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Passing Voice DigitallyPablo Oropin

BS in Computer EngineeringBS in Computer Science

Harding University

poropin@harding.edu

ABSTRACTThe purpose of this paper is to describe how voice is passeddigitally across a given network: telephone or internet.

Categories and Subject DescriptorsE.4 [Coding and Information Theory]: Formal model ofcommunication.

General TermsStandardization, and Theory.

KeywordsAmplitude, Frequency, A/D Converter, Sampling, NyquistTheorem, Noise, Bandwidth, T1 line, Bit

1. INTRODUCTIONBecause passing voice by using analog signals was a problem forthe network suppliers and the telcos alike, both of them turned to

passing voice digitally. Digitizing voice requires the integration of

many devices that go from a simple analog transmitter to a morecomplex analog-to-digital converter. Even though passing voicedigitally requires more equipment across a given network, manyadvantages come from doing that such as eliminating noise when

people talk, and being capable to repeat a voice signal across longdistances without any problems. Thus, it is necessary to know

how voice is digitized by understanding how a technique calledsampling is works and how an analog-to-digital (A/D) converter

employs sampling to represent voice in a digital fashion.

2. ANALOG DEVICES FORTRANSMITTING VOICEBefore fully jumping into how an A/D converter is used to passvoice digitally, it is important to understand how voice is

converted into an electrical signal by using an analog device. Oneof the most common analog devices to accomplish this task is atransmitter which has a diaphragm (see Figure 1) composed ofloosely packed carbon particles. The sound pressure causes the

diaphragm to move back and forth which at the same time causes

the carbon particles to move back and forth as well whichproduces an electrical resistance to the charge. What is interestingabout this is that such electrical resistance happens at the same

amplitude and frequency as the sound pressure that caused thediaphragm to move back and forth does. In other words, the final

result is an electrical representation proportional to the frequencyand amplitude of the sound that caused the diaphragm to vibrate.

Figure 1. Representation of a diaphragm

2.1 Characteristics of Analog Signals That

Transmit VoiceAn analog signal that transmits voice has two constantly changingcharacteristics: the amplitude and the frequency of the signal. The

first one refers to the strength of the electrical signal which varieswith the loudness of the voice. Meanwhile, the frequency changeswith the pitch or tone of the voice. Those two variables will belimited to humans ability to generate voice in a given range

which is approximately 300 to 3300Hz in the frequency domainand up to 70dB in amplitude domain.

3. ANALOG-TO DIGITAL CONVERSIONIn order to convert voice into a digital format, it has to be

converted into an analog format first as explained above. Once theanalog representation of the voice is obtained, the next step is toconvert it to digital by using a device called an analog-to-digital(A/D) converter which employs a technique called sampling.

Sampling refers to the process of measuring the value of ananalog signal at regular rates called samples and assign a digitalvalue to that sample (See Figure 2). If the samples of an analogsignal that transmits voice are taken frequently enough (using an

A/D converter) and those samples are played back genuinely, theear will not be able to notice any difference from the originalsound.

3.1 Sampling

The bandwidth of the voice signal that needs to be passed digitallyis 3000Hz (3300Hz-300Hz) which is the range of frequencies thathumans are able to speak. However, to establish a

good sample rate to measure the analog voice signal at afrequently enough rate, the Nyquist theorem has to be use todetermine such rate. This theorem states that the minimumsampling rate must be at least twice the highest frequency.

Therefore, the sampling rate for an analog voice signal has to be6600 (2*3300Hz, the highest frequency) Hz or samples / second.As a matter of fact, the official sampling rate is 8000 samples/second which is what a T1 line uses and it is a little bit higher to

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otherwise, or republish, to post on servers or to redistribute to lists,requires prior specific permission and/or a fee. Please communicate

with my supervis Jon White professor at Harding University. A small

fee may be incurred due to the nature.Pablo Oropin, Passing Voice Digitally

Harding University

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what humans need in order to convert their voice into a digitalformat. This sampling rate is defined to address the higher rangeof frequencies that may occur in a conversation when people say

S and F sound; furthermore, it was standardized by ANSI(American National Standard Institute).

Each sample taken every 1/8000th of a second measures theamplitude level of the analog voice signal at a given time and it

contains 8 bits of information where each of those bits representsa 0 or a 1. 8 bits can represent a range of values from 0 to 255(See

figure 2). Thus, a total number of 256 levels are possible, which isenough to sample an analog voice signal and play it backfaithfully enough that the ear will not be able to notice anydifference.

Figure 2. Sampling 8000 times/second and assigning a value

from 0 to 255.

Obviously, the more samples are taken per second the higher the

quality of the sound is when it is played back, but the problem isthat the ear is not sensitive enough to notice those changes. That isthe reason why voice is digitized using only 8000 samples persecond.

3.2 Digital-to-Analog

Every bit obtained from sampling done by the A/D converter is inthe form of a square wave as opposed to the analog voice signalwhich is a continuously sinusoidal signal. The square wave signalfrom the bits travels in a pair of wires across the network where

other devices such as repeaters are found to strength the digitalsignal. Ultimately, the digital samples of the voice are played backat the destination point where an analog-to-digital converter is

employed to transform the digital signal into an audible form that

the human ear can recognize. In fact, the digital format of thehuman voice is converted back to the analog one (See Figure 3).

Figure 3. Voice is converted from analog to digital and back to

analog again

4. REMARKSTo represent voice in a digital fashion still requires converting itinto an analog electrical signal. Thus, the human voice is passed

digitally by sampling the analog representation of the sounds at agiven rate using an A/D converter. Nonetheless, all of this is done

because digital technology makes easier the manipulation and

combination of signals for carriers giving them more flexibility tomove digital signals from source to destination (From human tohuman).

5. ACKNOWLEDGMENTSI am really thankful for the Telecommunication class taught byDr. White, it helped me to understand better the concept of

passing voice digitally

6. REFERENCES

[1] Arcomano, R. 2002. VoIP Howto. DOI=http://tldp.org/HOWTO/VoIP-HOWTO.html#toc4 .

[2] Bates, Gregory, D. 2001. Voice & Data CommunicationsHandbook. The McGraw-Hill, California, 53-90.

[3] Sheldon, T. 2001. ADC (Analog to Digital Conversion).DOI=http://www.linktionary.com/a/adc.html.

[4] Torres, G. 2007. How a Analog-to-Digital Converter(ADC)works. DOI=http://www.hardwaresecrets.com/article/317.

[5] Voice Over IP. Wikipedia 2007. DOI=http://en.wikipedia.org/wiki/Voice_over_IP