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8

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.


5

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 ...


4

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 ...


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

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 ...


4

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,...


4

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

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'...


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

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 ...


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

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

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

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

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

You are missing several steps. You have (complex) time-domain samples. You need to convert that to scalar frequency domain data to get something like your example plot. Many steps are required to do that conversion before there are any frequency peaks to be found. You first need to: Pick a segment of complex IQ data from the RTL-SDR of length N (you ...


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

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 ...


1

The only way to entirely corrupt this file format is to cut 32 bits somewhere, as that would invert I and Q for the rest of the file. If you remove any multiple of 2*32 bits from the middle of the file you would only corrupt that single sample (and maybe the one before/after it), but the integrity of the file as a whole will not be compromised. Well, yeah, ...


1

Because the decimate function does not try to keep power constant. Remember that decimation is not inherently an LTI operation; so, probably, the authors of that function simply did not care too much for keeping the amplitude of a passband signal constant. Note that a decimation step typically includes filtering to $\frac1{\text{ds_factor}}$ of the ...


1

25.6 MHz is larger than the largest possible sampling rate of these two USRP models (25MHz). Therefore, this is impossible to implement directly; in any way, you'll need to generate your 25.6 MS/s signal first, and then resample to 25 MS/s. That resampler will be very CPU-consuming, probably more than your whole OFDM signal processing. The underruns happen ...


1

Of the methods you've already tried, the "looking for peaks in the spectrum" to me sounds like the most promising, but: My guess is that your "spectrum" is actually a PSD estimate done with the FFT. Which is a fine method of getting an understanding of the spectrum – but as you probably noticed, the "peaks" will not be very clear. Think about it: The ...


1

Is it really difficult to really know how much power you are receiving with your dongle. Each stage (antenna, amplifiers, mismatch losses etc) can not be easily calculated nor measured. What you could do though, is calibrate your dongle with a known source (i.e. a source that you already know transmits predetermined power . From a calibrated device, of ...


1

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 ...


1

that's pretty simple: in digital domain, real-world frequencies have no inherent "meaning". That is, a period of a real-world sine of frequency 10 Hz might be 2, 200, 1233 or whatever samples long, depending on the sampling rate. Thus, periods in DSP can only be measured in samples; for example, a sine of period 1 second has a period of 5.4 samples if the ...


1

There can be several reasons. Computer processing: One reason to use IQ data for SDR processing is to lower the computational processing rate (to use a slower or lower power processor) for visualization (panadapter) or demodulation without an additional conversion step. Many modulation schemes have asymmetric sidebands. IQ signals can carry disambiguated ...


1

Given sufficient input bandwidth to cover all your signals, this depends strictly on how much processor performance each filter, demodulator and/or modulator takes. If you can run 20 filters and demodulators simultaneously in 20 threads in real-time on your processor(s), and the modulation and power of each signal provides a sufficient S/N ratio, you should ...


1

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.


1

A couple of things jump out when looking at your diagram: The first low-pass filter has a decimation of 4, so the sampling rate at its output is 256 kHz. However, you have set the sampling rate of the FM receiver and the second LPF to 224 kHz. These should be 256 kHz. The cut-off frequency of the first LPF is 150 kHz. This means a total passband of 300 kHz. ...


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