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I want to measure amplitude of a sine wave input precisely with a limited resolution ADC.
As an example suppose that I have $1\textrm{ MHz}$ pure sine wave input to the $320\textrm{ Msps}$ $10$-bit ADC. I beleived there is many redundancy in data and with some signal processing I could get more Precision than $10$-bit.
Is there any way that I can do this without any change in circuit, like adding noise or other hardware change?
A sine wave has infinitesimally little bandwidth. By rotating, filtering appropriately and decimating, you can reduce the sample rate very much.
Each of these filtering operations is typically a summing operation, in which you "average" out noise (which isn't your main concern), but also get a more precise estimate for the amplitude.
Decimation in DSP is very commonly done. You'd probably want to do that anyway – 320 MS/s is really no fun to deal with, and you don't need that bandwidth.
Of course, you can also correlate with a synthesized sine, and measure the correlation coefficient to get the power/amplitude.
Other options are things like proper spectral estimators – there's a lot to choose from, including Welch's method, or Pisarenko-based approaches. Especially if your signal is noisy, these might be interesting, but it really depends a lot on what exactly you're measuring and how you model your noise.
To add to Marcus' good answer, the decimation Marcus mentions WILL increase the number of bits, down to the Spurious-Free Dynamic Range (SFDR) of your ADC. Also if you can modify your sampling rate so that it is incommensurate with your signal of interest: 320 MHz is an integer multiple of 1 MHz, so you will ...to the extent your input signal is coherent with your sampling clock ... only land on the same 320 samples in one cycle of the sine wave. This pattern repetition will lead to spurs (in this case at harmonics of 1 MHz) and cause quantization in time (if you were concerned with delay precision). By choosing a sampling clock that is incommensurate, the samples will roll over the sinewave, and since your input signal and sampling clock will be decorrelated the harmonic spurs will be significantly reduced.
For more details on how your precision will be increased by oversampling in this case with a traditional ADC, see What are advantages of having higher sampling rate of a signal?
It can be done by the least square fit of the signal to the sine wave
Signal(A, Phi, W, t) = A * sin(2 * PI * W * t + Phi)
You have three parameters to fit here:
A - signal amplitude
W - signal frequency
Phi - signal phase
If you know frequency already, there are just two parameters to be fitted.
Calculating the root-mean-square deviation you can get evaluation of the precision of the amplitude calculation
