scholarly journals Design of low‐pass and band‐pass infinite impulse response maximally flat stable digital differentiators

2021 ◽  
Author(s):  
Huoping Yi ◽  
Xiaoping Lai
Author(s):  
Gordana Jovanovic Dolecek ◽  
Javier Diaz Carmona

Stearns and David (1996) states that “for many diverse applications, information is now most conveniently recorded, transmitted, and stored in digital form, and as a result, digital signal processing (DSP) has become an exceptionally important modern tool.” Typical operation in DSP is digital filtering. Frequency selective digital filter is used to pass desired frequency components in a signal without distortion and to attenuate other frequency components (Smith, 2002; White, 2000). The pass-band is defined as the frequency range allowed to pass through the filter. The frequency band that lies within the filter stop-band is blocked by the filter and therefore eliminated from the output signal. The range of frequencies between the pass-band and the stop-band is called the transition band and for this region no filter specification is given. Digital filters can be characterized either in terms of the frequency response or the impulse response (Diniz, da Silva & Netto, 2002). Depending on its frequency characteristic, a digital filter is either low-pass, high-pass, band-pass, or band-stop filters. A low-pass (LP) filter passes low frequency components to the output, while eliminating high-frequency components. Conversely, the high-pass (HP) filter passes all high-frequency components and rejects all low-frequency components. The band-pass (BP) filter blocks both low- and high-frequency components while passing the intermediate range. The band-stop (BS) filter eliminates the intermediate band of frequencies while passing both low- and high-frequency components. In terms of their impulse responses digital filters are either infinite impulse response (IIR) or finite impulse response (FIR) digital filters. Each of four types of filters (LP, HP, BP, and BS) can be designed as an FIR or an IIR filter (Ifeachor & Jervis, 2001; Mitra, 2005; Oppenheim & Schafer, 1999).


2015 ◽  
Vol 06 (02) ◽  
pp. 30-48 ◽  
Author(s):  
Emmanuel S. Kolawole ◽  
Warsame H. Ali ◽  
Penrose Cofie ◽  
John Fuller ◽  
C. Tolliver ◽  
...  

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.


2010 ◽  
Vol 17 (2) ◽  
Author(s):  
Eduardo Pinheiro ◽  
Octavian Postolache ◽  
Pedro Girão

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.


Author(s):  
Andrzej Handkiewicz ◽  
Mariusz Naumowicz

AbstractThe paper presents a method of optimizing frequency characteristics of filter banks in terms of their implementation in digital CMOS technologies in nanoscale. Usability of such filters is demonstrated by frequency-interleaved (FI) analog-to-digital converters (ADC). An analysis filter present in these converters was designed in switched-current technique. However, due to huge technological pitch of standard digital CMOS process in nanoscale, its characteristics substantially deviate from the required ones. NANO-studio environment presented in the paper allows adjustment, with transistor channel sizes as optimization parameters. The same environment is used at designing a digital synthesis filter, whereas optimization parameters are input and output conductances, gyration transconductances and capacitances of a prototype circuit. Transition between analog s and digital z domains is done by means of bilinear transformation. Assuming a lossless gyrator-capacitor (gC) multiport network as a prototype circuit, both for analysis and synthesis filter banks in FI ADC, is an implementation of the strategy to design filters with low sensitivity to parameter changes. An additional advantage is designing the synthesis filter as stable infinite impulse response (IIR) instead of commonly used finite impulse response (FIR) filters. It provides several dozen-fold saving in the number of applied multipliers.. The analysis and synthesis filters in FI ADC are implemented as filter pairs. An additional example of three-filter bank demonstrates versatility of NANO-studio software.


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.


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