Amplitude Distributions of Cochlear Microphonic Response to an Acoustic Sinusoid in Noise

1968 ◽  
Vol 11 (1) ◽  
pp. 63-76
Author(s):  
Donald C. Teas ◽  
Gretchen B. Henry

The distributions of instantaneous voltage amplitudes in the cochlear microphonic response recorded from a small segment along the basilar membrane are described by computing amplitude histograms. Comparisons are made between the distributions for noise and for those after the addition to the noise of successively stronger sinusoids. The amplitudes of the cochlear microphonic response to 5000 Hz low-pass noise are normally distributed in both Turn I and Turn III of the guinea pig’s cochlea. The spectral composition of the microphonic from Turn I and from Turn III resembles the output of band-pass filters set at about 4000 Hz, and about 500 Hz, respectively. The normal distribution of cochlear microphonic amplitudes for noise is systematically altered by increasing the strength of the added sinusoid. A decrease of three percent in the number of small amplitude events (±1 standard deviation) in the cochlear microphonic from Turn III is seen when the rms voltage of a 500 Hz sinusoid is at −18 dB re the rms voltage of the noise (at the earphone). When the rms of the sinusoid and noise are equal, the decrease in small voltages is about 25%, but there is also an increase in the number of large voltage amplitudes. Histograms were also computed for the output of an electronic filter with a pass-band similar to Turn III of the cochlea. Strong 500 Hz sinusoids showed a greater proportion of large amplitudes in the filter output than in CM III . The data are interpreted in terms of an anatomical substrate.

Author(s):  
K.R. Shankarkumar ◽  
Gokul Kumar

: Filtering is an important step in the field of image processing to suppress the required parts or to remove any artifacts present in it. There are different types of filters like low pass, high pass, Band pass, IIR, FIR and adaptive filtering etc.., in these filters adaptive filters is an important filter because it is used to remove the noisy signal and images. Least Mean Square filter is a type of an adaptive filtering which is used to remove the noises present in the medical images. The working of LMS is based on the minimization of the difference between the error images using a closed loop feedback. Therefore presented technique called as Q-CSKA. Here the CSKA performs its operation in stages which is based on the nucleus stage. In the traditional CSKA the nucleus stage is depend on the parallel prefix adder in this work it is replaced by the QCA adder. The QCA adder utilizes the less area compared to PPA and it can be realized in Nanometer range also. For multiplexers, And OR Invert, OR and Invert logic is used to reduce the area and delay. Due to these advantages of the QCA, AOI-OAI logic the proposed method outperformed the LMS implementation in area, power, and accuracy and delay, this based five type image noise of medical pictures related to the best technique is out comes. It helps to medicinal practitioner to resolve the symptoms of patient with ease.


2007 ◽  
Vol 16 (04) ◽  
pp. 507-516 ◽  
Author(s):  
SHAHRAM MINAEI ◽  
ERKAN YUCE

In this paper, a universal current-mode second-order active-C filter for simultaneously realizing low-pass, band-pass and high-pass responses is proposed. The presented filter employs only three plus-type second-generation current-controlled conveyors (CCCII+s). This filter needs no critical active and passive component matching conditions and no additional active and passive elements for realizing high output impedance low-pass, band-pass and high-pass characteristics. The angular resonance frequency (ω0) and quality factor (Q) of the proposed resistorless filter can be tuned electronically. To verify the theoretical analysis and to exhibit the performance of the proposed filter, it is simulated with SPICE program.


2005 ◽  
Vol 14 (01) ◽  
pp. 159-164 ◽  
Author(s):  
SUDHANSHU MAHESHWARI ◽  
IQBAL A. KHAN

A novel voltage-mode universal filter employing only two current differencing buffered amplifiers (CDBAs) is proposed. The filter uses four inputs and single output to realize six responses, viz. low-pass, high-pass, inverting band-pass, noninverting band-pass, band-elimination, and all-pass through input selection with independent pole-Q control. Computer simulation results using SPICE are also given to verify the theory.


Author(s):  
Emre Cancioglu ◽  
Gokberk Cakiroglu ◽  
Alkim Gokcen ◽  
Yilmaz Sefa Altanay

This study provides design and implementation of four digital filters (low pass, high pass, band pass and band stop) for ECG (electrocardiogram) data on FPGA with MATLAB by a serial communication. The study is conducted with using ECG data which is obtained from PhysioBank Database platform. SysGen (System Generator for DSP) which is a toolbox for MATLAB is used for designing and implementing the digital filters. The aim of the study is to perform four different digital filters with various blocks on the SysGen Toolbox. The study then examines the results of four different digital filters.


Author(s):  
Umar Mohammad ◽  
Fang Tang ◽  
Shu Zhou ◽  
Mohd Yusuf Yasin

A new study imitating the design and implementation of single-input–single-output (SISO) filters as bilateral filters has been presented in this paper. Second generation current controlled current conveyor (CCCII), being a popular low power active element was considered for the realization of the proposed design. Complete design, analysis and implementation of the voltage mode SISO filter was done using only two CCCII’s and two passive parasitic components. The striking feature of this work is that the proposed design can be made to work at either the input node or the output node, as well as in the cases; the change of direction changes the filter into an inverse filter and buffer filter. Basic filter applications like low-pass, high-pass, band-pass and band-stop were aimed to check the uniformity of the proposed design at different frequencies. Results perceived from the simulation study were fare enough on both the side nodes of the proposed design. Categorically, the circuit can be aimed to work in lieu of a filter transceiver. The consistency of the circuit was analyzed by the nodal analysis. Whereas the working performance was enormously analyzed and evaluated during the simulation analysis. The proposed design was simulated in HSPICE tool to exhibit and exploit the delivery, using the 45[Formula: see text]nm predictive technology model (PTM) parameters, with [Formula: see text][Formula: see text]V rail to rail voltages. Maximum power consumption of the circuit is around 138.5[Formula: see text][Formula: see text]W. Finally, the design was also implemented in Cadence Virtuoso using 40[Formula: see text]nm SMIC parameters.


2016 ◽  
Vol 25 (12) ◽  
pp. 1650154 ◽  
Author(s):  
Ahmet Abaci ◽  
Erkan Yuce

In this paper, two new second-order voltage-mode universal filters are proposed. Both of the proposed filters use only two differential voltage current conveyors (DVCCs), four resistors and two grounded capacitors which are advantageous from integrated circuit technology point of view. They can simultaneously provide second-order low-pass, high-pass, band-pass, notch and all-pass responses. They offer orthogonal control of angular resonance frequency and quality factor. However, they have a single matching condition for only all-pass responses. A number of simulations based on SPICE program are accomplished in order to demonstrate the performance of both filters.


2011 ◽  
Vol 20 (03) ◽  
pp. 549-555 ◽  
Author(s):  
A. K. SINGH ◽  
R. SENANI ◽  
D. R. BHASKAR ◽  
R. K. SHARMA

A number of configurations for realizing voltage-mode (VM) biquads using op-amps and OTAs have been presented in the literature, however, none of these provide the following desirable properties simultaneously: (i) realizability of all the five standard filters (namely; low pass, high pass, band pass, band stop and all pass), (ii) tunability of all the three filter parameters (namely; ω0, bandwidth or Q0 and gain) and (iii) not requiring any realization condition in any of the five filter responses. This paper presents a new configuration which does possess all the above mentioned desirable properties simultaneously while using only two internally-compensated type op-amps and a reasonable number of OTAs. The workability of the new configuration has been demonstrated by SPICE simulations based upon CMOS Op-amp and CMOS OTAs.


2008 ◽  
Vol 17 (01) ◽  
pp. 55-66 ◽  
Author(s):  
A. G. RADWAN ◽  
A. M. SOLIMAN ◽  
A. S. ELWAKIL

Traditional continuous-time filters are of integer order. However, using fractional calculus, filters may also be represented by the more general fractional-order differential equations in which case integer-order filters are only a tight subset of fractional-order filters. In this work, we show that low-pass, high-pass, band-pass, and all-pass filters can be realized with circuits incorporating a single fractance device. We derive expressions for the pole frequencies, the quality factor, the right-phase frequencies, and the half-power frequencies. Examples of fractional passive filters supported by numerical and PSpice simulations are given.


2016 ◽  
Vol 07 (03) ◽  
pp. 83-99 ◽  
Author(s):  
Raj Senani ◽  
Abdhesh Kumar Singh ◽  
Ashish Gupta ◽  
Data Ram Bhaskar

2015 ◽  
Vol 39 (4) ◽  
pp. 591-597
Author(s):  
Yao Shu-Nung

Abstract A hybrid method is presented for the integration of low-, mid-, and high-frequency driver filters in loud-speaker crossovers. The Pascal matrix is exploited to calculate denominators; the locations of minimum values in frequency magnitude responses are associated with the forms of numerators; the maximum values are used to compute gain factors. The forms of the resulting filters are based on the physical meanings of low-pass, band-pass, and high-pass filters, an intuitive idea which is easy to be understood. Moreover, each coefficient is believed to be simply calculated, an advantage which keeps the software-implemented crossover running smoothly even if crossover frequencies are being changed in real time. This characteristic allows users to efficiently adjust the bandwidths of the driver filters by subjective listening tests if objective measurements of loudspeaker parameters are unavailable. Instead of designing separate structures for a low-, mid-, and high-frequency driver filter, by using the proposed techniques we can implement one structure which merges three types of digital filters. Not only does the integration architecture operate with low computational cost, but its size is also compact. Design examples are included to illustrate the effectiveness of the presented methodology


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