if I have an analog modulated signal with a DSBSC modulation (modulation index 100%) as an input in my receiver and I want to convert it to a sampled signal, do I need to demodulate the signal (with coherent detector) and then sample it ? Or the other way around?

I know that from the front end of the receiver you have an LNA, BPF, LO for getting the signal to the baseband, LPF and then is it sampling or demodulation?

  • $\begingroup$ What do you want to convert to a digital (more likely you meant a discrete-time sampled) signal? The modulated (high-frequency) signal as received? The modulating (low-frequency) signal? And how is the analog signal modulated onto the high-frequency carrier? AM DSB at less than 100% modulation? FM? PM? $\endgroup$ May 4, 2021 at 14:25
  • $\begingroup$ You can sample first and then demodulate digitally but you need to have an AD converter and processing system that's fast enough. $\endgroup$
    – Hilmar
    May 4, 2021 at 14:51
  • $\begingroup$ Do a web search on "software defined radio". $\endgroup$
    – TimWescott
    May 4, 2021 at 17:33

1 Answer 1


Demodulation of a DSBSC signal is done by simply multiplying the signal with a coherent (phase-locked) carrier frequency. Either approach (demodulate in analog and then sample, or sample and then demodulate digitally) will work, but there is compelling advantages to sample and do all demodulation digitally. This would be preferred given the flexibility and adaptability a digital solution can provide.

With the digital approach, the sampling clock is unrelated to the waveform as an independent clock, and it only need to be twice the bandwidth of the modulated waveform (plus additional margin for economical filtering). An important criteria is the analog input bandwidth of the ADC, ensuring it is wide enough for the location in frequency of the modulated waveform when it is presented to the input of the ADC. Consider not down-converting in the analog to baseband but instead down-convert to a lower intermediate frequency or IF. (Down-converting in the analog to baseband would require either a coherent phase locked carrier locked to the carrier in the received signal which is suppressed, or two ADC's to sample the In-phase (I) and Quadrature (Q) components from an analog IQ down-converter, as needed to resolve carrier offsets and complete the demodulation digitally).

As long as we pay attention to proper anti-alias filtering ahead of the ADC, sufficient analog input bandwidth and ADC dynamic range (effective number of bits) and evaluate the frequency plan such that the resulting signal can be further digitally filtered; we could directly sample the modulated waveform with a sampling clock much lower than the carrier frequency (under-sampling), or down-convert the analog signal to a lower carrier frequency (IF) that is within the first Nyquist zone (DC to $f_s/2$) of the ADC (close to $f_s/4$ is a good choice to simplify analog filtering), or something in between. Once sample we can then digitally resolve residual carrier offsets, as well as gain leveling and optimum demodulation with matched filtering etc.


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