New answers tagged digital-communications
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The difference in implementation between a FLL and PLL is completely dictated by what device is used as the error detector: if the error detector produces an error that is directly proportional to phase (phase detector) that is then passed into the loop filter, then the implementation will be a PLL and we would treat the VCO (Or NCO) as a phase integrator (...
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I'm giving a new answer because I suspect my previous answer was based on a misunderstanding of your question, but I'm not sure (so I'm not deleting the previous question right away).
I think you meant to ask
What are the limits to increasing the subcarrier count in an OFDM system? It seems like as-long-as-possible OFDM symbols are advantageous.
And the ...
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Sure, you can !
Practically, some systems are using OFDM system with number of subcarriers 4096, but the PAPR will be an issue in that case, and if your channel is fast-fading, the time-domain equalizer will be high-complex since one-tap frequency-domain equalizer will be useless.
Usually, choosing the length of the symbols depends on the channel delay, e.g....
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No, that can't work.
The guard interval has the purpose of "swallowing" the previous' symbol's impulse response.
Your longer DFT distributes energy all over the symbol duration. This doesn't solve the problem.
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In practice, most speakers seems quite predictable.
If you could do speech-to-text followed by a per-speaker prediction based on his/her precious speeches, I assume that the truely unpredictable/novel part of anything I say is going to be quite miniscule. Some machine learning ought to be able to generate something resembling my response to any topic based ...
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The symbol duration $T_p$ is related to the symbol rate $R_s$ by the simple relationship $$T_p = \frac{1}{R_p}$$
In the case of BPSK, $R_p$ is equal to the bit rate $R_b$, and $$T_p = \frac{1}{R_b}$$
For constellations with $M$ elements, each symbol encodes $k = \log_2(M)$ bits, the symbol rate is $R_p = R_b / k$, and the symbol duration is $$T_p = \frac{1}{...
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If you're defining the stepped-frequency waveform as a pulse train where each pulse has a different frequency, then it indeed does have advantages. One of the main ones being bandwidth requirements, which we'll go over.
Both FMCW and stepped-frequency techniques use frequency-domain range finding via the DFT. They map beat frequencies resulting from the ...
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That link provides a very superficial Q & A on signals and systems.
It's not clear which kind of a sampler or quantizer they are referring to.
However, if sampling and quantization are defined as ideal mathematical operators then,
you should get exactly the same result by altering their orders. Hence the sample values should be same (meaning that the ...
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I like the Rice's book as it gives enough details and explanations to be able to get to working implementation with a full understanding of what is actually happening. So the following is basically a citation from Rice.
1. What are the differences in PLL implementation?
A phase locked loop (PLL) in general has three basic components: a phase detector, a loop ...
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A receiver needs to synchronize on those two timings in order to demodulate with best performance:
Transmitter symbol rate
Transmitter carrier
Synchronization state of the art way is to use a PLL. You need two loops processing at the same time. The same PLL architecture can be used for both synchronizations but instantiated separately. That is the case for ...
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Let P(0) ( P(1) ) is a probability of transmitting bit zero (one); P(e|0) ( P(e|1) ) is a probability of error when detecting bit zero (one).
The probability of erroneous detection is
$$
P(e) = P(e|0)P(0) + P(e|1)P(1) \tag {1}
$$
Let $V_0$ ( $V_1$ ) be a nominal signal voltage of bit zero ( one ) signal at the transmitter.
$$
P(e|0) = \int_T^{\infty}{{\frac ...
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I commend you for using an intuitive algorithm. However, there are already established algorithms with far better performance. Phase recovery algorithms work by filtering the error signal down to zero. An example is the Costas loop structure shown in the figure below for QPSK.
Let's start with phase error detectors (Highlighted in yellow). The arctan ...
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The term "Hanning" is illiterate. The window is attributed to Julius von Hann. So it is a Hann window, or von Hann window.
If you use overlap add/save fast convolution on von Hann windowed data where the windows are 50% overlapped, you get the same result as rectangular windows of data that are non-overlapped, except for the initial and ending ...
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Is it necessary to take a derivative
Yes
or just get the first moment of the stochastic signal and eliminate
That's just a very crude approximation of taking the derivative. In some cases that's sufficient, but in most it's not.
For DC blocking the capacitor is typically used together with a resistor to form a high pass filter. The approximation is often &...
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It pretty much doesn’t. One could say a boxcar window is used, but you most likely won’t hear any mention it. The trick with doing FFT convolution is making sure your FFT length is at least equal to the number of signal samples plus the impulse response length minus one. Both your input signal and impulse response need to be padded with zeros appended to the ...
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what role if any does windowing (e.g. Blackman, Hanning etc.) play
None. If you do apply a window (other than rectangular) for a regular overlap add/save algorithm, your results will be wrong.
Time domain windowing can be helpful if your frequency domain processing is time variant, but that requires more complicated algorithm with partially overlapping and ...
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Constructive interference just increases the gain of the signal at the receiver, which is designed (using automatic gain control) to adjust the signal level over a very wide range to that desired for reception. So the issue the OP refers to is of no real consequence as long as the received signal is in the (relatively wide) range of the receiver between ...
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Most likely yes. FSK is already generated using many samples per symbol. Look at Matlab fskmod function.
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