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the general relaship between snr and E0/N0 is as folow:
E0/N0= SNRxW/R, where W the frequency bandwidth of the modulated signal
and R is bit rate,
u can read some details in the book of Sklar "fundamental of digital communication"
best regads
if I dont know if I well understand your question: we sample the MTI signal as any other signal. we must respect the shannon criteria.
in the other hand usually we apply MTI on each channel of I&Q detector, so after applying MTI filters we take the sqrt (I2+Q2) and this signal we display at the...
the first step is to acquir the sound. u can do it by connecting the micro to your sound card and read samples of the sound continously.
the second step which is mor complecated is to identify the desired sound. for this step u must use some of digital signal processing algorithms, for example u...
do attention that u must put the data in the vector that is symetric according to the middle point.
yhe essay way for u check first the output of FFT fuction and let the data as well
the ideal low pass filter has impulse response sinc function
so first determine the fc
calculate impulse response 2fc*sinc(2fct)
then truncate the impulse response by 2N, by multipling the result h(t) by appropraite window such as hamming or hannig window
the result is your impulse response
u...
u must have dsp with D/A or codec
the write a program that read stored samples of the signal and the output it on D/A
if the stored sound is compressed by MPEG 3
by example then u must write program that decode the compressed saples
or use specific IC for doing that
if u have modulator produce y1(t) for input x1(t)
and produce y2(t) for input x2(t)
we say the system is linear if we make the input = x(t) = ax1(t)+bx2(t)
and the output becom y(t)= ay1(t)+by2(t)
I dont thing thant any of PSK or FSK is a linear modulator
Re: ica
after u analyise the signal u can select only desired signal
check the folowing sites
www.cis.hut.fi/projects/ica/fastica/
www.cs.berkeley.edu/~fbach/kernel-ica/
I did not well understand your quetion, but in digital filter design we normalise the frequencis to 1.
in matlab we make the fs/2 =1
so we normalise the all frequencis to fs/2
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