By Susskind A.K. (ed.)
These notes from designated in depth summer time courses on Analog-Digital conversion held at MIT from 1956-1957 concentrate on difficulties created while electronic apparatus is associated with actual platforms. A "language" challenge arises, for the language of the information-processing apparatus is electronic and the language of communique within the remainder of the method is almost regularly within the type of electric indications or mechanical displacements analogous the the actual parameters concerned. therefore, there's a desire for units to accomplish the language translation. units taht practice analog-to-digital conversion are referred to as coders, and units that practice digital-to-analog conversion are known as decoders.The subject material is split itno 3 components. the 1st half relates to structures features of electronic info processing that effect the standards for analog-to-digital and digital-to-analog conversion units. within the moment half, a close engineering research and evaluate of numerous conversion units is gifted. The 3rd half is dedicated to a case learn in response to improvement paintings performed on the Servomechanisms Laboratory of the MIT division of electric Engineering.
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Extra info for Analog-digital conversion techniques
The input data stream is denoted as xk, and the output is yk. The “k” subscript is used to identify the sequence of data. For example, xkþ1 follows xk, and xk 1 precedes xk. Often for the purpose of defining a steady state response, we assume that the data streams are infinitely long in time, or that k extends from –1 to þ1. The coefficients are usually static (meaning they do not change over time) and determine the filter’s frequency response. In equation form, the filter could be represented as yk ¼ C0 Á xk þ C1 Á xk 1 þ C2 Á xk 2 þ C3 Á xk 3 þ C4 Á xk 4 It is just the sum of multipliers.
In the digital world, the concepts of resistance and power do not exist. A given signal has specific amplitude, expressed in a digital numerical system (such as signed fractional or integer, for example). Understanding dB increases using the two measurement methods is important. Let’s look at doubling of the amplitude ratio and doubling of the power ratio: 6:02 dBvoltage ¼ 6:02 dBdigital value ¼ 20 Á logð2=1Þ 3:01 dBpower ¼ 10 Á logð2=1Þ This is why shifting a digital signal left 1 bit (multiply by 2) causes a 6 dB signal power increase, and why so often the term 6 dB/bit is used in conjunction with ADCs, DACs, or digital systems in general.
Then the microphone signal is sampled at 8000 Hz, or 8 kHz. All subsequent digital signal processing occurs on this 8 kSPS (kilo-samples per second) signal. That is why if you hear music in the background while on the telephone, the music will sound flat or distorted. Our ears can detect up to about 15 kHz frequencies, and music generally has frequency content exceeding 3600 Hz. But little of this higher frequency content will be passed through the telephone system. In the next chapter, we will start representing our signals and sampling in the frequency (or spectral) domain.
Analog-digital conversion techniques by Susskind A.K. (ed.)