DOKUZ EYLUL UNIVERSITY Faculty of Engineering Electrical & Electronics Engineering Department
EE 413 DIGITAL COMMUNICATION SYSTEMS
PROJECT # 1 FEASIBILITY REPORT
2006502011 Onur NE 2006502012 Samet DEVEL
18 November 2011
THEORY
HOW IS SOUND RECORDED DIGITALLY? Recording onto a tape is an example of analog recording. Audacity deals with digital recordings - recordings that have been sampled so that they can be used by a digital computer, like the one you're using now. Digital recording has a lot of benefits over analog recording. Digital files can be copied as many times as you want, with no loss in quality, and they can be burned to an audio CD or shared via the Internet. Digital audio files can also be edited much more easily than analog tapes. The main device used in digital recording is an Analog-to-Digital Converter (ADC). The ADC captures a snapshot of the electric voltage on an audio line and represents it as a digital number that can be sent to a computer. By capturing the voltage thousands of times per second, you can get a very good approximation to the original audio signal:
Figure 1: Audio signal with audio samples Each dot in the figure above represents one audio sample. There are two factors that determine the quality of a digital recording:
Sample rate: The rate at which the samples are captured or played back, measured in Hertz (Hz), or samples per second. An audio CD has a sample rate of 44,100 Hz, often written as 44 KHz for short. This is also the default sample rate that Audacity uses, because audio CDs are so prevalent.
Sample format or sample size: Essentially this is the number of digits in the digital representation of each sample. Think of the sample rate as the horizontal precision of the digital waveform, and the sample format as the vertical precision. An audio CD has a precision of 16 bits, which corresponds to about 5 decimal digits. Higher sampling rates allow a digital recording to accurately record higher frequencies
of sound. The sampling rate should be at least twice the highest frequency you want to represent. Humans can't hear frequencies above about 20,000 Hz, so 44,100 Hz was chosen as
the rate for audio CDs to just include all human frequencies. Sample rates of 96 and 192 KHz are starting to become more common, particularly in DVD-Audio, but many people honestly can't hear the difference. Playback of digital audio uses a Digital-to-Analog Converter (DAC). This takes the sample and sets a certain voltage on the analog outputs to recreate the signal, which the Analog-to-Digital Converter originally took to create the sample. The DAC does this as faithfully as possible and the first CD players did only that, which didn't sound good at all. Nowadays DACs use Oversampling to smooth out the audio signal. The quality of the filters in the DAC also contributes to the quality of the recreated analog audio signal. The filter is part of a multitude of stages that make up a DAC. PULSE - CODE MODULATION (PCM) PCM is a digital transmission system with an analog-to-digital converter (ADC) at the input and a digital-to-analog converter (DAC) at the output. When the digital error probability is sufficiently small, PCM performance as an analog communication system depends primarily on the quantization noise introduced by the ADC. PCM Generation and Reconstruction
Figure 2: PCM Generation System Figure 5 is the functional blocks of a PCM generation system. The analog input waveform x(t) is lowpass filtered and sampled to obtain x(kTs). A quantizer rounds off the sample values to the nearest discrete value in a set of q quantum levels. The resulting quantized samples xq(kTs) are discrete in time (by virtue of sampling) and discrete in amplitude (by virtue of quantizing). To display the relationship between x(kTs) and xq(kTs), let the analog message be a voltage waveform normalized such that . Uniform quantization subdivides the 2-
V peak to peak range into q equal steps of height 2/q V, as shown in figure 6.
Figure 3: Quantization characteristic The quantum levels are then taken to be at 1/q, 3/q, , (q-1)/q in the usual case corresponds to any
when q is an even integer. A quantized values such as sample value in the range .
Next an encoder translates the quantized samples into digital code words. The encoder works with M-ary digits and produces for each sample a codewords with unique encoding of the q different quantum levels requires that and should be chosen to satisfy the equality, so that digits per word,
. The parameters
Thus, the number of quantum levels for binary PCM equals some power of 2, namely . Finally, successive codewords are read out serially to constitute the PCM waveform, an M-ary digital signal. The PCM generator thereby acts as an ADC, performing analog - to digital conversions at the sampling rate and parallel - to serial read out. Each encoded sampl is represented by a becomes transmission is with -digit output word, so the signaling rate . A timing circuit coordinates the sampling
. Therefore, the bandwidth needed for PCM baseband
Fine grain quantization for accurate reconstruction of the message waveform requires , which increases the transmission bandwidth by the factor times
the message bandwidth W. Now consider a PCM receiver with the reconstruction system in figure 7.
Figure 4: PCM receiver The received signal may be contaminated by noise, but regeneration yields a clean and nearly errorless waveform if is sufficiently large. The DAC operations of serial to
parallel conversion, M-ary decoding, and sample and hold generate the analog waveform drawn in figure 8. This waveform is a staircase approximation of , similar to flat
top sampling except that the sample values have been quantized. Lowpass filtering then produces the smoothed output signal , which differs from the message . to the extent
that the quantized samples differ from the exact sample values
Figure 5: Reconstructed waveform Perfect message reconstructed is therefore impossible in PCM, even when random noise has no effect. The ADC operation at the transmitter introduces permanent errors that appear at the receiver as quantization noise in the reconstructed signal.
PROPOSAL
Learning objectives In order to accomplish this project we have to know digital communication techniques. The learning subjects can be listed as below:
General Digital Communication Systems
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Source of the Digital Communication Systems Advantages & Disadvantages of Digital over Analog Systems The Main Blocks that forms the Digital Communication Systems General functions that are going to be used Learning performance improvement algorithms Propose and Importance of the Coding Techniques for Source Coding & Decoding Detailed examination Pulse Code Modulation (PCM) Signal to Quantization Noise Ratio (SQNR) Types of the Channel Models Important Characteristics of the Channels Detailed Analysis of Additive White Gaussian Noise (AWGN) The importance of these blocks Types of the Modulator & Demodulator Learning PSK technique Learning Gray Mapping Techniques The Main Propose of this Coding Techniques for Channel Coding & Decoding Learning Convolution Code (Non-systematic) Techniques Viterbi Decoding Algorithm Important Characteristics and Issues that Affect the System Performance To evaluate the Bit Error Rate (BER)
Learning MATLAB
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Source Encoding and Decoding
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Channels
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Modulator & Demodulator
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Channel Encoding & Decoding
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Signal to Noise Ratios Preparation of some important performance test and their outputs
The Goal of This Project: 1. To understand the fundamentals of digital communications techniques. 2. To understand the underlying concepts of channel encoding decoding and baseband digital modulation-demodulation. 3. To generate a white Gaussian noise sequence and to observe the effect of channel noise with varying power on overall system performance. 4. To evaluate the Bit Error Rate (BER) versus signal to noise ratio (SNR) of a digital communication system.
MATLAB CODES
Analog to Digital Conversion (ADC)
%EE413_Digital Communication Systems %Project 1 %Group Members: ONUR NE - SAMET DEVEL %ADC- Analog to Digital Conversion clc clear all close all Fs = 8000;%sampling frequency n=3;%digits is eqal to n-1 x=sin(2*pi*(0:(Fs-1))/Fs); %sampling equation of sinusoidal input signal %eps returns the distance from 1.0 to the next largest double-precision %number %find(x)= Find indices and values of nonzero elements x(find(x>=1))=(1-eps); %B = floor(A) rounds the elements of A to the nearest integers less than or %equal to A.For complex A, the imaginary and real parts are rounded %independently.
xq=floor((x+1)*2^(n-1)); xq=xq/(2^(n-1)); xq=xq-(2^(n)-1)/2^(n); stem(xq) grid on
%quantized samples signal
%quantized signal plot
hold on plot(x,'r') %sinusoidal sampling signal plot xlabel('fs') ylabel('voltage') title('quantized signal and sinusoidal sampling signal(red one)') figure plot(x,xq) %quantization characteristic screen grid on xlabel('x(kTs)') ylabel('xq(kTs)') title('quantizatation characteristic')
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Digital to Analog Conversion (DAC)
%EE413_Digital Communication Systems %Project 1 %Group Members: ONUR NE - SAMET DEVEL %DAC- Digital to Analog Conversion clc clear all close all Fs = 8000;%sampling frequency n=6;%digits is equal to n-1 x=sin(2*pi*(0:(Fs-1))/Fs); %sinusoidal input signal %eps returns the distance from 1.0 to the next largest double-precision %number %find(x)= Find indices and values of nonzero elements x(find(x>=1))=(1-eps); %B = floor(A) rounds the elements of A to the nearest integers less than or %equal to A.For complex A, the imaginary and real parts are rounded %independently. xq=floor((x+1)*2^(n-1)); xq=xq/(2^(n-1)); xq=xq-(2^(n)-1)/2^(n); stem(xq) %quantized samples signal
%quantized signal plot
grid on hold on plot(x,'r') %sinusoidal sampling signal plot xlabel('fs') ylabel('voltage') title('quantized signal and sinusoidal sampling signal(red one)') figure plot(x,xq) %quantization characteristic screen grid on xlabel('x(kTs)') ylabel('xq(kTs)') title('quantizatation characteristic') k=0:7999; %sample m = 0:length(xq)-1; for i=1:length(k) y(i) = sum(xq.*sinc(m- k(i))); end figure plot(k,y); %DAC signal hold on plot(x,'r') %sinusoidal sampling signal plot title('DAC signal and sinusoidal signal plot(red one)') grid on
q u a n t iz e d s ig n a l a n d s in u s o id a l s a m p lin g s ig n a l(r e d o n e ) 1 0 .8 0 .6 0 .4 0 .2 v oltage 0 -0 .2 -0 .4 -0 .6 -0 .8 -1
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quantizatation characteristic 1 0.8 0.6 0.4 0.2 xq(kTs) 0 -0.2 -0.4 -0.6 -0.8 -1 -1
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GROUP WORK PLAN
After taking project and researching some information about project topic, we determine tasks about project studying time. For this aim, we prepare gantt chart. This table includes planning programs for time duration. Both group members are responsible for each tasks.
Nov/1 5 Nov/1 8 Nov/2 2 Nov/2 9
Task \ Date
Preparing a feasibility report A/D and D/A conversion using MATLAB Preparing Interim Report 1 Modulator-demodulator using MATLAB Preparing Interim Report 2 Channel encoder/decoder using MATLAB Designing a GUI Preparing poster presentation Writing final report
Dec/6
REFERENCES
Communication Systems, A. Bruce Carlson, Paul B. Crilly, Janet C.Rutledge, Mc Graw Hill Modern Digital and Analog Communication Systems, B. P. Lathi, Oxford University Press 1998 Communication Systems, Simon Haykin, John Wiley & Sons, Inc.
