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I have series of amplitudes of a signal recieved at center frequency of 433.9MHz, sampling at 250000 Hz. I can see the information received is the packet I am after, just sampled too high. I believe the transmitter transmits at 3918bps. What is the best method to downsample from 250000 Hz to 3918 bps?

I have tried using a butterworth filter, which doesn't seem to work and I don't understand why. I've tried picking every 63 samples (because 250000/3918 = ~63), but didn't seem to work either. Please bare in mind I am a bit of a newbie regarding DSP.

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  • $\begingroup$ You can't sample at 250,000 bps. bps means bits per second, and its a unit of transmission rate. Sampling rate is measured in either samples per second, or (sometimes) hertz, where hertz is understood to stand for sampling frequency. Please clarify your question. $\endgroup$ – MBaz May 15 '15 at 13:54
  • $\begingroup$ Ahh ok... so I seem to be confusing 250000Hz with bps. So to clarify I am sampling at 250000Hz, to a signal that is being transmitted at 3918 bps. $\endgroup$ – craig1231 May 15 '15 at 14:16
  • $\begingroup$ If you have a signal at 433.9MHz, sampling at 250KHz could be a little risky, unless you failed to mention that first you are down converting the signal. You may "see" that you have the information but it might be not straight forward to recover it. $\endgroup$ – Moti May 15 '15 at 18:17
  • $\begingroup$ @Moti, could you explain why it's a little risky or not so straight forward to recover it? I'm using an RTL SDR, and the lowest sample rate is 250kHZ. $\endgroup$ – craig1231 May 15 '15 at 22:22
  • $\begingroup$ A too low sampling rate may introduce noise that could prevent you from proper recovery of the signal because of the effect of "folding". This relates to the Nyquist sampling rate. $\endgroup$ – Moti May 15 '15 at 23:57
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Based on your information a packet length is about 77.2 millisecond. With 3918 bps, each packet contains about 302 bits. Frequency is shifting about every 0.255 millisecond. you get about 256 samples per bit, so 128 point FFT seems to be correct. I assume you have a real signal, so you need to evaluate only 64 bins. If you are synchronized properly, each two consecutive FFTs will provide you with one frequency, switching at the other. If you are not synchronized with the start of the first bit, only every other bin will provide you with a nice single frequency (assuming that you do not have noise and spurious signals)

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