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7

Your signal recording clearly shows that you have long streaks of 1.0 – that probably means you're clipping. Your signal is thus broken. Make a new recording with less gain.


7

The problem that I have is that I always have a big spike (10-15 dB) directly on the center frequency (no matter what frequency I set). I am relatively new to all this so I would appreciate any pointers on how to get rid of the spike. That spike is probably nothing surprising – just the LO leakage/DC offset, a very common artifact in direct conversion ...


6

The SDR (or any general digital signal processing system) takes the received RF signal, and downconverts it from the carrier frequency to the baseband. Now, the real bandpass signal from the antenna does not necessarily have a symmetric spectrum around the carrier frequency, but it can be arbitrary. If the downconverter now shifts the spectrum to the center ...


5

Here are some graphics I have to add to the other good answers, as well as typical correction schemes for DC Offset and IQ Imbalance. DC OFFSET Below is an example DC Offset on a 16 QAM signal (with some added AWGN). In this example the offset if shifted by an amount such that the power in the fixed vector representing the shift is 10 dB below the total ...


5

Real RF devices are not ideal as in a simulation or theoretic calculation. In particular, RF devices produce (among others) the following effects: DC Offset IQ Imbalance The correct IQ option is there to estimate these effects and remove them via digital or analog signal processing. The "spike in the middle of the display" is an artifact that stems from ...


4

I also read this from a response to a USRP user's question about RSSI measurements: [The] Received Signal Strength [Indicator is] always relative to some signal model, incorporating considered bandwidth, assumptions on the modulation scheme, duration of transmission, generally: It's a estimation of received signal strength based on some property of ...


4

Theoretically it's possible to do a frequency multiplex of two different protocols with one SDR transmitter. However, the practical main limitation is the speed of the digital-to-analog converter (DAC). E.g. for a WLAN and GSM band the channels that are closest on the frequency axis are at 1800 MHz and 2412 MHz, respectively. For generating such a signal you'...


4

The channel frequency controls the local oscillator on your SDR which is the frequency about which it covers. So you are receiving signals from 16KHz below 107.5MHz to 16KHz above. The concept you want to look up is called heterodyning. When you modulate (multiply) a signal by a sine wave you end up with two copies shifted in frequency space by the ...


4

First and foremost, I would recommend against over the air testing for this given the significant challenge in really being able to provide the same signal to each radio (since you have both temporal and spatial constraints that you cannot simultaneously meet). I would instead use one GNU radio as a transmitter (or any other repeatable high quality source) ...


3

I do a lot of decimation in the frequency domain. Little details are important. I assume you already know the basic rules for fast convolution: the FFT length N is equal to the data blocksize L plus the length of the filter impulse response M minus 1. Each operation uses L samples of new data plus M-1 samples of data from the old block. Ensure that the ...


3

I'd say there are two ways to look at this problem. If you're receiving a conventional $M$-QAM signal with AWGN noise, then the I and Q streams are actually two independent, real, $\sqrt{M}$-PAM signals. You can filter and process them independently. The same is true over the wireless channel, as long as the fading is flat and you do the appropriate ...


3

It is common for low-cost direct-conversion (zero-IF) RF receivers to have a DC offset in their A/D hardware, which would correspond to a large spike at the frequency that you're tuned to. That's probably what you're seeing in the waterfall display, as some models of RTL2832U receivers use zero-IF architectures. For what it's worth, for such a cheap device,...


3

First I found the sine wave extrema (black markers at the bottom) and stretched and superimposed a sine wave graph (red) on top of the signal graph with the extrema in the same locations. The signal matches (0) the sine wave over some of the non-zero segments, and over the other non-zero segments it matches the sine wave with a sign flip (1). This gives a ...


3

The IQ data looks like a sinusoidal modulated pulse train with binary phase shift keying (BPSK) superimposed. The BPSK symbol rate appears to be at the same rate as the pulse repetition frequency (PRF). I was bored, so I went ahead and modeled the data in MATLAB. clear; clc; % Measured quantities from IQ data pw = 2.1771e-4; % Pulse Width pri = ...


3

For both receive and transmit SDR experimentation/testing at RF/HF/VHF/etc. frequencies, I use a LimeSDR Mini (far less expensive than an Ettus). But I stream from/to it using C programs and files, not direct from Matlab. (Is it possible to write a C stub to pass data back and forth with Matlab?) There is also the ADALM-Pluto kit from Analog Devices, ...


3

[Update: I mentioned a possible +3dB processing gain by including the Hilbert Transform prior to DDC for the case of a real IF signal in the first version of this post, which @MattL questioned so I dug into this further and confirmed that there is no such processing gain, so the only advantage to doing that would be to simplify filtering since it would ...


2

how does the SDR sample rate translate to sample rate and bit depth of the resulting audio signal? Extremely indirectly. I haven't got precise mathematical formulas for you, but here's an overview of the topic which should let you figure out where you want to dig deeper. First of all, FM is an analog modulation. This means that there is no inherent sample ...


2

I believe that what you are looking for is Bandpass Sampling. What Nyquist theorem says is that your sampling frequency must twice the bandwidth of your signal - not carrier frequency of it. Hence in FM modulated signals you don't need to take it into consideration.


2

I can envision advantages (or convenience) in doing this for lower IF signals certainly, as long as the sampling rate is significantly higher than the signal bandwidth to contain the phase dispersion you would get with this approach. An implementation can be lower cost and simpler in the analog with a single mixer/ADC front-end for example. This becomes ...


2

As far as I understand your question, you want to estimate the PSD in a large bandwidth. What you're currently doing is tuning to one part of that bandwidth, receive samples at a low sampling rate, calculate the power in that small bandwidth, retune, and repeat. So, yes, using a larger sampling rate will obviously reduce the time you need to sense a large ...


2

Frequency is the derivative (or 1st difference) of phase. To get an IQ signal from which you can get phase angles, first multiply your wave file samples by a cosine at 3.15kHz and a sinewave at a frequency of 3.15kHz. That will hetrodyne your signal down to baseband IQ. Use atan2 of the IQ array to get an array of angles, take the 1st difference between ...


2

One may "Correct IQ" to correct gain and phase imbalances in the I signal and the Q signal. These imbalances can lead to artifacts in the output (resulting in signal distortion). The IQ signaling is based on the idea that a real signal has been down converted with two sine waves of equal amplitude that are at 90 degree angles with respect to each other. An ...


2

The article mentions that the sequence is likely pulse duration modulation, but the RfCat library doesn't support PDM. Instead, it supports OOK. So what the author is doing is representing PDM as OOK. In particular, the author has likely measured the length of the pulses and determined that the short pulse is 1/3 as long as the long pulse, and that the ...


2

Apparently, your gain is too much in this case and the ADC is getting overloaded. Try to reduce the gain and capture again. Plus, I think your sampling rate is not correct here.


2

It's very hard to associate the numbers you're calculating with the actual signal power at the antenna. The SDR will filter, amplify, and AGC the signal, and then it'll convert it to digital. At best, you can say that the power you're calculating is somewhat related to the true signal's power. The power you're calculating is in linear, not logarithmic, ...


2

As @Marcus_Müller points out, dropping samples will corrupt whatever digital symbol or analog waveform you were operating on at that instant. You'll also introduce discontinuities in the time domain, which means you'll get broadband noise in the frequency domain at that time. Probably the worst effect is, that if you drop a half sample, you end up swapping ...


2

There's a lot of different domains of knowledge coming together here, so I'll split my answer into multiple sections, each answering an implicit question that you raise in your explicit question. Hope that helps! Can your RTL-Dongle actually receive at 1.72 GHz? So, first the bitter pill: There's a lot of sellers out there that offer RTL dongles and claim ...


2

The OP stated he was interested in $\pi/4$-DQPSK (not QPSK), so phase synchronization is presumably not an issue for him. As far as the actual implementation is concerned, you'll save yourself some time if you become familiar with the bottom of pages 29 (symbol mapping) and 37 (differential detection) in this student paper. Ignore all the old TI DSP-chip ...


2

The high-frequency (RF) section of an SDR is all analog. Typically, the analog receiver downconverts the RF signal to an intermediate frequency that is within the Nyquist range of the ADC. As Stanley points out, you can also do bandpass sampling, though that is less common, in my experience.


2

The first concept that I think will be helpful to see is pulse shapes that repeat in time at consistent rates. If you repeat a fundamental pulse shape in time, the result in frequency will have the envelope of the Fourier Transform for that pulse but will only contain frequency components at multiples of the repetition rate (harmonics). Thus the Fourier ...


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