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A* sin(2*pi*f*t + phase)
OR
A* sin(w*t + phase)
** if we increase the Frequency too much the signal will be corrupted for example x= 5*sin (2*pi*100*t + 0) .
because of how fast is the signal is different than how fast is the sampling.
Thus a simple solution is to have t=0:(2*pi)/1000000:pi;
** Corruption might be because of wrong sampling rate.
** assume t2=t1 meaning it do a complete 1 cycle in the time where t1 is in its half of the cycle meaning when
t1 have a complete 1 cycle t2 have 2 cycles. Therefore t2 happens more frequently than t1 and have shorter
wavelength.
DSP
** Analog: Continuous function V of continuous variable t. V(t)
Ex: time,space.
** Digital: Discrete function Vk of Discrete Sampling variable tk. where k is an integer Vk=V(tk)
** Misnomers
- Analog: Continuous time Continuous Amplitude
- Discrete: Discrete time Continuous Amplitude
- Quantized: Continuous time Discrete Amplitude
- Digital: Discrete time Discrete Amplitude
** Why Digital??
simpler no need to do integration only addition & subtraction is enough.
because our computer is a digital system.
** Digital cheaper than Analog.
** Applications of DSP?
Doctors Answer - Phones. - Mother and Child heart beats Monitoring System.
slides Answer is: - Predicting a systems output. - Implementing a certain processing task. -Studying a certain signal.
** DSP Hardware?
- (GPP) General purpose processors, micro-controllers. - (DSP) Digital Signal Processors. - (PLD, FPGA)
Programmable logic . are faster, real time DSPing.
** DSP Software?
- Programmable languages: Pascal, C/C++. - High level Languages: MATLAB, Mathcad,Mathmatica. -Dedicated
tools(ex:filter design s/w packages).
** DSP SYSTEM :
1) Analog Signal > 2) Antialiasing Filter from Noise (LPF) > 3) A/D > 4) Processing > 5) D/A > 6)
(Reconstruction)Filter (LPF)
LPF is used for smoothing.
DSP SYSTEM Decisions: see the slides
DSP Applications and Algorithms are in the slides.
Sampling
** Sampling: is how fast must we sample * a continuous signal to preserve its info content.
** Sampling Theorem= A signal s(t) with Maximum Freq Fmax can be recovered if sampled at frequency
fs>2fmax
** Fourier Series: Any Periodic signal can be generated by a combination of sine and cosine signals
** Square Wave can be generated by a combination of sine and cosine signals
ADC
** The More the Bits the More is the Resolution, the More Precision.
** The minimum change in LSB.
** Types of ADC? 1) Flash Type. 2) Successive Approximation 3) Counter Based Single Ramp, Dual
Ramp, Sigma Ramp.
** Successive Approximation: 8 bits means also at 8 clock cycles needed.
** Convolution: check the slides. 1) Inversion 2) Shifting 3)Multiplying 4) Summiation.
** From Time domain to Frequency domain >(Analysis) ** From Frequency domain to Time domain >
(Synthesis)
** In the Frequency Domain, angle-time is not important what is important is the the Frequencies & its
Magnitude.
** Fourier said that a signal can be decomposed into a sum of sinusoids.
** Fourier Synthesis of Square Wave : look in the Phone
T= 2*pi;
d=1000; // displacement # of points from 0 to 2pi
N=101;
// as we increase N the graph becomes more squared
k=0:T/d:T;
f=[zeros(size(k))];
for n=1:2:N
fn=4/(pi*N) * sin(2*pi*n*k/T);
f=f+fn;
plot(k,f)
pause(0.1);
end