I have been using GNURadio for different purposes for some time. However, in most of those times, I used either a very simple flow graph created by myself or a complicated flow graph created by somebody else. Therefore, in the case of complicated flow graphs, I didn't even bother to explain how they work as I didn't really understand their internal functionality. Yesterday, I came across a need to demodulate a signal captured from an RTL-SDR and saved to a file. From the output I observed through the GQRX tool, the signal is modulated using Amplitude Modulation (AM). I wanted to implement a demodulator by myself and therefore I searched the web for a tutorial regarding this kind of stuff. I found a tutorial given in the references section which was really useful [1].
In the rest of this post, I will be explaining how this AM demodulation flow graph is implemented and the functionality of each block according to my understanding. There can be missing details and inaccurate things. If you notice some glitch, please be kind to let me know by leaving a comment. Here we go.
(1) The signal I'm trying to demodulate is in 32MHz frequency and therefore, I tuned an RTL-SDR dongle to that frequency and captured data to a file with a sample rate of 1.8MHz. The tool I used for this purpose is a command line tool called osmocom_fft which comes in handy when we need to save captured data into the .cfile format.
(2) Let's start with a File Source block in the flow graph. Double click on it and then browse to select the source file which contains data. Set the Repeat option to yes.
(3) The data in the file is recorded in 1.8MHz sample rate and the samples are centered to the 32MHz signal. If we draw an FFT spectrogram from the raw data of the file, we will see the X-axis from (32-1.8)MHz to (32+1.8)MHz. However, the real AM signal does not span such a width. The real signal was within about 5kHz width (not exactly). Therefore, let's narrow down our data scope to somewhere about 10kHz in order to focus on our signal. To narrow down our spectrum information from 1.8MHz width to 10kHz, we need to resample the data. So, let's use a Rational Resampler block as the next step.
(1) The signal I'm trying to demodulate is in 32MHz frequency and therefore, I tuned an RTL-SDR dongle to that frequency and captured data to a file with a sample rate of 1.8MHz. The tool I used for this purpose is a command line tool called osmocom_fft which comes in handy when we need to save captured data into the .cfile format.
(2) Let's start with a File Source block in the flow graph. Double click on it and then browse to select the source file which contains data. Set the Repeat option to yes.
(3) The data in the file is recorded in 1.8MHz sample rate and the samples are centered to the 32MHz signal. If we draw an FFT spectrogram from the raw data of the file, we will see the X-axis from (32-1.8)MHz to (32+1.8)MHz. However, the real AM signal does not span such a width. The real signal was within about 5kHz width (not exactly). Therefore, let's narrow down our data scope to somewhere about 10kHz in order to focus on our signal. To narrow down our spectrum information from 1.8MHz width to 10kHz, we need to resample the data. So, let's use a Rational Resampler block as the next step.
In the Rational Resampler block, we have two important parameters; Interpolation and Decimation. Interpolation value represents the number by which the input signal's sample rate is multiplied. Decimation value represents the number of which the input signal's sample rate is divided. In order to resample a 1.8MHz signal into the 1kHz signal, we have to divide the input by 180 and multiply by 1. That means, set the Interpolation to 1 and Decimation to 180 in the Rational Resampler block.
Here, instead of directly putting the resampling value 18 in the field, let's make the value a variable so that later we can change it easily. Therefore, add a variable called resamp_factor and set the value to 180.
(4) It is important to note that the purpose of the Rational Resampler is to change the sampling rate of some data. The output of the current setup is a signal with a sample rate of 10kHz. However, to actually isolate the signal we need to extract, we should filter out the other frequencies and keep the target frequency. This is where we need a filter. The target signal is about 5kHz wide. And it is centered at the 35MHz. Therefore we need to add a Low Pass Filter block with a cutoff frequency at 5kHz. Any signals which contain more than 5kHz components will be removed by this block.
When setting the sample rate of the Low Pass Filter block, it is important to set it to the value (samp_rate / resamp_factor) since, the sample rate is now changed by the previous block, the Rational Resampler.
(5) Now we are focused to the exact signal we need to demodulate. Since we are interested in amplitude modulation (AM), we are interested in the amplitude of the signal. We can convert the complex data we were receiving out of the filter to a series of numbers which represent the amplitudes using a Complex to Mag module. Now the output of this module is basically the demodulated base band signal.
(6) It is useful to have a way to increase and decrease the amplitude of the demodulated signal in order to listen to it conveniently. In order to achieve that, we add a Multiply Const block. In the constant field, instead of adding an exact number, we put a name called volume. Then, we should add WX GUI Slider block separately. Set the ID of this new block to be volume. Set the minimum value 0 and maximum value 10 with a number of steps set to 100. Now, when we change the volume in the WX GUI slider, it will change the value in the Multiply Cont block.
(7) Before we add an audio sink to listen to the sounds, we need one more thing. Since an audio sink would need the sample rate to be around 48kHz, let's resample the sample rate of the current setup from 10kHz to 48kHz by using another Rational Resampler. Set the decimation to 10 and interpolation to 48 for the obvious reasons.
(8) Now the last thing. Add an Audio Sink block and set the sample rate to 48kHz. If you think it's useful, add a WX GUI Scope Sink which will show the waveform of the audio signal being heard. The sample rate of that block too has to be 48kHz.
Run this flow graph and listen to the output sound. You should be able to hear the demodulated signal. In case you need to take a look at the original GNURadio Companion based file I created for the above AM demodulator, download the GRC file in the GIST given in this link. Cheers!
References:
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