In principle, sampling is not necessary in order to do filtering digitally. This is discussed in the following paper: 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
Continuous-time DSP with no sampling
Started by ●November 2, 2005
Reply by ●November 2, 20052005-11-02
Hello Yannis, Yes the paper sounds interesting, but if you could, can you place a .pdf of it on your web site and provide a link to it? Thanks, Clay "Yannis" <ytctdsp@yahoo.com> wrote in message news:1130977760.809235.133270@f14g2000cwb.googlegroups.com...> In principle, sampling is not necessary in order to do filtering > digitally. This is discussed in the following paper: > > 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.) >
Reply by ●November 2, 20052005-11-02
Reply by ●November 2, 20052005-11-02
Can I have a copy? I don't know if the idea is similar to "A Nonaliasing, Real-Time Spatial Transform Technique", from Theseus Researh, Inc. [RJiang]
Reply by ●November 2, 20052005-11-02
"Yannis" <ytctdsp@yahoo.com> writes:> I will be happy to send you a copy. > > YannisCan I have a copy too, please? -- % Randy Yates % "My Shangri-la has gone away, fading like %% Fuquay-Varina, NC % the Beatles on 'Hey Jude'" %%% 919-577-9882 % %%%% <yates@ieee.org> % 'Shangri-La', *A New World Record*, ELO http://home.earthlink.net/~yatescr
Reply by ●November 2, 20052005-11-02
in article 1130977760.809235.133270@f14g2000cwb.googlegroups.com, Yannis at ytctdsp@yahoo.com wrote on 11/02/2005 19:29:> 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!thanks for sending the paper, Yannis. well, i got a couple of comments/questions: 1. what is new or novel in concept? when i read your first post, i thought "well something like this could be done with a flash A/D and then analog delay lines applied to the bits and combinatorial logic to do the arithmetic on the bits to accomplish some algorithm." this appears to be what your paper is suggesting. i had a prof at the University of North Dakota suggest this to us in a class in 1978. it's like a fleeting "what if..." idea. 2. how is the analog delay done? transmission line? surface acoustic wave (SAW) device? (i see that you deal with this a little in section 3.) 3. what about A/D "glitches". these glitches happen when bits, in a 2's complement representation, do not switch simultaneously when the quantized value changes. i.e. in an 8-bit flash A/D, what happens when you go from 01111111 (127 decimal) to 10000000 (128)? what if the MSB flips a few nanoseconds before the other bits flip and your A/D output passes through the 11111111 (255) state. that is a "glitch". in discrete-time, we were able to deal with these by waiting until all bits settle before latching the word. (i have a paper published in JAES in Nov. 1988 about this on the D/A end, if you're interested. since you're at a nifty ivy league school, your library should have that old JAES.) 3a. now i know there is this "Gray code" that could be used (if a flash A/D was designed to output Gray Code) in which only one bit toggles as your continuous (in amplitude) analog signal passes from one quantized level to the next, (see http://en.wikipedia.org/wiki/Gray_code ) and then you would never get more than one bit that should be changing at a time. accordingly, the combinatorial logic would be modified to do the same arithmetic, but with the Gray encoded words instead of 2's complement. 4. but, even so, what if the analog delay lines on the bits are not perfectly matched? then you get glitches in your output because of a momentary error in the data your arithmetic logic is looking at. i realize that you mention some similar issues in section 3, but saying "The result is not serious, as the energy of such spikes is primarily at out-of-band frequencies, and no new frequency components are introduced to the output spectrum" does not seem to be, to be taking the glitch problem sufficiently seriously. 5. and finally, what if the arithmetic logic does not produce the correct internal states at precisely simultaneous times. again, the result will be glitches. sorry, if you find my comments critical, but they're meant to be given and taken in good faith. -- r b-j rbj@audioimagination.com "Imagination is more important than knowledge."
Reply by ●November 3, 20052005-11-03
Randy Yates wrote:> "Yannis" <ytctdsp@yahoo.com> writes: > > >>I will be happy to send you a copy. >> >>Yannis > > > Can I have a copy too, please?Jes: me too. I want to know where the numbers come from. jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ●November 3, 20052005-11-03
robert bristow-johnson wrote: (snip)> > 3a. now i know there is this "Gray code" that could be used (if a flash A/D > was designed to output Gray Code) in which only one bit toggles as your > continuous (in amplitude) analog signal passes from one quantized level to > the next, (see http://en.wikipedia.org/wiki/Gray_code ) and then you would > never get more than one bit that should be changing at a time. accordingly, > the combinatorial logic would be modified to do the same arithmetic, but > with the Gray encoded words instead of 2's complement. >(snip) Robert, in general, consecutive analog samples will differ by more than one quantisation level, and so the feature of Gray code that you mention will be of no particular significance. Regards, John
Reply by ●November 3, 20052005-11-03
robert bristow-johnson wrote:> [...] > >3a. now i know there is this "Gray code" that could be used (if a flash A/D >was designed to output Gray Code) in which only one bit toggles as your >continuous (in amplitude) analog signal passes from one quantized level to >the next, (see http://en.wikipedia.org/wiki/Gray_code ) and then you would >never get more than one bit that should be changing at a time. accordingly, >the combinatorial logic would be modified to do the same arithmetic, but >with the Gray encoded words instead of 2's complement. > >In current usage, a flash converter has a very narrow window analogue gate in front of it. After snapshoting with this, we wait one settling time of the comparator stack, and get a stable output. Successive outputs of large signals near the Shannon rate are separated in value by almost the whole swing of the converter. If you have a scheme where all successive outputs must be passed through, in Grey code style, your maximum operating speed would be truly snail's pace. Steve
Reply by ●November 3, 20052005-11-03
"Yannis" <ytctdsp@yahoo.com> wrote in message news:1130977760.809235.133270@f14g2000cwb.googlegroups.com...> In principle, sampling is not necessary in order to do filtering > digitally. This is discussed in the following paper: > > 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 >Well I always say 'Don't let triffling problems like the practicalities of whether it works or not get in the way of a good paper'!! McC
