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Explain the Limiting the Spectrum
This low-pass filter is known an anti-aliasing filter. These filters are originated at the input to most applications. Since they are required before the signal is sampled, they are approximately always analog filters. It is theoretically possible to make use of the DSP device to remove the aliasing, but this would need a high sample rate. Actually, this is rarely done as it would consume too much of the processing power of the DSP, which we need to carry out other tasks. The ideal anti-aliasing filter characteristics are displayed in the lower half of the above diagram. It is flat along with gain A over the pass band and zero at any other point.
An ideal filter would as well have a linear phase response. Variations from this ideal may have an important effect on performance. For instance, in control applications, a badly planed filter may mean that the system spends all of its time trying to compensate for the filter, rather than trying to keep a constant motor speed. In audio applications, a poor phase response can lead to harmonic distortion and degradation that is frequently audible. For audio applications, commercially available switched-capacitor filters are easy to use and offer a reasonably linear response.
Q. Sketch the timing diagram for a 4-bit ripple counter which uses T flip-flops.
Machine Language ( First Generation Language) As we have already discussed that all digital circuits can identify only two states which can be specified by 0 and i henc
A linear circuit component is one that has a linear I V characteristic
sketching of wire 1 is charged and wire 2 uncharged
how to design a 32:1 multiplexer using two 16:1 multiplexers and a 2:1 multiplexer?
Supercomputer The largest fastest and most powerful are called supercomputers. Supercomputers are mainly used for weather forecasting remote sensing image processing, biom
Prepare a User Requirement Specification for the monitor and alarm system. User Requirements Specification should be written in a clear and unambiguous manner to state: • The
Using the audiorecorder() function in MATLAB, make a recording of your voice. Use a sample rate of 8kHz, 16 bits per sample, and aim to record around 2 seconds worth. Quantize th
i want to prodused domestic power factor correction like industial
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