviate the need for a large number of comparators or interstage digital-to-analog converters and residue amplifiers. The pipelining scheme incorporates distributed sampling between the stages so as to relax the linearity-speed tradeoffs in the sample-and-hold circuits, A clock edge reassignment technique is also introduced that suppresses timing mismatches in interleaved systems, and a punctured interpolation method is proposed that reduces the integral nonlinearity error with negligible speed or power penalty. Fabricated in a 0.6-/spl mu/m CMOS technology, the converter achieves differential and integral nonlinearities of 0.62 and 1.24 LSB, respectively, and a signal-to-(noise+distortion) ratio of 43.7 dB at a sampling rate of 150 MHz. The circuit draws 395 mW from a 3.3-V supply and occupies an area of 1.2/spl times/1.5 mm/sup 2/.

234 citations


Journal ArticleDOI
R.H. Walden1
TL;DR: The data imply that the performance measure, P=2/sup SNRbits//spl middot/f/sub samp/, uncertainty in the sampling process (aperture jitter) over a very wide range of sampling rates.
Abstract: Analog-to-digital converters are key components of signal processing systems, and may even dictate system architectures due to their limitations on sampling rate and resolution. The state of the art for ADCs, including both experimental converters and commercially available parts, is reviewed. The data imply that the performance measure, P=2/sup SNRbits//spl middot/f/sub samp/, uncertainty in the sampling process (aperture jitter) over a very wide range of sampling rates. For ADCs operating at multi-GSPS rates, the speed of the device technology is also a limiting factor (due to comparator ambiguity). Technological progress as measured by P is discussed.

177 citations


Journal ArticleDOI
15 Feb 1999
TL;DR: This ADC uses a queue-based architecture for creating calibration time slots without disturbing the sampling of the input signal and an adaptive algorithm to improve linearity, both of which are independent and can be used separately.
Abstract: The linearity of analog-to-digital converters (ADCs) is often limited by component mismatches. Trimming can be used to achieve high linearity but cannot track variations over time caused by component aging or by temperature and power-supply changes. Background calibration overcomes this limitation. However, previous background-calibration methods require complicated post processing, occupy some of the range of the analog signal under conversion, or are tailored for a specific type of converter. This ADC uses a queue-based architecture for creating calibration time slots without disturbing the sampling of the input signal. The digital background calibration uses an adaptive algorithm to improve linearity. The queue-based architecture for generating the calibration time slots and the digital-background-calibration method are independent and can be used separately.

137 citations


Patent
Juin-Hwey Chen1
30 Mar 1999
TL;DR: In this paper, a scalable and low-complexity adaptive transform coding method for speech and general audio signals is presented. But the method is not suitable for the Internet Protocol (IP)-based multimedia communications.
Abstract: High-quality, low-complexity and low-delay scalable and embedded system and method are disclosed for coding speech and general audio signals. The invention is particularly suitable in Internet Protocol (IP)-based multimedia communications. Adaptive transform coding, such as a Modified Discrete Cosine Transform, is used, with multiple small-size transforms in a given signal frame to reduce the coding delay and computational complexity. In a preferred embodiment, for a chosen sampling rate of the input signal, one or more output sampling rates may be decoded with varying degrees of complexity. Multiple sampling rates and bit rates are supported due to the scalable and embedded coding approach underlying the present invention. Further, a novel adaptive frame loss concealment approach is used to reduce the distortion caused by packet loss in communications using IP networks.

135 citations


Patent
23 Dec 1999
TL;DR: In this article, a generic multi-functional transcoder architecture has a post-preprocessing engine that provides a number of processing functions for implementing desired format conversions according to a user selection signal or an automatically generated selection signal.
Abstract: A method and apparatus for transcoding digital video data, for example, at the headend of a cable or satellite television network A generic multi-functional transcoder architecture has a “post-pre-processing engine” that provides a number of processing functions for implementing desired format conversions according to a user selection signal or an automatically generated selection signal The processing functions can change frame size, frame rate, color space sampling format, interlaced or progressive scan format, resolution, and provide noise/deblocking filtering, for example The system also allows the control of several transcoders to convert several bit streams with different formats into corresponding bit streams with a common format Furthermore, the transcoder avoids the need for motion estimation

134 citations


Proceedings ArticleDOI
16 May 1999
TL;DR: In this article, a 10-bit converter that can sample input frequencies above 100 MHz is presented, which consumes 55 mW when sampling at f/sub s/= 40 MHz from a 3 V supply which also includes a bandgap and a reference circuit.
Abstract: A low power 10-bit converter which can sample input frequencies above 100 MHz is presented. The converter consumes 55 mW when sampling at f/sub s/= 40 MHz from a 3 V supply which also includes a bandgap and a reference circuit. It exhibits higher than 9.5 effective number of bits (ENOB) for an input frequency at Nyquist (f/sub in/= 1/2 f/sub s/=20 MHz). The differential (DNL) and integral (INL) nonlinearity of the converter are within /spl plusmn/0.3 LSB and /spl plusmn/0.75 LSB respectively when sampling at 40 MHz, and improve to a 12-bit accuracy level for lower sampling rates. The overall performance is achieved using a pipeline architecture without a dedicated sample/hold circuit at the input.

128 citations


Journal ArticleDOI
TL;DR: In this paper, the determina- tion of the Nyquist frequency in the irregular case of variable stars has been investigated and a specic example is shown, drawn from MACHO databases.
Abstract: In the analysis of variable stars, the problem of sampling is central This article focusses on the determina- tion of the Nyquist frequency It is well dened in the case of regular sampling However, the time series of variable stars observations are generally unevenly sampled Fourier analysis using the spectral window furnishes some clues about the equivalent Nyquist frequency in the irregular case Often it is pushed very high, and thus very short pe- riods can be detected A specic example is shown, drawn from MACHO databases

Journal ArticleDOI
TL;DR: The authors review the most important polynomial predictive filtering methods and algorithms, their design and implementation techniques, and a collection of successful applications.
Abstract: Additional delay is an unavoidable drawback of conventional filters used frequently in industrial electronics. This delay is particularly harmful if the filtered primary signal is to be used for time-critical feedback or synchronization purposes. Therefore, predictive signal processing methods can offer significant advantages for these real-time applications. Polynomial predictive filters are specified without explicit passbands and stopbands, and they are behaving delaylessly or predictively for smoothly varying signal components. The degree of smoothness of the incoming signal sets the requirements for the applied filtering scheme and its parameters. Smoothness of a signal is a fuzzy and application-specific concept: the degree of smoothness depends on the ratio of the bandwidth of the primary signal and the applied sampling rate, as well as the noise component. In this paper, the authors review the most important polynomial predictive filtering methods and algorithms, their design and implementation techniques, and a collection of successful applications.

Patent
27 Sep 1999
TL;DR: In this article, a method of operating a radar system, including the steps of digitally sampling a received signal at a predetermined sampling rate, to periodically provide a set of selected samples, including positive and negative going ramp samples, and CW burst samples, is described.
Abstract: A method of operating a radar system, including the steps of digitally sampling a received signal at a predetermined sampling rate, to periodically provide a set of selected samples, the set of selected samples including positive going ramp samples, negative going ramp samples and CW burst samples and performing a first fast Fourier transform (FFT) on the positive going ramp samples, performing a second fast Fourier transform on the negative going ramp samples and performing a third fast Fourier transform on the CW burst samples is described. Utilizing the subsequent radar operations the method further includes the steps of tracking each resulting signal from the first fast Fourier transform performing steps, tracking each resulting signal from the second fast Fourier transform performing steps and tracking each resulting signal from the third fast Fourier transform performing steps and associating any resulting signals from the tracking steps to periodically provide output signals indicative of other vehicles.

Journal ArticleDOI
TL;DR: In this article, a time-varying theory of Volterra series is developed and applied in the sampled data domain to solve for harmonic and intermodulation distortion of a MOS-based track-and-hold sampling mixer with a nonzero fall-time LO waveform.
Abstract: A time-varying theory of Volterra series is developed and applied in the sampled-data domain to solve for harmonic and intermodulation distortion of a MOS-based track-and-hold sampling mixer with a nonzero fall-time LO waveform. Distortion due to sampling error is also calculated. These results, when combined with the continuous-time solution, quantify harmonic and intermodulation distortion of a track-and-hold type mixer completely. Closed form solutions are obtained. As a practical consequence, it is shown that for certain fall-time, the distortion of track-and-hold mixers can be better than what would be predicted by a simple application of time-invariant Volterra series theory.

Patent
25 Aug 1999
TL;DR: In this paper, a PLL (35) based clock recovery unit is used to adjust the sampling and comparing of the signal in the data channel and the clock signal is derived by the microprocessor.
Abstract: A microprocessor (45) controlled data recovery unit with an adjustable sampling and signal comparison level. The data recovery unit includes a data channel (47a) and a monitor channel (47b). The monitor channel samples an incoming data stream in a varying manner. The results of the sampling in the monitor channel are used to adjust the sampling and comparing of the signal in the data channel. The data recovery unit includes a PLL (35) based clock recovery unit in one embodiment, and in another embodiment the clock signal is derived by the microprocessor.

PatentDOI
17 Dec 1999
TL;DR: In this article, a sonar depth sounder is used for processing reflected sonar signals, reflected from objects within a body of water, utilizing a processor, a memory, a display, and a keypad connected to the processor.
Abstract: A sonar depth sounder device and method for processing echo signals, reflected from objects within a body of water, utilizes a processor, a memory, a display, and a keypad connected to the processor. The receiver receives sonar signals indicative of ambient noise in an underwater environment. The processor receives an electrical signal representative of the ambient noise, and calculates a detection threshold. The detection threshold is calculated by multiplying a scaling factor times the variance of the signal indicative of the ambient noise, and adding that product to the mean of the signal indicative of the ambient noise in the underwater environment. A transmitter then excites a transducer which emits sonar pulses into a body of water, and a receiver receives reflected sonar echo signals. When the intensity of the reflected sonar echo signals is less than the detection threshold, the reflected echo signals are eliminated from processing. When, however, the amplitude of a reflected sonar signal is greater than the detection threshold, the processor causes data indicative of the reflected sonar signal to be displayed on the display. Further, the detection threshold is increased from its starting point over time to compensate for increasing a gain in the receiver of the sonar depth sounder device. Additionally, the detection threshold is periodically updated to compensate for changes in ambient noise in the underwater environment. In an alternate embodiment, after a preliminary value is determined, based upon ambient noise or predicted ambient noise, the sonar depth sounder device generates a continuous time-varying detection threshold by sampling a received echo signal and storing the samples in memory, applying a continuous time averaging technique to the sampled data, wherein sample data comprises a time varying detection threshold. That time varying detection threshold is then compared on a sample-by-sample basis with corresponding samples of the originally received data, such that the originally received data that is greater than its corresponding detection threshold sample is displayed, whereas originally received data is less than its corresponding detection threshold sample is rejected and not displayed. In a preferred embodiment, in order to display data indicative of an originally received data sample that is greater than the detection threshold, it must be part of a sequence of consecutive data samples that are greater than the corresponding detection threshold samples, wherein that sequence has an associated time that is greater than a selected time limit.

Patent
10 Feb 1999
TL;DR: In this article, a sliding correlator for use in a spread spectrum receiver divides baseband signal samples into different groups, associates each group with a different section of a spreading code, and combines ones of the signal samples with corresponding values in the spreading code section.
Abstract: A flexible sliding correlator for use in a spread spectrum receiver divides baseband signal samples into different groups, associates each group with a different section of a spreading code, and combines ones of the signal samples with corresponding values in the spreading code section. The groupings and spreading code sections can be changed during operation of the receiver to maximize performance of the receiver under different or changing conditions. In addition, the sample and spreading code value combinations can be further combined in different ways, and the further combinations can be changed during operation of the receiver. According to another aspect of the invention, the baseband signal can be sampled either uniformly or non-uniformly. The phase and frequency of the baseband sampling can be adjusted during operation of the receiver so that samples are taken very close to the optimum sampling position, at the peak of a chip waveform in the baseband signal.

09 Jun 1999
TL;DR: An analysis is presented to explain the noise behavior, show evidence of degraded performance under low-light levels, and describe new pixels that eliminate non-linearity and lag without compromising noise.
Abstract: Noise in photodiode-type CMOS active pixel sensors (APS) is primarily due to the reset (kTC) noise at the sense node, since it is difficult to implement in-pixel correlated double sampling for a 2-D array Signal integrated on the photodiode sense node (SENSE) is calculated by measuring difference between the voltage on the column bus (COL) - before and after the reset (RST) is pulsed Lower than kTC noise can be achieved with photodiode-type pixels by employing "softreset" technique Soft-reset refers to resetting with both drain and gate of the n-channel reset transistor kept at the same potential, causing the sense node to be reset using sub-threshold MOSFET current However, lowering of noise is achieved only at the expense higher image lag and low-light-level non-linearity In this paper, we present an analysis to explain the noise behavior, show evidence of degraded performance under low-light levels, and describe new pixels that eliminate non-linearity and lag without compromising noise

Patent
25 Feb 1999
TL;DR: In this article, the authors propose a system and method for enabling an efficient zero phase restart (ZPR) of a device based on deploying normalized timing gradient (NTG) blocks ( 501 and 502 ) in pairs, each circuit employing an orthogonal phase error transfer function characteristic ( having one TG circuit sample orthogonally in relation to the other), for example, PR4 and EPR4 modes ideal sampling instances of a preamble.
Abstract: A system and method for enabling an efficient Zero Phase Restart (ZPR) of a device. The structure is based on deploying normalized timing gradient (NTG) blocks ( 501 and 502 ) in pairs, each circuit employing an orthogonal phase error transfer function characteristic (having one TG circuit sample orthogonally in relation to the other), for example, PR4 and EPR4 modes ideal sampling instances of a preamble. An NTG block ( 501 or 502 ) is selected based on having a native timing sampling instance with a phase error that is closest to zero. Since there is an equal chance that either of the circuits in a circuit pair will be selected, if the circuit implementing the current non-native architecture is selected, a separate signal is generated. This signal adds the equivalent of 180° to the error value that is provided to the timing recovery circuit. For example, by iterating the process after the special case of a zero phase restart (ZPR) operation, the native sampling instance is “forced” to be selected thereafter.

Patent
18 Oct 1999
TL;DR: In this article, a method and apparatus that utilize time-domain measurements of a nonlinear device produce or extract a behavioral model from embeddings of these measurements, and verify the fitted function.
Abstract: A method and apparatus that utilize time-domain measurements of a nonlinear device produce or extract a behavioral model from embeddings of these measurements. The method of producing a behavioral model comprises applying an input signal to the nonlinear device, sampling the input signal to produce input data, measuring a response of the device to produce output data, creating an embedded data set, fitting a function to the embedded data set, and verifying the fitted function. The apparatus comprises a signal generator that produces an input signal that is applied to the nonlinear device, the device producing an output signal in response. The apparatus further comprises a data acquisition system that samples and digitizes the input and output signals and a signal processing computer that produces an embedded data set from the digitized signals, fits a function to the embedded data set, and verifies the fitted function.

Patent
11 Jun 1999
TL;DR: In this paper, a method and apparatus for evaluating signal strength of a channel received at a mobile station within a spread spectrum communication system is disclosed, where the receiver at the mobile station receives a broadcast signal and converts the received signal into a first sample stream as a first sampling.
Abstract: A method and apparatus for evaluating signal strength of a channel received at a mobile station within a spread spectrum communication systems is disclosed If the receiver at the mobile station receives a spread spectrum signal, a first sampling means converts the received signal into a first sample stream as a first sampling A second sampling means converts the first sample stream into a second sample stream at a second sample rate, different from the first sample rate The signal strength of a pilot channel is measured based upon the first and second sample streams

Patent
08 Dec 1999
TL;DR: In this paper, a received signal obtained from an antenna is subjected to high-frequency amplification, and the amplified signal is supplied to a first bandpass filter, which extracts only signals of all the channels of a communications system concerned while filtering out other radio signals.
Abstract: A received signal obtained from an antenna is subjected to high-frequency amplification. The amplified signal is supplied to a first bandpass filter, which extracts only signals of all the channels of a communications system concerned while filtering out other radio signals. The extracted signals are frequency-converted by using a local oscillation frequency, and only a desired wave is passed by a second bandpass filter. The desired wave is supplied to a sample-and-hold circuit, which performs sampling according to the bandwidth-limiting sampling theorem. A resulting discrete signal is supplied to an I-axis-component and Q-axis-component separating circuits, where the polarity of sample values is inverted for every other clock pulse with respect to each of the I and Q axes to thereby effect Hilbert transform. Resulting two orthogonal components on a phase plane are supplied to a complex coefficient filter.

Patent
16 Aug 1999
TL;DR: In this article, a system and method for detecting a stable region in a data signal to facilitate the alignment between the data signal and a corresponding clock signal is presented, which includes a processor coupled to a local interface and a memory coupled to the local interface.
Abstract: A system and method are provided for detecting a stable region in a data signal to facilitate the alignment between a data signal and a corresponding clock signal. The system includes a processor coupled to a local interface and a memory coupled to the local interface. The system also includes a boundary detection circuit configured to perform a simultaneous sampling of a reference signal and a delayed reference signal to ascertain a degree of stability of a position in the reference signal. The reference signal is the signal received from the target system and the delayed reference signal is a delayed copy of the reference signal. The system also includes boundary detection logic stored on the memory and executed by the processor to control the operation of the boundary detection circuit. The boundary detection logic includes logic to detect a boundary of the stable region of the reference.

Patent
Hideshi Ohya1
26 Jul 1999
TL;DR: In this paper, the authors presented a system that generates resolver reference signal on the basis of the clock of CPU 202, and the servo control loop trigger signals in synchronization with its own clock.
Abstract: The disclosed system generates resolver reference signal on the basis of the clock of CPU 202. The CPU 202 generates resolver output signal sampling and servo control loop trigger signals in synchronization with its own clock. Thus, the synchronization between the reference signal and the sampling is assured. Because the sampling is constantly performed at a fixed phase point of the reference signal, the sampling accuracy is improved, while stable servo control loop response is obtained because the sampling is also synchronous with servo control loop so that the sampling will not interrupt the servo control loop.

Patent
13 Jul 1999
TL;DR: In this paper, an improved version of the frequency-domain SAFT (F-SAFT) based on the angular spectrum approach is described, which includes temporal deconvolution of the waveform data to enhance both axial and lateral resolutions, control of the aperture and frequency bandwidth to improve signal-tonoise ratio, as well as spatial interpolation of the subsurface images.
Abstract: A method and system is provided for enhanced ultrasonic detection and imaging of small defects inside or at the surface of an object The Synthetic Aperture Focusing Technique (SAFT) has been used to improve the detectability and to enhance images in conventional ultrasonics and this method has recently been adapted to laser-ultrasonics In the present invention, an improved version of the frequency-domain SAFT (F-SAFT) based on the angular spectrum approach is described The method proposed includes temporal deconvolution of the waveform data to enhance both axial and lateral resolutions, control of the aperture and of the frequency bandwidth to improve signal-to-noise ratio, as well as spatial interpolation of the subsurface images All the above operations are well adapted to the frequency domain calculations and embedded in the F-SAFT data processing The aperture control and the spatial interpolation allow also a reduction of sampling requirements to further decrease both inspection and processing times This method is of particular interest when ultrasound is generated by a laser and detected by either a contact ultrasonic transducer or a laser interferometer

Patent
13 Aug 1999
TL;DR: In this paper, the uplink signal estimate is calculated using the sequence of downlink signal samples and a first sequence of tap coefficients, while the downlink signals are estimated using a sequence of previous uplink estimates and a second sequence of tapp coefficients.
Abstract: A receiving and transmitting apparatus includes receiver circuitry, sampling circuitry, downlink prediction circuitry, uplink prediction circuitry and gain calculation circuitry. A downlink signal is received at the receiver and sampled at the sampling circuitry. Downlink and uplink prediction circuitry is used to estimate transmission properties of the downlink and uplink signals, respectively. The downlink and uplink prediction circuitry estimates downlink and uplink signal properties based on a sequence of previous downlink signal samples and the uplink estimates, respectively, as well as corresponding sequences of tap coefficients. The gain calculation circuitry is coupled to the uplink predication circuitry and to a transmitter and can set a transmitter gain level based on the predicted uplink signal properties. A method of controlling power in a communication device includes storing a sequence of downlink signal samples, calculating a downlink signal estimate and an uplink signal estimate, and setting a transmission power level based on the estimated uplink signal. The downlink signal estimate is calculated using the sequence of downlink signal samples and a first sequence of tap coefficients. The uplink signal estimate is calculated using a sequence previous uplink signal estimates and a second sequence of tap coefficients.

Patent
24 May 1999
TL;DR: In this article, a graphics processing system (100) incorporates a calibrator module (150) into the system, which automatically increments the number of stages of delay (170), which are integrated into a delayed clock signal.
Abstract: A graphics processing system (100) incorporates a calibrator module (150) into the system. As a memory module (120) continuously transmits a model data signal, the calibrator module (150) automatically increments the number of stages of delay (170), which are integrated into a delayed clock signal. Each delayed clock signal triggers the sampling of the model data signal by a plurality of latches (130). The calibrator module compares (220) each of these sampled data signals with the original model data signals. If the delayed clock signal is properly aligned with the model data signal to cause the two signals to match, the calibrator module stores a result signal in a '1' logic state (230). If the delayed clock signal is misaligned with the model data signal, the calibrator module will store the result signal in a '0' logic state (230). When all of the possible stages of delay have been activated by the calibrator module and the corresponding sampled data signals analyzed, a processor module determines the optimum number of stages of delay needed for proper alignment of the delay clock signal with the transmitted model data signal.

Patent
21 Sep 1999
TL;DR: In this article, a video signal is processed to generate a first signal indicative of detected edges in the video signal, which is then processed with a nonlinear transfer function to generate an output signal having enhanced edges with reduced or minimal amounts of undershoots and overshoots.
Abstract: Techniques for enhancing edges in video signals while reducing the amounts of undershoots and overshoots. A video signal is processed to generate a first signal indicative of detected edges in the video signal. The first signal can be generated by lowpass filtering the video signal to generate a lowpass signal and subtracting the lowpass signal from a luminance signal that has been extracted from the video signal. The first signal is then processed with a “non-linear” transfer function to generate a second signal having enhanced edges. The second signal is used as the correction or enhancement signal, and is added to the lowpass signal to provide an output signal having enhanced edges with reduced or minimal amounts of undershoots and overshoots. The second signal has one or more of the following characteristics: (1) it is dynamically generated based on characteristics of the detected edges in the video signal; (2) it provides varying amounts of enhancement across the detected edges in the video signal; (3) it provides higher amounts of enhancement near the center of the detected edges and smaller amounts of enhancement away from the center; and (4) it provides an amount of enhancement that is dependent on the slope of the detected edges.

Journal ArticleDOI
TL;DR: In this article, a dynamic MRT method has been developed and measured which uses motion to provide enhanced sampling, but the authors don't feel that static MRT measured with either optimum or unknown phase relationships correlate well with field performance.
Abstract: An infrared imager (often referred to as a FLIR, from the term forward looking infrared) can be characterized in terms of sensitivity, resolution and human performance. Sensitivity, resolution and human performance have been classically described by the following measurable parameters: noise equivalent temperature difference (NETD), modulation transfer function (MTF) and minimum resolvable temperature difference (MRTD or MRT). These are laboratory measurable quantities that can be used to verify the performance of an infrared system. These laboratory measurements are used to evaluate expected design performance and to periodically test an imager during its life cycle. These quantities are predictable in sensor design through the use of analytical models. Both model estimates and laboratory measurements can be used in a target acquisition model to determine the field performance (probability of detection, recognition or identification) of the imager. Sensitivity, resolution, and human performance are influenced by sampling artifacts that must be characterized. First, sensitivity is no longer sufficiently described by a single-valued NETD. The 3-D noise methodology, inhomogeneity equivalent temperature difference (IETD), and correctability are noise figures that have been developed over the past decade to more adequately describe both temporal and fixed pattern noise associated with focal plane arrays. Undersampled imaging systems are not shift-invariant. The shift-invariance condition, in particular, is compromised by under-sampling so that sensor performance becomes a function of phase (relative position between the image and the detector array). Resolution depends on the target-to-imager phase, so the MTF measurement may reveal sampling artifacts that give large MTF variations with target to sensor position. Finally, the human-performance parameter is perhaps most affected by undersampling. MRT can be strongly dependent on phase, where dramatic differences in measured MRT are attributed to different phase shifts. MRT is normally measured at optimum phase, yet the authors don't feel that static MRT measured with either optimum or unknown phase relationships correlate well with field performance. In the same way, it is not clear how to write field acquisition calculations based on MRT values measured past the half-sample (Nyquist) rate. A dynamic MRT method has been developed and measured which uses motion to provide enhanced sampling.

Patent
15 Sep 1999
TL;DR: In this article, the authors proposed a new and more advanced method for frequency and optimum sampling phase determination based on analyzing the content of the image to arrive at an optimum value of phase and frequency by directly optimizing image quality.
Abstract: Pixel clock frequency and optimum sampling phase adjustment is an important requirement in Flat panel display monitors (FPDM) with an analog video interface. This invention proposes a new and more advanced method for frequency and optimum sampling phase determination. It is based on analyzing the content of the image to arrive at an optimum value of phase and frequency by directly optimizing image quality. The method differs from existing methods on two counts. First, no assumptions are needed about the precise value of expected frequency. Second, instead of following a two step approach of first determining frequency and then phase, this invention makes possible a single pass phase-frequency optimization.