Re: Continuous-time DSP with no sampling

Yannis wrote: 
In principle, sampling is not necessary in order to do filtering
digitally. This is discussed in the following paper:

Finally read at least some of the paper. Look at Stark and Tutuer's , Modern Electrical Communications Systems, Prentice Hall 1979, in section 4.3 entitled, "Practical Aspects of Sampling." They assert that actual samplers have in effect some integration time. In your discussion on in-band quantization aliasing it looks like you use an ideal sampler. If the sampler has some integration associated with it, I believe that the harmonics will be a bit more attenuated, and thus less energy will fold in. I think your presentation about harmonics aliasing is kind of a worse case argument.




Y. Tsividis, "Digital signal processing in continuous time: a
possibility for avoiding aliasing and reducing quantization error",
Proc. 2004 IEEE Int. Conf. on Acoustics, Speech, and Signal Processing,
vol. II, pp. 589-592, Montreal, May 2004.
(If you are interested but cannot obtain this paper, please let me
know.)

In the above paper, I discuss a method to do DSP in continuous time,
without sampling, resulting in a system with no aliasing. The system
has no quantization error at non-harmonic frequencies, and exhibits
10-15 dB lower total quantization error than classical DSP, for a given
number of bits. Power dissipation decreases when the input frequency is
low, or in general when there is little activity. However, although
breadboard measurements and simulations show that the idea works, there
is a lot of work to be done before one can know whether all this is
practically feasible. This work is at the early research stage, and no
commercial feasibility is claimed at this point.

I would be very interested in the opinion of DSP experts on this idea.
We are currently looking for an appropriate application in order to
demonstrate the concept. I welcome any comments!

Yannis Tsividis Columbia University

.

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