I'm capturing an RF signal transmitted at 433.8MHz, with a bandwidth of about 50KHz. My goal is to be able to save it as a .wav file.
I'm using an RTL-SDR dongle and a software named HDSDR, where I set the input sample frequency to 3.2MHz and the output sample frequency to 192MHz. In this case, when I open the .wav file in Audacity, I see the waveform is correct.
Then, I'm trying to do the same using another software called GNU Radio where you develop your own system by adding each block involved in the processing. In order to get a good sync (so I try to match my HDSDR signal with the recorded here), I have to set the input sample frequency to 2MHz and the output sample frequency to 192KHz.
However, there are some other blocks in the demodulation process which parameters are not shown in HDSDR so I'm obtaining a signal with pulse lengths that don't match the correct one (the one get by HDSDR).
The first block I've got is Rational Resampler, where I set a decimation factor of 10 and an interpolation factor of 1.
Then, we've got the AM Demodulator, where I set a channel rate of 200K, audio decimation factor od 1, passband at 10K and stop frequency at 11K. I don't fully understand this block so it's possible the problem is here.
Then I've got an LPF with a sample rate of 200K, cutoff frequency of 70K, transition width of 1K, Hamming window and beta=6.76.
And then there are some gain factors and the wav file converter with a sample rate of 192K and 16 bits per sample.
It'd be great if someone could explain me if the values at the Ration Resampler, AM Demodulator and LPF blocks make sense.