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Published byAbel Bailey Modified over 9 years ago
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Introduction to D/A and A/D conversion Professor: Dr. Miguel Alonso Jr.
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Outline Analog to Digital Conversion Process Sampling – lowpass and bandpass signals Uniform and non-uniform quantization and encoding Oversampling in A/D D/A conversion: signal recovery The DAC Oversampling in D/A conversion
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Analog to digital conversion process Most signals in nature are in analog form In order for transmission through a digital communication system, they must be sampled Untill now we have seen, PAM, PWM, PPM, and DM DM was the first step towards representing the amplitude of the analog signal ( the intelligence or message we are trying to send) into a binary number for transmission
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Steps for A/D conversion are Bandlimit the signal: anti-aliasing low-pass filter Sample the analog signal into a discrete-time and continuous amplitude signal Convert the amplitude of each signal sample into one of 2 B levels, where B is the number of bits used to represent a sample in the ADC The discrete amplitude levels are represented or encoded into distinct binary words each of length B bits
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Analog input signal – continuous in time and amplitude Sampled Signal – continuos in amplitude, but only defined at discrete points in time. Thus, the signal is zero except at time t=nT ( where T is the sampling period and n is the sample number Digital signal – signal exists only at discrete points in time and at each time point, can only have one of 2 B values. Discrete time and discrete amplitude
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The discrete-time signal and the digital signal can each be represented as a sequence of numbers, x(nT), or simply x(n) where n=0,1,2,3,4…
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Sampling- lowpass and bandpass The sampling theorem: if the highest frequency component in a signal is fmax, then the signal should be sampled at a rate of at least 2*fmax for the samples to describe the signal completely Fs ≥ 2*fmax
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Aliasing and spectra of sampled signals Suppose a signal is sampled at a frequency of 1/T hertz There exists another frequency component with the same set of samples as the original. Thus, the frequency component can be mistaken for the lower frequency component This is aliasing
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Anti-aliasing filtering To reduce the effects of aliasing, sharp cutoff anti-aliasing filters are used to bandlimit the signal Or, the sampling frequency is increased Ideally, the AA filter should remove all frequency components above the fold over frequency Practical filters: stop band attenuation is given by Amin = 20 log (sqrt(1.5) * 2 B ) Where B is the number of bits in the A/D
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Key Equations for A/D Amplitude response of a butterworth filter: where N is the filter order RMS of the input: A/sqrt(2) Quantization Step Size: q = 2*A / 2 B - 1≈ 2*A / 2 B RMS quantization noise: q/(2*sqrt(3)) fs ≥ 2*fmax from computed from the minimum attenuation level Example Problem:
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A to D system with 3 rd Order butterworth AA filter 12-bit ADC with sample and hold Find: the minimum stop band attenuation, Amin, for the AA filter Minimum sampling frequency Fs
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Types of A/D chips
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