Hi. I'm referring to a signal that is the result of the sampling an audio analogue signal at a sampling rate that meets or bests the rate calculated by the Nyquist equation. I'm not referring to the modulation of a carrier.
My difficulty is not in grasping what I think is the theory, it's believing it. So, I'm looking, in a way, for confirmation of what I think I'm understanding.
With respect, you don't quite understand the theory.
You are all around it, but not there yet.
If we sample an analogue signal, by say using an electronic switch, which simply opens and closes, we are modulating the audio analogue signal.
Yes....., but the usual convention is that the lower frequency signal is modulating the higher frequency signal.
This is shown by the fact that the spectrum of the output, of the sampling process, contains modulation products, that is new sidebands. We then filter out all the new sidebands, by a low pass filter, we are left with a spectrum that matches the original analogue signal.
Let's just for the moment, neglect the pulse side of things & look at a simple AM modulator, with a single modulating frequency.
We will call the Carrier frequency
fc, & the Modulating frequency
fm.
Appearing at the output of the modulator, are the two original frequencies,
fc, &
fm,as well as the sidebands,
fc+
fm,&
fc -
fm
For instance, if the carrier is at 1.0MHz, & the modulating frequency is at 1.0kHz.
This results in a lower sideband at 0.999MHz, & an upper sideband at 1.001MHz.
These sidebands, the original carrier, & the original modulating frequency will all appear at the modulator output.
Any of them, or any combination can be selected or rejected by the use of filters.
Imagine the sampling was a switch in series with one of the wires to a loudspeaker, and there is a low pass filter before the loudspeaker. OK. Now, this is what I think:
* If we analyze the signal at the loudspeaker terminals, we see that it is a series of pulses of varying amplitudes.
* Speech or music is heard to the same quality as if we were feeding the speaker with the original (pre-sampled and filtered) analogue signal.
* That although modulation has been involved (PAM is involved), there is no need to demodulate the audio signal. (Unless the filtering is seen as demodulating process).
That's my understanding. Am I correct? Thanks.
Imagine the sampling was a switch in series with one of the wires to a loudspeaker, and there is a low pass filter before the loudspeaker. OK. Now, this is what I think:
* If we analyze the signal at the loudspeaker terminals, we see that it is a series of pulses of varying amplitudes.
* Speech or music is heard to the same quality as if we were feeding the speaker with the original (pre-sampled and filtered) analogue signal.
* That although modulation has been involved (PAM is involved), there is no need to demodulate the audio signal. (Unless the filtering is seen as demodulating process).
That's my understanding. Am I correct? Thanks.
In your example, the speaker is, effectively, a Low Pass Filter.