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signal with high- and low-frequency components, real-time application

Started by Unknown May 7, 2008
On 8 Mai, 14:35, Richard Owlett <rowl...@atlascomm.net> wrote:
> stefan_bai...@yahoo.com wrote: > >[snip] > > The uploaded signal was just an example, so another sequence could > > look different with other underlying signal components on it. But > > anyway, thanks for your analysis! > > "Noise" in my situation is the part of the signal I need, the > > superimposed lower frequencies are due to disturbance - I know, > > normally it's the other way round;-) > > Not so abnormal as you may think.
I've seen that sort of stuff in systems that were subject to strong electrical interference. One example was near a furnace in a metal plant. Another was a sound recording in a metal shop where somebody welded nearby. Rune
On 10 Mai, 23:32, "SteveSmith" <Steve.Smi...@SpectrumSDI.com> wrote:
> Hi Stefan, > I think your question may be more basic than the other posts have assumed. > I understand that your problem involves receiving one sample each second, > and I have looked at the data you provided. &#4294967295;Say you are at some instant of > time, t, and have just received a sample. &#4294967295;Of course, you have also > received previous samples, at time t-1, t-2, t-3, and so on. &#4294967295;I think you > have the impression that a filter cannot give you an output at time t, that > represents what is happening in the signal at time t. &#4294967295;This is not correct. > For example, consider taking the sample at time t, and subtracting the > average of the last 20 or so samples. This is an FIR high-pass filter that > only uses past values of the signal. &#4294967295;This is the first algorithm that you > should try (it is even optimal for many problems of this type). Experiment > by adjusting the number of samples in the average. &#4294967295; > > If you want better performance, try weighting the average so that more > recent samples are given more importance. &#4294967295;Alternatively, try the simple > "single pole" IIR filter described below. &#4294967295;This is the digital equivalent > of an analog RC filter. Good luck! > Regards, > Steve > > http://www.dspguide.com/ch14.htmhttp://www.dspguide.com/ch15.htmhttp://www.dspguide.com/ch19/2.htm > > &#4294967295; that 'm not sure that the answers you have receive really address your > question. So let me give it is ahow you have >
Steve, thanks for your comments! The type of filter you described is some kind of moving-average. The more samples you use, the more delay you get. I've already experimented with such an algorithm, problem is, that fast changes in the signal leed to errors opposite to the direction of the measured signal's change. This is not the solution for me. I'll look at the single pole IIR filter you mentioned... The other idea is to estimate/predict measurements for the next few seconds. This way might be the best solution, but I'm not in this field and time for finding an appropriate algorithm is limited. Anyway, hints are always welcome!