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Showing papers on "Low-pass filter published in 1992"


Journal ArticleDOI
Bram Nauta1
TL;DR: In this article, a linear, tunable integrator for very high-frequency integrated filters can be made, which has good linearity properties and non-dominant poles in the gigahertz range owing to the absence of internal nodes.
Abstract: CMOS circuits for integrated analog filters at very high frequencies, based on transconductance-C integrators, are presented. First a differential transconductance element based on CMOS inverters is described. With this circuit a linear, tunable integrator for very-high-frequency integrated filters can be made. This integrator has good linearity properties and nondominant poles in the gigahertz range owing to the absence of internal nodes. The integrator has a tunable DC gain, resulting in a controllable integrator quality factor. Experimental results of a VHF CMOS transconductance-C low-pass filter realized in a 3- mu m CMOS process are given. Both the cutoff frequency and the quality factors can be tuned. The cutoff frequency was tuned from 22 to 98 MHz and the measured filter response is very close to the ideal response of the passive prototype filter. Furthermore, a novel circuit for automatically tuning the quality factors of integrated filters built with these transconductors is described. >

674 citations


Journal ArticleDOI
TL;DR: Among the topics discussed are sampling in multiple dimensions, multidimensional perfect reconstruction filter banks, the two-channel case in several dimensions, the synthesis of multiddimensional filter Banks, and the design of compactly supported wavelets.
Abstract: New results on multidimensional filter banks and their connection to multidimensional nonseparable wavelets are presented. Among the topics discussed are sampling in multiple dimensions, multidimensional perfect reconstruction filter banks, the two-channel case in multiple dimensions, the synthesis of multidimensional filter banks, and the design of compactly supported wavelets. >

481 citations


Journal ArticleDOI
TL;DR: In this paper, a control procedure that uses time delay control to achieve input/output linearization of a class of nonlinear systems is presented, which is characterized by a simple algorithm and enhanced robustness properties in comparison with current control algorithms.
Abstract: A control procedure that uses Time Delay Control to achieve input/output linearization of a class of nonlinear systems is presented. The control system is characterized by a simple algorithm and enhanced robustness properties in comparison with current control algorithms. The paper first reviews the fundamentals of input/output linearization. The use of Time Delay Control is then shown to result in an exact linear system for sufficiently small delay time. Modified controllers for systems with a low-pass filter are also investigated. Simulation results show that the algorithm works well with measurement noise. The controller is also tested on a single-link flexible arm to show the effectiveness of the simple algorithm in the control of complicated systems.

192 citations


Journal ArticleDOI
TL;DR: A set of time-domain conditions for reconstruction which can be used directly in a filter bank design procedure is derived, which allows for the design of many useful banks.
Abstract: The authors present a new time-domain approach for the analysis and design of a broad class of general analysis/synthesis systems based on M-band filter banks. They derive a set of time-domain conditions for reconstruction which can be used directly in a filter bank design procedure. The general and unrestricted nature of this framework allows for the design of many useful banks. In addition to the complete derivation of the time-domain conditions, they also describe the associated filter bank design procedure and a number of design examples are included. >

180 citations


Patent
15 Oct 1992
TL;DR: In this paper, a preamplifier is used to attenuate currents near the resonance frequency of the imager, and a low pass filter is provided to remove induced radio frequency components from the signal.
Abstract: Magnetic resonance imaging hardware (A) defines a patient receiving region (20) that is surrounded by a bore liner (22). A socket (50) is mounted in the bore liner with an appropriate receptacle for receiving a standard plug (52) of a conventional pulse oximetry system. Conventional pulse oximetry systems include a sensor unit (54) connected with a cable (56) having the plug (52) at one end thereof. A notch filter (62) attenuates currents near the resonance frequency of the imager. A preamplifier (60) amplifies signals from the sensor unit. Within the shielding (66) of the preamplifier, a low pass filter (68) is provided to remove induced radio frequency components from the preamplified sensor unit signal. A radio frequency filter (70) mounted at the shield of the shielded room (B) prevents radio frequency signals from reaching an exterior processing and display unit (E) and prevents radio frequency signals from a clock (72) of the processing and display unit from being conveyed into the shielded room (B). The processing and display unit processes the signal received from the preamplifier to generate a pulse rate display (78) and a blood oxygen concentration display (80).

143 citations


Book
30 Sep 1992
TL;DR: In this article, the authors propose a transductor design for very high frequency filter synthesis for high frequency transductors, which is based on the idea of non-idealities.
Abstract: Preface. 1. Introduction. 2. Filter Synthesis for (very) High Frequencies. 3. Effect of Non-Idealities. 4. Transductor Design. 5. Tuning. 6. Filter Realizations. 7. Conclusions. References. Subject Index.

138 citations


Journal ArticleDOI
TL;DR: In this article, the design of quadrature mirror filter (QMF) banks whose analysis and synthesis filters have linear phase is considered and an analytical solution formula is obtained, leading to a very efficient procedure.
Abstract: The design of quadrature mirror filter (QMF) banks whose analysis and synthesis filters have linear phase is considered. Because the design problem in the frequency domain is a highly nonlinear optimization problem, a linearization technique is proposed. An analytical solution formula is obtained, leading to a very efficient procedure. Computer simulations show that the design technique achieves better results in fewer iterations than conventional approaches when starting at the same preset initial guess. Moreover, the technique produces almost the same good results in six iterations if it starts at a better initial guess compared to the preset initial guess. By incorporating the technique with a weighted least squares, (WLS) algorithm, the design of QMF banks whose overall reconstruction error is minimized in the minimax sense over the entire frequency band is facilitated. Computer simulations for illustration and comparison are provided. >

127 citations


Journal ArticleDOI
TL;DR: In this article, a general class of current amplifier-based biquadratic filter circuits capable of realizing arbitrary filter functions including the low-pass, high-pass and bandpass transfer functions is presented.
Abstract: A general class of current amplifier-based biquadratic filter circuits capable of realizing arbitrary filter functions including the low-pass, high-pass, and bandpass transfer functions is presented. These realizations are derived from a class of well-known low sensitivity single amplifier biquadratic (SAB) filter circuits using the principle of adjoint networks. The salient features of the proposed circuits are that they are synthesized using the same procedure as their op-amp-based SAB circuit counterparts, and they possess the same sensitivities to component variations as the original SAB circuits. However, it is demonstrated experimentally that unlike op-amp-based SAB realizations whose effective operating bandwidth is much less than the unity-gain bandwidth of the op-amp, these current-based filter circuits are effective over the entire bandwidth of the current amplifier. >

117 citations


Journal ArticleDOI
TL;DR: In this paper, a self-tuning continuous-time RC filter with high-linearity self-tuneable capacitors is presented. Butler et al. used switchable arrays of highly linear double-polysilicon capacitors in an active RC filter structure, resulting in tunable filters with very low signal distortion.
Abstract: High-linearity self-tuning continuous-time filters, fabricated in a standard 1.6- mu m 5-V CMOS process, are presented. Frequency control is achieved using switchable arrays of highly linear double-polysilicon capacitors in an active RC filter structure, resulting in tunable filters with very low signal distortion. One filter, a Tow-Thomas biquad, exhibits dynamic range and signal linearity of typically 91 dB. Another smaller implementation, a Sallen and Key filter, attains >or=76 dB. Cutoff frequency response is maintained to an accuracy of around +or-5%. >

112 citations


Patent
31 Mar 1992
TL;DR: In this paper, a drop detector and a low pass filter are used to detect drops passing through an optical sensing path between a detector and at least one light source, where a capacitor is connected between the detector and an amplifier to block the DC component of the output signal.
Abstract: A drop detector system and method are provided for a drop detector of the type including a drop chamber and an electro-optical sensor. The detector system detects drops passing through an optical sensing path between a detector and at least one light source. In response to the detection of a drop passing through the optical path, the detector produces an output signal. A capacitor is connected between the detector and an amplifier to block the DC component of the output signal. After amplification, the signal is passed through a low pass filter to further block signals caused by undesirable factors. The cutoff frequency of the low pass filter is controlled by a microprocessor that controls the pump that pumps liquid from the drop chamber. The detector and light source or sources are arranged to detect drops falling in the drop chamber at virtually any angle and in virtually any ambient light condition.

106 citations


Journal ArticleDOI
TL;DR: In this article, a second-order low-pass filter using the new transconductor realized in a 2- mu m BiCMOS technology is reported, and the cutoff frequency f/sub 0/ of the cell is tunable in the range of 8-32 MHz.
Abstract: A BiCMOS fully differential transconductor based on MOS transistors operating in the linear region is presented. The circuit has an equivalent nondominant pole located above 1.5 GHz. This makes it suitable for high-frequency continuous-time filters. A second-order low-pass filter using the new transconductor realized in a 2- mu m BiCMOS technology is reported. The cutoff frequency f/sub 0/ of the cell is tunable in the range of 8-32 MHz and the quality factor is 2. The filter THD stays lower than -40 dB for an output signal up to 3.2 V/sub p-p/ at 5-MHz frequency. The area of the cell is 0.322 mm/sup 2/ and the power consumption (with f/sub 0/=25 MHz) is 30 mW with a single 5-V power supply. >

Journal ArticleDOI
TL;DR: A bipolar seventh-order 0.05 degrees equiripple linear phase (constant group delay) transconductance-capacitor (g/sub m/-C) low-pass filter with a cutoff frequency (f/sub c/) tunable between 2 and 10 MHz is presented.
Abstract: A bipolar seventh-order 0.05 degrees equiripple linear phase (constant group delay) transconductance-capacitor (g/sub m/-C) low-pass filter with a cutoff frequency (f/sub c/) tunable between 2 and 10 MHz is presented. Programmable equalization up to 9 dB at f/sub c/ is also provided. Total harmonic distortion at 2 V/sub p-p/ is less than 1%, with a dynamic range equal to 49 dB. Nominal power consumption from a single 5-V supply is 135 mW. The circuit also has a low-power mode ( >

Journal ArticleDOI
TL;DR: It is concluded that error feedback is a very powerful and versatile method for cutting down the quantization noise in any classical infinite impulse response (IIR) filter implemented as a cascade of second-order direct form sections.
Abstract: The problem of solving the optimal (minimum-noise) error feedback coefficients for recursive digital filters is addressed in the general high-order case. It is shown that when minimum noise variance at the filter output is required, the optimization problem leads to set of familiar Wiener-Hopf or Yule-Walker equations, demonstrating that the optimal error feedback can be interpreted as a special case of Wiener filtering. As an alternative to the optimal solution, the formulas for suboptimal error feedback with symmetric or antisymmetric coefficients are derived. In addition, the design of error feedback using power-of-two coefficients is discussed. The efficiency of high order error feedback is examined by test implementations of the set of standard filters. It is concluded that error feedback is a very powerful and versatile method for cutting down the quantization noise in any classical infinite impulse response (IIR) filter implemented as a cascade of second-order direct form sections. The high-order schemes are attractive for use with high-order direct form sections. >

Journal ArticleDOI
TL;DR: The authors present two approaches to the design of two-channel perfect-reconstruction linear-phase finite-impulse-response (FIR) filter banks, and covers the design for all parts of linear phase perfect reconstruction constraint equations.
Abstract: The authors present two approaches to the design of two-channel perfect-reconstruction linear-phase finite-impulse-response (FIR) filter banks. Both approaches analyze and design the impulse responses of the analysis filter bank directly. The synthesis filter bank is then obtained by simply changing the signs of odd-order coefficients in the analysis filter bank. The approach deals with unequal-length filter banks. By designing the lower length filters first, one can take advantage of the fact that the number of variables for designing the higher length filters is more than the number of perfect-reconstruction constraint equations. The second approach generalizes the first, and covers the design for all parts of linear phase perfect reconstruction constraint equations. >

Journal ArticleDOI
TL;DR: In this article, a finite impulse response (FIR) filter that can synthesize any fractional sample delay by a nonlinear interpolation technique is presented, and analytically closed-form solutions for the tap weights of such an FIR filter and their frequency responses are also presented.
Abstract: A finite impulse response (FIR) filter that can synthesize any fractional sample delay by a nonlinear interpolation technique is presented. Analytically closed-form solutions for the tap weights of such an FIR filter and their frequency responses are also presented. >

Journal ArticleDOI
TL;DR: It is proposed to specify a filter only in terms of upper and lower limits on the response, find the shortest filter length which allows these constraints to be met, and then find a filter of that order which is farthest from theupper and lower constraint boundaries in a minimax sense.
Abstract: It is proposed to specify a filter only in terms of upper and lower limits on the response, find the shortest filter length which allows these constraints to be met, and then find a filter of that order which is farthest from the upper and lower constraint boundaries in a minimax sense. The simplex algorithm for linear programming is used to find a best linear-phase FIR filter of minimum length, as well as to find the minimum feasible length itself. The simplex algorithm, while much slower than exchange algorithms, also allows the incorporation of more general kinds of constraints, such as concavity constraints (which can be used to achieve very flat magnitude characteristics). Examples are given to illustrate how the proposed and common approaches differ, and how the proposed approach can be used to design filters with flat passbands, filters which meet point constraints, minimum phase filters, and bandpass filters with controlled transition band behavior. >

PatentDOI
TL;DR: In this paper, an apparatus for detecting a polishing endpoint during chemical-mechanical planarization/polishing of a wafer senses an acoustic wave generated by rubbing contact between a polish pad and a hard surface underlying a softer material being removed.
Abstract: An apparatus for detecting a polishing endpoint during chemical-mechanical planarization/polishing of a wafer senses an acoustic wave generated by rubbing contact between a polish pad and a hard surface underlying a softer material being removed. The apparatus includes a transducer for converting the acoustic wave energy in the range of 30 to 100 Hertz into an audio signal. The audio signal is processed by a low pass cutoff filter to remove high frequency noise. The filtered audio signal is supplied to a phase lock loop to detect a predetermined audio frequency and, in response, provide a logic signal to an integrator. The integrator integrates the logic signal over time to eliminate transient noise spikes, and supplies a detection signal only upon receiving the logic signal for a predetermined period. The detection signal starts a counter to provide a predetermined overpolishing time prior to termination of polishing operations.

Journal ArticleDOI
TL;DR: A resolver-to-digital (R/D) conversion method in which a bang-bang type phase comparator is used for fast tracking in which the low-pass filter needed to reject carrier signal and noise is eliminated from the R/D conversion loop.
Abstract: A resolver-to-digital (R/D) conversion method in which a bang-bang type phase comparator is used for fast tracking is proposed. The low-pass filter needed to reject carrier signal and noise is eliminated from the R/D conversion loop. Instead, two prefilters outside the R/D conversion loop take the role of the low-pass filter, resulting in a fast and accurate tracking R/D converter. Some simulation and experimental results and a mathematical performance analysis are presented to demonstrate the superior tracking performance. >

Journal ArticleDOI
01 Dec 1992
TL;DR: In this article, a real-coded genetic algorithm is used to design optimal multilayer filters for low-pass and high-pass optical filters, operating between practical terminal conditions.
Abstract: A novel approach for designing optimal multilayer filters based on a real-coded genetic algorithm is presented. Given the total number of layers in the filter, as well as the electrical properties of the materials constituting each layer, the algorithm iteratively constructs multilayers whose frequency response closely matches a desired frequency response. In contrast to existing iterative techniques, this method does not require a preliminary design using classical techniques. Also, the design procedure is independent of the nature of the multilayer as well as the characteristics of the incident and substrate media. The algorithm is applied to the design of various lowpass and high-pass optical filters, operating between practical terminal conditions. The performance of the resulting designs matches or improves on that for filters that were synthesised using semiclassical techniques.

Patent
14 Jul 1992
TL;DR: In this paper, a step frequency ground penetrating radar system is described, comprising an RF signal generating section capable of producing stepped frequency signals in spaced and equal increments of time and frequency over a preselected bandwidth which serves as a common source for both a transmit portion and a receive portion of the system.
Abstract: A stepped frequency ground penetrating radar system is described comprising an RF signal generating section capable of producing stepped frequency signals in spaced and equal increments of time and frequency over a preselected bandwidth which serves as a common RF signal source for both a transmit portion and a receive portion of the system. In the transmit portion of the system the signal is processed into in-phase and quadrature signals which are then amplified and then transmitted toward a target. The reflected signals from the target are then received by a receive antenna and mixed with a reference signal from the common RF signal source in a mixer whose output is then fed through a low pass filter. The DC output, after amplification and demodulation, is digitized and converted into a frequency domain signal by a Fast Fourier Transform. A plot of the frequency domain signals from all of the stepped frequencies broadcast toward and received from the target yields information concerning the range (distance) and cross section (size) of the target.

Patent
30 Oct 1992
TL;DR: In this article, an improved active power line conditioner is described, in which a series inverter is controlled by a series filter controller which performs synchronous transformations on a load current to generate a parallel filter feedforward signal corresponding to the harmonic ripple components of the load current.
Abstract: An improved active power line conditioner is disclosed. A series inverter is controlled by a series filter controller which performs synchronous transformations on a load current to generate a series filter feedforward signal corresponding to the fundamental components of the load current. The series filter controller also generates a series filter reference signal corresponding to a negative sequence fundamental output voltage. The series filter feedforward signal and the series filter reference signal are combined to form a series filter compensation signal. The series filter compensation signal is applied to the series inverter to generate sinusoidal input currents, with negative sequence fundamental output voltage compensation, for a non-linear load. A parallel inverter is controlled by a parallel filter controller which performs synchronous transformations to generate a parallel filter feedforward signal corresponding to the harmonic ripple components of the load current. The parallel filter controller also generates a parallel filter reference signal corresponding to a negative sequence fundamental source current. The parallel filter feedforward signal and the parallel filter reference signal are combined to form a parallel filter compensation signal. The parallel filter compensation signal is applied to the parallel inverter to generate sinusoidal voltages, with source current negative sequence fundamental compensation, for the non-linear load.

Journal ArticleDOI
TL;DR: In this paper, the sliding fast Fourier transform (FFT) filter bank has an exceedingly low complexity of one multiplication per channel per sample, however, its frequency selectivity and passband response are poor.
Abstract: The sliding fast Fourier transform (FFT) filter bank has an exceedingly low complexity of one multiplication per channel per sample. However, its frequency selectivity and passband response are poor. It is shown that the sliding FFT filter bank is in fact a particular member of a new family of fast filter banks (FFBs). In the case of FFT, each cluster of butterflies can in fact be derived from a pair of complementary two-tap (i.e. first-order) prototype FIR filters. The poor selectivity and degraded passband response of the FFT filter bank is a direct consequence of the poor frequency response of the prototype first-order filter. It is shown that by increasing the order of the prototype filters, it is possible to implement a filter bank with arbitrarily good selectivity and flat passband response. The FFB retains the low-complexity feature of the FFT. Because of its very much improved frequency response characteristics, the FFB be suitable for use in many applications where the FFT filter bank is unsuitable. >

Journal ArticleDOI
TL;DR: A new method involves designing a finite-impulse-response (FIR) filter satisfying the given frequency response specifications and subsequently obtaining a significantly lower order IIR filter using model reduction based on impulse-response gramians.
Abstract: A new method for the design of a linear-phase infinite-impulse-response (IIR) filter is presented. It involves designing a finite-impulse-response (FIR) filter satisfying the given frequency response specifications and subsequently obtaining a significantly lower order IIR filter using model reduction based on impulse-response gramians. The general outline of the method and a brief overview of the existing linear-phase FIR filter design and model-reduction techniques are presented. The impulse-response gramian and the model-reduction algorithm used are presented. The method is illustrated by design examples and is compared with other methods for the design of linear-phase IIR filters using equalizers. >

Journal ArticleDOI
P.W. Wong1
TL;DR: The author suggests an FIR filter structure where both the input signal and the impulse response are encoded using sigma-delta modulation, and the convolution can be obtained by summing a collection of binary or ternary numbers, depending on the quantizer in the sigma/delta modulator.
Abstract: The author suggests an FIR filter structure where both the input signal and the impulse response are encoded using sigma-delta modulation. As a result, the convolution can be obtained by summing a collection of binary or ternary numbers, depending on the quantizer in the sigma-delta modulator. This filter structure can accept analog input signals directly and can be built using modular hardware. The overall error spectrum due to the encoding/decoding processes is derived, and the implementation issues of this filter structure are discussed. An example of its performance with different input signals is provided. >

Patent
27 Aug 1992
TL;DR: In this paper, a digital adaptive finite impulse response filter circuit for a PR4,ML sampled data channel including an analog-to-digital sampler for providing raw digital samples of data to the filter circuit and a sampled data detector for detecting coded data from conditioned digital samples received from the filter circuits.
Abstract: A digital adaptive finite impulse response filter circuit is provided for a PR4,ML sampled data channel including an analog to digital sampler for providing raw digital samples of data to the filter circuit and a sampled data detector for detecting coded data from conditioned digital samples received from the filter circuit. The filter circuit comprises a multi-tap transversal filter structure wherein each tap is connected to receive a selected coefficient, a source of a plurality of coefficients for each tap, a coefficient selector connected to the source to receive the plurality of coefficients and to provide a selected coefficient to each tap of the transversal filter structure, and a control circuit for controlling the coefficient selector and the source for providing each of the selected coefficients to a corresponding tap of the transversal filter structure. Training and adaptation methods and circuits for adapting the filter structure are also described.

Patent
09 Jul 1992
TL;DR: In this article, a main adaptive digital filter and a sub-adaptive digital filter are provided, and these two adaptive digital filters share a filter coefficient to be controlled, on the side of the main adaptive filter, the shared coefficient is updated so that the difference between the output and a desired response is minimized.
Abstract: A main adaptive digital filter and a sub adaptive digital filter are provided, and these two adaptive digital filters share a filter coefficient to be controlled. On the side of the main adaptive digital filter, the shared coefficient is updated so that the difference between the output and a desired response is minimized and on the side of the subadapted digital filter, the above-stated shared filter coefficient is updated so that the output is minimized. A prescribed limitation is given to the frequency characteristic of a filter coefficient to be adapted, by treating the input of the sub adaptive digital filter as a signal weighted on the frequency or a noise having its band limited with respect to the input signal or the output signal of the main adaptive digital filter, and coefficient updating control is conducted so that the coefficient will not go beyond the limitation.

Proceedings ArticleDOI
10 May 1992
TL;DR: In this paper, the authors proposed a beamforming method using FIR (finite impulse response) fan filters for wideband beamforming using a linear array of sensors, which is based on two-dimensional filtering for the outputs of the sensors.
Abstract: A novel approach to wideband beamforming using FIR (finite impulse response) fan filters is presented. The method is based on two-dimensional filtering for the outputs of a linear array of sensors. To beamform in the broadside of the array, a symmetrical 90 degrees /n filter is employed. The filter in which the angle of fan is more acute can form the narrower beam over the full band while the order of the filter becomes higher. To beamform in the direction along the array, an asymmetrical 45 degrees fan filter is used. As aliasing occurs in this beamforming, the band which is available is limited a little in width. A general implementation of the FIR fan filter beamformer is given as the parallel connection of one-dimensional FIR filters. >

Journal ArticleDOI
TL;DR: In this paper, several techniques for the design of high-performance fully differential continuous-time filters are introduced, and they are used for a fourth-order doubly terminated low-pass filter.
Abstract: In this paper, several techniques for the design of high-performance fully differential continuous-time filters are introduced. These techniques are used for the design of a fourth-order doubly terminated low-pass filter. The filter is based on operational transconductance amplifier (OTA) resistor-capacitor (RC) building blocks. For the voltage-to-current transducer a linearized OTA with a novel common-mode feedback (CMFB) system is employed. The CMFB takes advantage of the direct connection of the OTA's. For the control of the filter ripple, a low-distortion floating resistor with low sensitivity to transistor mismatches is used

PatentDOI
TL;DR: In this article, a method and apparatus for providing a differential microphone with a desired frequency response is described, which is provided by operation of a filter, having an adjustable frequency response, coupled to the microphone.
Abstract: A method and apparatus for providing a differential microphone with a desired frequency response are disclosed. The desired frequency response is provided by operation of a filter, having an adjustable frequency response, coupled to the microphone. The frequency response of the filter is set by operation of a controller, also coupled to the microphone, based on signals received from the microphone. The desired frequency response may be determined based upon the distance between the microphone and a source of sound, and may comprise both a relative frequency response and absolute output level. The frequency response of the filter may comprise the substantial inverse of the frequency response of the microphone to provide a flat response. Furthermore, the filter may comprise a Butterworth filter.

Journal ArticleDOI
TL;DR: The electroretinogram (ERG) first harmonics which varied as a function of adaptation level, mean illuminance, and modulation depth was observed, and a minimum of 3 parallel mechanisms operating at 3 frequency regions within the ERG were identified.
Abstract: We observed multiple maxima and minima in the electroretinogram (ERG) first harmonics which varied as a function of adaptation level, mean illuminance, and modulation depth Based on differences in response characteristics we identified a minimum of 3 parallel mechanisms operating at 3 frequency regions within the ERG: a low frequency region (less than 10 Hz), a midfrequency region (centered near 20 Hz), and a high frequency region (centered near 40 Hz) The low frequency region was observed both in dark- and light-adapted conditions, was basically linear, and showed nonlinear behavior only at high contrasts in either light or dark adaptation It may be modeled as a low pass filter The midfrequency region was clearly observed only above cone threshold, was nonlinear at low contrasts, and may be modeled as a linear filter followed by an essential nonlinearity The high frequency region was observed under high levels of light adaptation and also was nonlinear at low contrast It may be modeled also as a linear filter followed by an essential nonlinearity