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Re: [Discuss-gnuradio] Ideas for improved FM reception
From: |
Martin Dvh |
Subject: |
Re: [Discuss-gnuradio] Ideas for improved FM reception |
Date: |
Fri, 07 Dec 2007 18:28:52 +0100 |
User-agent: |
Icedove 1.5.0.14pre (X11/20071018) |
Daniel Povey wrote:
> Guys [+girls?],
> I'm new to the list. I joined because I do research in signal processing
> (speech recognition) and I have the idea
> that it ought to be possible to use statistical signal processing to get
> much better FM radio reception than a normal tuner.
> I basically want to process the quadrature output (the 2 signals that you
> get out after multiplying by sin + cos and
> lopass filtering) in a different way, not just doing atan but doing a bunch
> of predictive modeling at that stage, assuming
> the signal is a mix of multiple stations + noise, and trying to separate the
> signals out.
> I want to ask this list two things:
> (1) is anyone aware of anyone having tried something like this in the past
> (or know what search
> terms I might use to find this out), and
I am working on an optimized FM receiver for gnuradio.
At the moment my main concern is getting the processing load down.
But I am also going to implement improvements in quality.
The first thing I am going to implement is implementing MIMO.
That is, using multiple tuners to get signals from multiple antennas and
combine these signals to get a better signal quality.
(Phase array)
The second thing I want to do is detecting if multiple transmitters (at
different locations and frequencies) transmit the same radiostation.
With a single tuner I can capture all these signals at the same time.
Then do a correlation to get there phase and time relationship.
And then smartly combine them to get better signal quality.
The simplest thing to do is lineup the signals (phase and time shift) and add
them up.
A better approuch would be to do some statistical analisis like you described.
This can be done both before and after frequency demodulation.
I don't know yet which one will give better results.
Your approuch sounds interesting.
Unfortunately I am not a statisticus aor signal modeling expert, but I am very
handy with GnuRadio and FFT processing.
So maybe we could combine forces.
> (2) Could anyone give me to work with, the sampled output at the quadrature
> stage (I assume this would be something
> like 2 synchronized .wav or a stereo .wav file (does it support stereo?),
> sampled at quite a high frequency).
> Dan
>
Walker, Robert CIV NSWC Crane, WC NAE Staff wrote:
> Daniel - There are a bunch of different types of captured samples at
> http://www.kd7lmo.net/ground_gnuradio_ota.html. The output isn't .wav,
> but you can find a format description at the gnuradio website or the
> tutorial at http://sdr.nd.edu/docs/.
Daniel, did this work out for you?
If you need more captured signals, I can capture some for you.
I can even capture using two tuners / two antenna's or capture two frequencies
which transmit the same radio station at the same time.
What samplerate do you need/want.
The USRP can produce the following samplerates (Samples/second):
8000000, 5333333.33, 4000000, 3200000, 2666666.67, 2285714.29, 2000000,
1777777.78, 1600000, 1454545.45, 1333333.33, 1230769.23, 1142857.14,
1066666.67, 1000000, 941176.47, 888888.89, 842105.26, 800000, 761904.76,
727272.73, 695652.17, 666666.67, 640000, 615384.62, 592592.59,
571428.57, 551724.14, 533333.33, 516129.03, 500000, 484848.48, 470588.24,
457142.86, 444444.44, 432432.43, 421052.63, 410256.41, 400000,
390243.9, 380952.38, 372093.02, 363636.36, 355555.56, 347826.09, 340425.53,
333333.33, 326530.61, 320000, 313725.49, 307692.31, 301886.79,
296296.3, 290909.09, 285714.29, 280701.75, 275862.07, 271186.44, 266666.67,
262295.08, 258064.52, 253968.25, 250000,
for wideband FM usually 320000 samples/second is used
This is handy because after processing you can decimate by 10 to get a nice
32000 samples/second audiorate
Possible output formats:
raw complex shorts
or
raw complex floats
the gnuradio raw complex format is just all Real and Imaginary samples
interleaved.
Re0 Im0 Re1 Im1 Re3 Im3
for complex shorts this mean 16 bit signed short int Real0, 16 bit signed short
int Imag0, ....
for complex floats this mans 32 bit float Real0, 32 bit float Imag0, .....
It is also possible to output the Real and Imaginary samples in separate files
file one: Re0 Re1 Re2 ...
file two: Im0 Im1 Im2 ...
You can convert the complex captured signals into (stereo) wav files with
sox - Sound eXchange : universal sound sample translator
Sox is available in all linux distributions, and also available for windows.
http://sox.sourceforge.net/
I hope this helps,
Martin
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