Adaptive Filter | ADPCM | ADSP | ADSP-2181 | Aliasing | AMR | Anti-Aliasing | ARMA | Autocorrelation | AutoCovariance | Beamforming | Bessel | Blackfin | Butterworth | C6713 | CCS | Chebyshev | CIC Filter | Circular Convolution | Code Composer Studio | Comb Filter | Compression | Convolution | Cross Correlation | DCT | Decimation | Deconvolution | Demodulation | DM642 | DSP Boards | DSP/BIOS | DTMF | Echo Cancellation | Equalization | Equalizer | ETSI | EZLITE (Ez-kit Lite) | FFT | FFTW | FIR Filter | Fixed Point | FSK | G.711 | G.723 | G.729 | Gaussian Noise | Goertzel | GPIO | Hilbert Transform | IFFT | IIR Filter | Interpolation | Invariance | JTAG | Kalman | Laplace Transform | Levinson | LPC | McBSP | MIPS | Modulation | MPEG | Multirate | Notch Filter | Nyquist | OFDM | Oversampling | Pink Noise | Pitch | PLL | Polyphase | QAM | QDMA | Quantization | Quantizer | Radar | Random Noise | Reed Solomon | Remez | Resampling | RTDX | Sampling | Sharc | TI C6711 | Undersampling | Viterbi | Wavelets | White Noise | Wiener Filter | Windowing | XDS510PP | Z Transform

Re: Adaptive Filtering in Real Time

"Zachary Rimkunas" wrote in message news:_rnOa.16264$Ha.5943@nwrdny02.gnilink.net... > Hi, > I am a newbie trying to do real time adaptive filtering on an Analog > Devices blackfin 21535. I'm not trying to do anything too complex, just a > simple LMS algorithm (1 mic, 2 loudspeake...

Re: DSP technique to estimate the amplitude of the signal

praveen wrote: > > Hello Jim, > But i don't know the exact freqency of f0 i.e if f0 is 200kHz then it > will be anywhere between 195 to 205 kHz. > > waiting for reply > praveen Set up a bandpass filter that cuts off outside those limits, and you will find that the filter removes ...

Re: LMS limitations?

"Zach R." wrote in message news:ZzkTa.61434$kI5.13224@nwrddc02.gnilink.net... > Hi, > The sample rate is 7040 Hz (the lowest that my ADC will go). As far as > the order goes, I would like it to be longer but the program takes too long > to run when I use a larger order (such as ...

Re: Linear Time Variant systems

"Rune Allnor" wrote in message news:f56893ae.0307241918.67e54977@posting.google.com... > Hi all. > > I've been thinking a bit about processing of signals that have propagated > through an ocean waveguide that change, either because of waves on the > surface, tides, or for other ocean...

Re: Audio subtract function thru programming to create multitrack from stereo sources

usenet@apple-o.com (Mad Scientist) wrote in message news: ... > I'm looking for a utility or to program one that can take a sound file > (WAV) and take a second file, and isolate what the two files have in > common, and what is different about them. > > For instance if you have two WAV f...

Re: problem with reference signal in LMS algoritm

"ralph" wrote in news:1132325514.108631.22490 @o13g2000cwo.googlegroups.com: > I want to use LMS algoritm to adaptive noise cancellation. The problem > is that I have only primary signal and thats all what I have, I don't > know anything aboyt reference signal, and my question is: how ca...

kalman filter implementation and basics

Hello, I am very new to adaptive filter. Is there any article about kalman filter basics and its implementation (basically i am first looking for matlab implementation). The article in the net is very difficult to understand and implement. Is there any article or reference which easier to under...

Re: What's difference between Active NC and Adaptive NC

To answer the more general question first. Active noise cancellation involves producing a physical anti-noise to cancel the undesired noise. An application of this is the reduction in the noise of a manufacturing machine in the area of the operator. The disadvantage is that the noise reductio...

Re: Basic Filter Query

"Sandeep Chikkerur" wrote in message news:d5d88eb5.0309030114.3374f77e@posting.google.com... > Hi, > > The order of a digital filter is the number of previous inputs (stored > in the processor's memory) used to calculate the current output. > > My doubt is, how can one estimate the...

can this be done in real time???

I am thinking of programming a dsp development board (no particular model chosen yet) to implement an RLS adaptive filter but with a Volterra Series front end to provide an adaptive non-linear equaliser. If this doesn't mean anything to you, take a look at IEEE Trans on Sig Proc Vol 41 No 3 March...

Re: anti-Larsen circuit

Jan Rørgård Hansen wrote: > "Jerry Avins" wrote in message > news:bn68vh$tb$1@bob.news.rcn.net... > > > Jan Rørgård Hansen wrote: > > > > > > > Hi > > > > > > Can anyone explain the principle of an anti-Larsen circuit or give > > > references. > > > I know that it is used ...

Re: what are the training algorithm you guys are using for adpative filtering for image processing?

"walala" wrote in message news:bna5kv$890$1@mozo.cc.purdue.edu... > Dear all, > > I have a question about adaptive filtering for image enhancement. > > I want to design an algorithm for image enchancement. It should work on most > of images. So I need to make it adaptive. > > ...

Speech Recognition

What algorithms are commonly in speech recognition? I have planned to use Fourier Transform (how many samples do you think is good? 4096?) and a adaptive filter. What do you think? Thanks in advance //Spike ...

DSP (book) library (was: Oppenheim & Schafer)

Al Clark wrote in message news: ... > ricklyon@REMOVE.onemain.com (Rick Lyons) wrote in > news:3fb4021e.48006015@news.west.earthlink.net: > > > > > Hi Rune, > > > > if you want a complete DSP library (which may > > cause you to go to your local bank for a financial l...

Re: Acoustic Echo Cancellation source

"john" wrote in message news:19778961.0311250632.cd2e23a@posting.google.com... > Hello , > > We are 2 students writing a VideoConference application on > linux; > We want to use microphones for recording and loudspeakers for hearing. > Now , test we made so far show that there is a...

Why minimising in the mean-error sense.

Hello group. Quote from Simon Haykin's book "Adaptive Filter Theory", 4th edition, on the Wiener filter (chap. 2): "We now summarize the essence of the filtering problem with the following statement: Design a linear discrete-time filter whose output y(n) provides an estimate of a desired...

Re: Question about ISI and square root cosine filter

freelait2000@yahoo.com (Jeff) wrote in message news: ... > Hi, > I want to simulate two cascaded square root cosine filters to decrease > ISI in baseband using Matlab. The signal modulation is I, Q type, such > as QPSK. When I begin the job, I find there are some thing unclear to > me. Th...

Re: Help with Digital PLL motor control

Hello Fred, Jim First, thank you both for your response. I read your original post Fred, but ran out of time to respond, and I see there is a new post so I'll try to answer all your questions (and pose some new ones of my own) in one shot... In article , fmarshallx says... > > "J...

Re: Levinson algo. wtih complex coef. required

Haykin also has the complex-valued derivation in his "Adaptive Filter Theory" book, Maurice Givens allnor@tele.ntnu.no (Rune Allnor) wrote in message news: ... > Bob Cain wrote in message news: ... > > Bob Cain wrote: > > > > > > On page 865-868 of the third edition of Dig...

Questions about QR-RLS algorithm and antenna beamforming

Hi, I want to simulate an adaptive beamforming algorithm (using QR-RLS). From what I learned, I have the following thoughts and questions. Because I am not sure whether they are right or not (and the question is still unsolved), I want to get your answer to these. From «Adaptive Filter TheoryÂ...

Re: Filtering base sound

It almost certainly won't work - because of phase and amplitude miss-matching. The adaptive filter goes a long way in trying to match the phase and magnitude up again but even it is not perfect. Naebad ...

Re: echo canceller

"Jerry Avins" wrote in message news:401343c9$0$7333$61fed72c@news.rcn.com... > Randy Yates wrote: > > > Jerry Avins writes: > > > > > > > michael yarwood wrote: > > > > > > > > > > data call (e.g. fax,modem etc.) setup signals that the echo cancellers > > > > should b...

Which noise?

Dear chaps, I have a problem. I have a signal recorded from a ultrasonic transducer of a chemical reaction. It is rich of noise, expecially the white one. I tried to filter with normal filter (butterworth band pass), but I did not obtained a lot of success. So, I thought was better delete the...

noise..how to regulate it?

Dear chaps, I have a problem. I have a signal recorded from a ultrasonic transducer of a chemical reaction. It is rich of noise, expecially the white one. I tried to filter with normal filter (butterworth band pass), but I did not obtained a lot of success. So, I thought was better delete the...

Re: Speech Hiss Reduction

John Cristion wrote: > I'm looking for an algorithm for hiss reduction of speech (and/or other > aural) signals. > > Any ideas? > > Thanks, > > John > > I have seen an adaptive filter design in the amateur radio literature (I think it was "QST", but it may be "QEX") t...

Re: Adaptive arrays

> Subject: Adaptive arrays > From: forevermav2001@yahoo.com (John) > Date: 5/7/2004 11:43 PM Eastern Daylight Time > Message-id: > > Hey, > I need to design an adaptive array for automatic interference > rejection using the LMS algorithm in MATLAB. There is a target signal > cos(2*pi*f...

Please help me regd. adaptive filter techniques

Hi there, I´m going to start my thesis work on Inertial navigation system for mobile Robot(Gyro-inclinometer),so i need to use filter algorithm.I´ve some idea about kalman filter,but i would like to know which filter is most suitable to my project.What are the alternative adaptive filter t...

Re: NEW FREQUENCY ESTIMATION METHOD

dt@soundmathtech.com (Dmitry Terez) wrote in message news: ... > Hello, everybody, > > Thank you for all of your comments. I felt like I needed to sample > people's opinions on the issue and there is no better place to do it > than here. > > I can try to answer some of your question...

Re: Levinson-Durbin Adaptive Filter....How?

OK, I'm still being a bit thick! What would anyone recommend as a simple test signal that I could use to put through the L-D block in simulink and what should I expect as the output? If I have only one signal can I put that through the L-D? I know that the LD solves for a number of equatio...

Re: 60 Hz Hum removal

"Rob Hutchinson" wrote in message news:10e8scqobsj7f3@corp.supernews.com... > What is the preferred method for removing 60 hz hum from a signal without > wiping out signal info around 60 hz? A 60 hz notch filter would not be > useful because it would attenuate the signal as well. I'm i...

Re: Source separation of harmonic sounds when f0 and harmonics are given??

Are the notes they playing "harmonically related" i.e. is the fo of the flute a mutliple of the fo of the cello? If they are not related, some adaptive filter approaches may be possible. -Shawn Steenhagen Applied Signal Processing "Arie" wrote in message news:64a73096.0407031424.25...

Re: Measuring Lag of a Black-Box Filter

In article , Tim Wescott wrote: > axlq wrote: > > I'm trying to develop a 1-dimensional adaptive low-lag tracking > > filter, and one of the early steps in this process is to understand > > what "lag" actually is, and how to measure it, for non-adaptive > > filters. To me, "lag" is ...

Re: Financial Work

In article , Jerry Avins wrote: > What assurance does one have that aliasing artifacts are smaller than > some epsilon? When a single number is recorded once a day at an > arbitrary time -- closing, say -- what is the upper limit of frequency > response implied by that? If you're think...

Echo Canceller: Measuring Performance

I'm simulating an echo canceller in Matlab, using a FIR adaptive LMS filter. I'm using the standard noise cancelling approach: -adaptive filter input u(n) = far end speech -adaptive filter output is y(n) -impulse response of echo path is h(n) -near end speech is ne(n) -d(n) = ne(n) + u(n) (*) ...

Litterature on adaptive MISO systems.

Hello group, I have been assigned a bachelor project (final thesis) and I'm looking for litterature for some particular subjects. Mainly, the objective is to cancel noise using an adaptive filter. The system consists of two noise sources and one primary signal, in short, an adaptive MISO filter....

Re: Filtered-X LMS - what's the purpose?

mkl wrote: > On Sun, 26 Sep 2004 00:54:23 +0200, "Jack L." > wrote: > > the problem with the LMS algorithm is that for systems where > eigenvalue spread is large, the convergence rate can be slow. > filtered-x algorithm is one of the family of self-normalizing lms like > algorithms ...

Re: FIR Lattice Filter Reflection Coefficients in Terms of Direct Coefficients?

You can find the relevant information in the following textbooks: (1) "Advanced Digital Signal Processing", J.G.Proakis, C.M.Rader, F.Ling, C.L.Nikias, ISBN 0-02-396841-9, Chapter 4 (Linear Prediction and Optimum Linear Filters), pp 219-221 (2) "Optimum Signal Processing, An Introduction, ...

c6711 dsk & adaptive filter problem

Hi everyone, I'm a student working on modified Griffith-Jim beamforming for my masters project. I'm implementing this algorithm in TI's c6711 dsk, and also using PCM3003 codec to get two input's from the microphones. I'm trying to reduce the background noise using an adaptive filter (its bas...

Re: In reality, how do people measure autocorrelation function?

In reality, if the input to the system is also available, one could easily identify a system by using an adaptive filter in the identification mode and one could decorrelate the output samples without even knowing the filter under consideration by using an adaptive filter in the deconvolution mo...

Re: V.27ter

Hi Luiz, We have no idea what the channel will look like, other than some overall constraints - e.g. the 8kHz sampling in the PCM sections of the channel will definitely impose hard frequency limiting. The received signal, after moving to baseband and root raised cosine filtering, can be ...