All that I have studied till now
- There are two kinds of Analog signals: Continuous and Discrete
- What is Sample & Hold?
When in a discrete signalling, the value of signal is selected after each interval of time, then that interval of time is known as sampling interval. And such a circuit which enables us to do so is called the Sample and hold circuit.
- We are getting the value of the signal at every interval, which is integral multiple of sampling interval (T).
- A digital signal is discrete in space as well as time.-Quantization: Discreetness in Amplitude as well as Time.
- Hence, every continuous time signal is an analog signal but every analog signal is not continuous time signal.
- We need to remove abrupt change and to remove them we need to improve the higher frequencies (which have infinite harmonics). That is why we pass it through low pass filter.
- Advantages of DSP
- Less Sensitivity: In case of analog, noise remains in the signal but in digital, it is less sensitive. For eg. if V=5V and 10% error (or noise) then in
- analog: 4.5 and 5.5 is Vmax
- digital: 4.5 and 5.5 but always represented by 1 as it is approximated to 5V which is taken to be 1.
- Volume Production: Tuning is not needed in digital signalling (opposite of analog- like its needed in radios).
- Integration of Circuits: Analog has inductors (which are impossible to minimize and are costly) whereas in Digital, there is no use of inductors
- Accuracy increases as the word length increases: in digital 4.2,4,4.212.4.12121 are all 4
- Dynamic range increases with floating point: Any length of data can be used for eg 5V can be 5000 and -5 as -5000.
- Time sharing of processor: Digital processor is multitasking (simultaneous tasking)
- Multirate processing: Sampling rates may be different in digital
- Easy storage
- Very low frequency processing: At low frequency inductance creates problems in analog which is not the thing in digital.
- Disadvantages of DSP
- Complexity increases
- Power dissipation in CMOS
- frequency range is limited (because of the 'T' of the sample and hold circuit which can not be decreased to 0 to make the frequency infinite).
Let xa (t)= Acos(wt+phi)
let T be the sampling time interval, so T=1/Fs (which is sampling frequency)
so x(n)=Acos(wnT+phi)
and w=2*pi*F
so x(n)=Acos(2*pi*F*n*T+phi)
now as, T=1/Fs so, x(n)=Acos(2*pi*F*n*1/Fs+phi)= Acos(2*pi*f*n+phi)
Here f=F/Fs (normalized frequency)
Sampling Theorem: Fs>=2Fm
Fm is obtained from, w=2*pi*Fm
Sampling frequency should be more than twice the Maximum Frequency
Minimum Fs that can be used is 2Fm
let T be the sampling time interval, so T=1/Fs (which is sampling frequency)
so x(n)=Acos(wnT+phi)
and w=2*pi*F
so x(n)=Acos(2*pi*F*n*T+phi)
now as, T=1/Fs so, x(n)=Acos(2*pi*F*n*1/Fs+phi)= Acos(2*pi*f*n+phi)
Here f=F/Fs (normalized frequency)
Sampling Theorem: Fs>=2Fm
Fm is obtained from, w=2*pi*Fm
Sampling frequency should be more than twice the Maximum Frequency
Minimum Fs that can be used is 2Fm
- Why should minimum Fs be 2Fm or more than that?
- When Fs=2*Fmax
- and f=Fmax/Fs so the maximum f can be 1/2 so w (=2*pi*f)=pi
- For analog frequency is between -inf to +inf but in digital frequency is between -1/2 to 1/2.
- f=F/Fs
- Multiplying 2pi both the sides, 2*pi*f= 2*pi*F/Fs (f is discrete frequency and F is analog frequency)
- This equation can be written as, w= (W)/Fs (w=sampled omega, W= complete omega)
- w=radian/sample and W=radian/sec
- Now for discrete signal, f is between +-1/2 and w is between +-pi.
- Difference between Continuous Time SIgnal and Discrete Time Signal
- Continuous Time Signal:
- W=2*pi*F
- rad/s
- F=Analog Frequency (Hz)
- Range:
- -inf<=W<=+inf
- -inf<=F<=+inf
Continuous Time Signal:
- W=2*pi*f
- rad/sample
- F=Normalized Frequency (cycles/sample)
- Range:
- -pi<=w<=+pi
- -1/2<=f<=+1/2


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