Hi all,

My background is digital design and i have ventured into the dark side of SDR. I'm struggling to get a grip of how IQ data relates to the RF input.

The SDR receiver has a 16 bit twos complement output and a selectable bandwidth of 5 or 10 MHz, the sample rate is 1.25 x BW i.e 12.5 Msps for a 10MHz BW. This is all the information i have on the SDR.The SDR creates a binary file which i read into LabVIEW and plot as a power spectrum density(PSD). The PSD envelope looks fine however the amplitude of the signals are all incorrect as well as the CF is incorrect ( it sits at 0Hz with signals +/- 6.5Hz.

my first question is:

If i have a CF of 100MHz and a BW of 10MHz how does the ADC sample this band? is there a filter which sweeps across the band and sampled every 1/12.5x10^6.

Also as the ADC is 16 Bits how is each bit/all 16 bits represented in IQ? is there a IQ part for each bit or each IQ represents the whole 16 bits.

The binary file i read into labVIEW has a 16bit I and 16bit Q word which i assume relates to a sample of the ADC. i read in 2^21 IQ points. which gives me a PSD resolution of 1-2HZ ( calculated by the PSD function)as i don't know the reference voltage of the ADC is there any way of converting the 2s complement IQ to dBM?

sorry for the mismatch of question im just struggling to get my head around every thing.

 

First read this
https://www.arrl.org/files/file/Technology/tis/info/pdf/020708qex013.pdf
and this for a SDR I/Q backgrounder.
http://whiteboard.ping.se/SDR/IQ

As you can see it's a direct conversion receiver with ADC I/Q samples from the Quadrature sampling mixer.

 

Perpendicular detection.

 

Thanks for the replies, The document makes a very good read and im a lot clearer now. The one thing i'm still unclear on is how the sampling is in keeping with Nyquist?

The input range of the sdr goes up to 3GHz. AS the sample rate is fixed to BW*1.25 i cant see how you could sample any higher then sample rate/2

For example if i had a CF of 1GHZ and a 10MHz BW the sample rate would be 12.5Msps. Would the LO down convert the incoming signal to a range that meats the Nyquist? which would mean i would need some how up convert the IQ samples im taking?

 

Thanks for the replies, The document makes a very good read and im a lot clearer now. The one thing i'm still unclear on is how the sampling is in keeping with Nyquist?

The input range of the sdr goes up to 3GHz. AS the sample rate is fixed to BW*1.25 i cant see how you could sample any higher then sample rate/2

For example if i had a CF of 1GHZ and a 10MHz BW the sample rate would be 12.5Msps. Would the LO down convert the incoming signal to a range that meats the Nyquist? which would mean i would need some how up convert the IQ samples im taking?

Yes, the Quadrature sampling mixer down converts the carrier frequency directly to base-band (spectrum centered at zero Hz) that would be limited in range using a LP filter and then digitized using the I/Q adc(s) for any deviation (voltage, frequency, phase-shift over time) of the incoming RF from the local oscillator. In the case of the SDR receiver we are not trying to reconstruct the carrier frequency, only the information contained in the modulation of the carrier so for our actual ADC sample we worry about the modulation bandwidth not the RF center frequency in the sampling system.

https://www.ieee.li/pdf/essay/quadrature_signals.pdf
http://www.ti.com/lit/an/slaa594a/slaa594a.pdf

What is Undersampling? If we use the sampling frequency less than twice the maximum frequency component in the signal, then it is called undersampling. Undersampling is also known as band pass sampling, harmonic sampling or super-Nyquist sampling. Nyquist-Shannon Sampling theorem, which is the modified version of the Nyquist sampling theorem, says that the sampling frequency needs to be twice the signal bandwidth and not twice the maximum frequency component, in order to be able to reconstruct the original signal perfectly from the sampled version. If B is the signal bandwidth, then Fs > 2B is required where Fs is sampling frequency. The signal bandwidth can be from DC to B or from f1 to f2 where B = f2 – f1. The aliasing effect due to the undersampling technique can be used for our advantage. When a signal is sampled at a rate less than twice its maximum frequency, the aliased signal appears at Fs – Fin, where Fs is the sampling frequency and Fin in the input signal frequency. In the above case, if we sample the 70-MHz signal with 100 MSPS sampling rate, the aliased component will appear at 30 MHz (100 – 70). As we know in advance that the signal is aliased, we can recover the actual frequency by using the Fs – Fin relationship. The undersampling technique allows the ADC to behave like a mixer or a down converter in the receive chain. For a band-limited signal of 70 MHz with a 20-MHz signal bandwidth, if the sampling rate (Fs) is 100 MSPS, the aliased component will appear between 20 MHz to 40 MHz (30 ±10 MHz).