c6711 dsk & adaptive filter problem

Started by yoga in comp.dsp16 years ago 1 reply

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...

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 based on NLMS). I have done the implementation in c-code in CCS, but I have a bit of problem with t...


Problems after freezing FIR coefficients of Acoustic Echo Canceller

Started by johan kleuskens in comp.dsp15 years ago 12 replies

Hi, We are currently working on an acoustic echocanceller based on the well know NLMS principle. This echocanceller works fine as long as we...

Hi, We are currently working on an acoustic echocanceller based on the well know NLMS principle. This echocanceller works fine as long as we feed the echocanceller with a echo signal that is generated by an audio processing program. When using this ideal echo signal, freezing the FIR coefficients works like it should: the echo is still cancelled because the FIR tabs contain a represen...


Writing an echo canceller

Started by Duane in comp.dsp15 years ago 2 replies

Hey everyone, I started writing an echo canceller as a side project, but am already hitting a few snags.. I have a few research papers on the...

Hey everyone, I started writing an echo canceller as a side project, but am already hitting a few snags.. I have a few research papers on the floor here I've been playing with. My first problem is with the NLMS algorithm.. Dispite implementing several versions of them, I still always seem to have a problem with it going unstable.. I'm not even sure if it's unstable or not, but it can ge...


adaptive filter

Started by ldkha1979 in comp.dsp15 years ago 1 reply

hi all, i have to design an adaptive filter using Matlab. What i need is to find a symmetric FIR filter from 2 signals, desired and input. I...

hi all, i have to design an adaptive filter using Matlab. What i need is to find a symmetric FIR filter from 2 signals, desired and input. I don't know why my results cannot be converged. I'm using NLMS algorithm. Here are my code. Can anyone help me to find my wrong part, or can give me a suitable one? Thks so much function h = adp_filter(d,x,f0) % h = adp_filter(d,x,f0) % Inputs: % ...


Can anybody tell me sth. about Leaky NLMS

Started by Jing in comp.dsp14 years ago 3 replies

Hi, I am trying to implement leaky NLMS for adaptive beamforming, but I found that the noise reduction performance is worse than that of NLMS....

Hi, I am trying to implement leaky NLMS for adaptive beamforming, but I found that the noise reduction performance is worse than that of NLMS. The data I used are simulated data, and I assume the noise and speech sources are well seperated. Can anybody give me some suggestions about applying leaky NLMS for GSC? Is Leaky really better than that of NLMS? Thanks a lot! Best Regards, Jing ...


echo canceller initial convergence

Started by Anonymous in comp.dsp14 years ago 2 replies

Hi All, I am trying to implement a long tail (selectable 512/1024 taps) echo canceller thats compliant at least with tests 2 and 3 of g.168....

Hi All, I am trying to implement a long tail (selectable 512/1024 taps) echo canceller thats compliant at least with tests 2 and 3 of g.168. The implementation is using leaky NLMS, coded in C using 16 bit fixed point arithmetic (the coefficients are also stored as 16 bit, not 32). I have two concrete questions and, in addition to those, wanted to put out a "general call for advice". The ...


Unstability in NLMS algorithm for Echo Cancellation

Started by MMHafezi in comp.dsp14 years ago 9 replies

Hi every body, I have implemented a line echo canceller based on NLMS algorithm (TI app. note SPRA129) but sometimes the coeficient...

Hi every body, I have implemented a line echo canceller based on NLMS algorithm (TI app. note SPRA129) but sometimes the coeficient updatation falls into a bad state and echo will amplify. I changed the algorithm in a way that when I find this situation has happend I reset the coeficients to zero so it takes time again for the coeficients to converge and during this time you will hear echo. B...


Implementing NLMS for Echo cancelation in Speech

Started by ilan_sin in comp.dsp14 years ago 6 replies

Hi I am trying to write an acoustic/line echo canceller. I choose the quite well known nlms algo. First I tried to implent it in...

Hi I am trying to write an acoustic/line echo canceller. I choose the quite well known nlms algo. First I tried to implent it in matlab. When I checked the algo with any "test input" like sin wave, zigsaw etc, I did get convergance. Then I tried it with speech, but failed. I tried to improve it by implementing anti-correlation filter ( Pre-whitening). I did saw minimal improvments in the...


Regarding Real Time Problems of AEC

Started by Aparna Ram in comp.dsp13 years ago

Dear All, I have implemented AEC with and with out subbanding using NLMS algorithm and that code is working fine for all recorded...

Dear All, I have implemented AEC with and with out subbanding using NLMS algorithm and that code is working fine for all recorded audio files. But for real time audio files this algorithm is not effecting. If any one know the reason behind this then please give me your valuable suggestions. Thanks in advance. Regards, Aparna Ram.K.


NLMS Step Size Question

Started by jai....@gmail.com in comp.dsp13 years ago 2 replies

I'm running into difficulty implementing the NLMS algorithm, w.r.t. step size.. I'm using Matlab to model the algorithm, and here is the core...

I'm running into difficulty implementing the NLMS algorithm, w.r.t. step size.. I'm using Matlab to model the algorithm, and here is the core of the calculation to update the tap weights: nlms_coeff = af_parms.mu/pwr(i) * en; af_parms.state.weights = af_parms.state.weights + xn .* nlms_coeff; - pwr(i) is the calculated signal power for sample 'i' - xn is the data samples used for 'L'...


adaptive filters; qmf sub-band versus one long filter

Started by Robert Adams in comp.dsp12 years ago 2 replies

I have been experimenting lately with adaptive filters running in the "system identification" mode. There are 2 approaches that are used; 1)...

I have been experimenting lately with adaptive filters running in the "system identification" mode. There are 2 approaches that are used; 1) A simple nlms fir filter approach, filter length = K 2) sub-band approach, where you break the signal into N critically- sampled bands, do N nlms filters of length K/N each, followed by a reconstruction filterbank. The second approach has ma...


PNLMS filter required!!!!

Started by Sharmin in comp.dsp11 years ago

Hi , I am working on adaptive filter algorithm. I have already implemented the NLMS algorithm but when I wrote the code for PNLMS algorithm ,I am...

Hi , I am working on adaptive filter algorithm. I have already implemented the NLMS algorithm but when I wrote the code for PNLMS algorithm ,I am not getting the desired output. If anyone already worked with PNLMS algorithm plz help me by sending the matlab code for PNLMS filter.


NLMS algorithm simulation

Started by chandrakanthv in comp.dsp10 years ago

Hello everyone, I am chandrakanth as can be seen by the user name :). I have a query regarding the implementation of Noise cancellation using...

Hello everyone, I am chandrakanth as can be seen by the user name :). I have a query regarding the implementation of Noise cancellation using NLMS. I have followed the theory from "Statistical DSP by Monson Hayes" I am not able to get the noise cancellation waveforms given in the text book. I am not able to debug the code for the results. Can anyone help me with it. I am pasting the code here....


Automatic gain control and acoustic echo cancellation

Started by John McDermick in comp.dsp9 years ago 4 replies

This is a general question about acoustic echo cancellation based on NLMS. If an AEC receives a speaker signal which has been processed by...

This is a general question about acoustic echo cancellation based on NLMS. If an AEC receives a speaker signal which has been processed by an automatic gain control algorithm, will that have any adverse effects on the AEC's performance? If the change in speaker gain level is too sudden or far from being smooth, doesn't that result in the AEC having to re-adapt? Thank you.


How do you define tail length for frequency domain acoustic echo cancellation algorithms?

Started by Mauritz Jameson in comp.dsp8 years ago 3 replies

For an NLMS time-domain algorithm, the tail length is defined as the number of taps of the NLMS filter, right? How is the tail-length defined...

For an NLMS time-domain algorithm, the tail length is defined as the number of taps of the NLMS filter, right? How is the tail-length defined if you are doing your NLMS in the frequency domain by adjusting the amplitudes of a 128-point FFT?


NLMS and ERL above 0dB

Started by Mauritz Jameson in comp.dsp8 years ago 27 replies

Hi I'm playing around with a traditional NLMS algorithm for acoustic echo cancellation. It seems like the NLMS algorithm have a hard time...

Hi I'm playing around with a traditional NLMS algorithm for acoustic echo cancellation. It seems like the NLMS algorithm have a hard time cancelling the echo as soon as the ERL level creeps above 0dB. Is that to be expected? The speaker signal is pure speech (no noise) and I have linearly filtered the speaker signal to have a simulated microphone signal. The adaptive filter order i...


NLMS question

Started by Mauritz Jameson in comp.dsp8 years ago
LMS

Hello, There are cases where my NLMS algorithm let's echo slip through. When I observe how the filter coefficients adapt, I see a lot...

Hello, There are cases where my NLMS algorithm let's echo slip through. When I observe how the filter coefficients adapt, I see a lot of fluctuations around the peak. Is that an indicator of non-linearities? Such as: phone case vibrations/reflections picked up by the mic which the algorithm has a hard time dealing with? It seems that the fluctuations around the peak only happens when the p...


Estimating time offset between two audio signals.

Started by Mauritz Jameson in comp.dsp7 years ago 22 replies

I'm looking for some recommendations on real-time algorithms which are able to estimate the time offset (delay) between two signals. One signal...

I'm looking for some recommendations on real-time algorithms which are able to estimate the time offset (delay) between two signals. One signal is the source signal (speaker signal). The other signal is a filtered version of the speaker signal (echo, microphone signal). Delay might be as large as 500ms. Running NLMS on downsampled signals to estimate the time offset works great if the dela...


Adaptive filter

Started by Jim craig in comp.dsp6 years ago 1 reply

Hi I am using a classic nlms adaptive filter to minimize the error between an unknown LTI system and the output of an adaptive FIR. However I...

Hi I am using a classic nlms adaptive filter to minimize the error between an unknown LTI system and the output of an adaptive FIR. However I have no control over the stimulus, which is a mix of sinusoids that change slowly over time. I know that with this input, the filter may not adapt to the true impulse response, but as long as the error goes to a small value I don't really c