Re: [Discuss-gnuradio] Improve FSK4 decodes in OP25

[Marcus Müller]

Hi Luke,

you don't happen to have a waterfall plot or a raw IQ recording?

Quadrature demodulation isn't the most resilient to noise. So, a low-pass filter that makes sure that events that are sufficiently shorter than a symbol after the Quadrature demodulator would sound wise. Then again, I don't know the OP25 infrastructure overly well, so chances are the FSK4 demod already contains one.

An alternative to your Quad FM demod would be a four-bandpasses-based decider. An easy way to build one is to either use four individual BPFs, or use one set of filter taps in the Polyphase Channelizer (which divides your f_sample_in  stream into four f_sample_in/4 -rate streams with channel centers). Just use a filter that would select the "zeroth" carrier. The magic of polyphase math will add three f_sample_in/4-shifted versions (you might need to use Rotator or similar to put the zeroth subcarrier at f=0). I have a "playing with channelizers" flow graph [1]. Advantage of the polyphase channelizer is that it uses very little more ressources than a single bandpass filter would – basically, you get the three additional filters for the cost of a short FFT. Yay!

Cheers,

Marcus

[1] https://gist.github.com/babffaa88f54a7fba9b52fc09bda0462

On 01/11/2017 04:44 PM, Luke Berndt wrote:

I am using the C++ OP25 libs to capture and record the P25 transmissions from a trunked radio system ( https://github.com/robotastic/trunk-recorder ).

Transmission can either use 4-Level FSK or CPSK. I am getting great decodes from a system using CPSK but I am having a lot of trouble with a system using 4FSK.

At a high level, here is my 4FSK flow graph (https://github.com/robotastic/trunk-recorder/blob/master/trunk-recorder/recorders/p25_recorder.cc):

Xlating + LPF to roughly capture the channel

ARB Resampler to get the system sample rate

Quad FM Demod

Multiply Const - to get the right levels, -3 to 3

FSK4 Demod - from OP25

FSK4 Slicer - from OP25

OP25 Decode

it then converts it to what is needed to record it as a Wav file

Are there some obvious blocks I should add in to help condition the input better? I think the incoming levels may not be constant. There also seems to be occasional interference. Is there a block that will discard values above a certain range? It would be great to throw-away samples above 3.5 or below -3.5.

Attached is a picture with what I think is an example of the interference.

Here is an example of a very clean recording: https://beta.openmhz.com/wmata?call-id=587652495833790100186198

Here is an example of a bad decode: https://beta.openmhz.com/wmata?call-id=58765250583379010018619c

Any tips would be greatly appreciated!!

- Luke

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