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Any good practical book on Adaptive filtering?
Hi, I'm looking for a good book on adaptive filtering, in the style of Richard Lyon's 'Understanding digital signal processing'. So far, everything I've found involves quite complicated formulas straight from page 1, with very little clear and intuitive description of how this all works. Number one on this list wo... 20 Jul 2010 06:58
Matlab 'mfilt.cicdecim' and 'filter' functions
I am trying to see the response of a CIC filter for a set of input data. I created the filter using the function Hd = mfilt.cicdecim(100, 1, 3) set(Hd, ... 'InputWordLength', 3, ... 'InputFracLength', 0, ... 'FilterInternals', 'FullPrecision'); I have some signed data in decimal in an ... 12 Jul 2010 16:15
Complete ADSP 218x environment on eBay...
Just saw this... Seems to include a full VisualDSP++ license & PCI-based JTAG ICE... Mike ... 12 Jul 2010 15:08
Experience all New Kenlighten and grow your contacts and knowledge
Dear friends, I am writing to share with you about a unique social knowledge network - Kenlighten . At Kenlighten you can register for a free account and create online tutorials in your area of expertise by uploading your ebooks, presentations, articles, videos, audio files. You can a... 12 Jul 2010 01:58
LMS maximum step size
Hi all, w(n+1) = w(n) + mu*e(n)*x(n) Can anyone tell me how to determine the maximum mu? I read some materials that mu should be less than 2/lambda_max, where lambda_max is the maximum eigen value of the autocorrelation matrix of the input. Take the 50Hz noise cancellation filter in Widrow's paper as an exampl... 15 Jul 2010 09:50
Time-Varying Systems
If you have a simple relationship y(t)=ax(t) where a is a constant then y'(t) = ax'(t) where ' is derivative wrt time. If however a is time varying then we get y(t)=a(t)x(t) y'(t)=a(t)x'(t) + x(t)a'(t) ie an extra term. In adaptive filters we derive the case for constant weights and then assume... 12 Jul 2010 17:22
invert filter bank of parallel second order sections?
Hi, I've a filter expressed as a parallel bank of (~50) second order sections. Each second order section represents a high-Q resonant feature. I need to invert the filter: i.e., if F = A1+A2+... and y = Fx, I need F^{-1} s.t. x = F^{-1}y. (F is invertible.) I'm stuck. Round-off precludes conversion to transfer ... 12 Jul 2010 14:01
THD without using an FFT
I am a C programmer working on an embedded application and I am looking into finding a less processor intensive algorithm for calculating the Total Harmonic Distortion than using an FFT. In my search, however, almost every calculation involves looking at each of the harmonics. Is there any way to subtract out the fu... 13 Jul 2010 21:55
Exocortex.DSP FFT implimentation? What should my result look like.
>bump . . .anyone have some new light to shed? Thanks Carmen I am aware that this is a really old thread, but seeing as Carmen came back years after the original posting, I though I would just shed what little light I can on the matter in case somebody else is still asking the same question. I was stuck th... 11 Jul 2010 18:11
multiple clock design
We are designing a circuit in FPGA which has many modules and each operating at different clock frequencies (but,each clock will be multiple of other). each module will be connected to other module in sequential fashion. Module A (20KHz) --> Module B (10KHz) --> Module C (2KHz) --> Module D (10KHz) --> Module E (20... 15 Jul 2010 01:15
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