Not just a DSP thing
Adaptive filtering isn't just a DSP thing. There's a big beautiful body of mathematical literature put together by guys like Wiener and Kalman on the art of detection, estimation, and optimal and adaptive filtering. It's probably fairer to call it a "linear algebra" thing than a "DSP" thing.
Current implementations are done in DSP because computers are fast enough and cheap enough that it's the only reasonable way to do it. But Wiener did a lot of work on analog mechanisms that drew upon the same body of mathematics.
I've added Echo cancellation and linked to Linear prediction. There is no hint as to what Channel equalization and Channel identification should refer to so I've boldly removed those. Feel free to restore them if you can explain to us what they're about. --Kvng (talk) 20:01, 2 September 2008 (UTC)
Someone could add http://www.eee.strath.ac.uk/r.w.stewart/adaptivejava/begin.htm in external links. Good examples with figures. —Preceding unsigned comment added by 18.104.22.168 (talk) 17:28, 10 November 2008 (UTC)
What specifically is too technical about this article? It seems clear and simple to me. Only the block diagram section even has any equations (which seems fine, since it's obviously for more advanced readers), and they're all nicely explained. —Keenan Pepper 18:06, 24 June 2010 (UTC)
- I have reworked the lead to improve accessibility - Discussion of coefficients in the lead was unnecessary, for instance. I think that's enough for me to remove the Technical tag. It would be nive to increase the word to formula ratio in the Block Diagram section. --Kvng (talk) 16:25, 25 January 2011 (UTC)
What use is the adaptive filter shown in the block diagram?
In the example diagram the right side input to the system is the desired signal. If you already have the desired signal, d(t); why use the filter to predict it??
It would make more sense to me if a signal that correlates with v (in the EKG medical example above it would be the line voltage) is the input on the left side of the system and the corrupted signal, x(t)=d(t)+v(t), is the input on the right. Then the filter would output an estimate of v(t) and the result of the sum would be an estimate of d(t).
I could be way off since this isn't my field of expertise. Please set me straight if I'm off in left field somewhere. I'm not sure if my proposed circuit would work, but it sure seems to me like something is wrong since d(t) is needed to estimate d(t). Hangingman (talk) 03:49, 25 January 2011 (UTC)
- You are confused because the example and equations do not correlate well with the block diagram. In the medical example, x(n) is line noise + heartbeat, d(n) is line noise only. The useful signal is e(n) which, once the adaptive filter has done it's magic, should be just the heartbeat. --Kvng (talk) 16:25, 25 January 2011 (UTC)