I work as audio applications engineer, dealing with common speech and audio signal processing algorithms (echo cancellation, noise suppression, dynamic range control, automatic gain control, equalization) and associated fixed point implementation specifics. My work involves "tuning" the end-to-end system to improve the sound quality subjectively and objectively. I'm looking forward to reading and contributing to various discussions on this forum, to improve my own understanding of various signal processing techniques as well as help others looking to learn more in this exciting and beautiful engineering discipline.

Re: Computing the delay from a room response FIR filter

Reply posted 6 months ago (05/06/2020)
Awesome! Thank you!gd1 and gd2 are the same and they also match with MATLAB's built-in grpdelay function which can be used directly on [num,den] coefficients.

Re: Computing the delay from a room response FIR filter

Reply posted 6 months ago (05/05/2020)
Thaks @kaz.Could you tell me why the sine sweep is better than white noise?Also, I tried your method (still with white noise) for estimating the group delay by taking...

Re: Computing the delay from a room response FIR filter

Reply posted 6 months ago (05/05/2020)
Hello David,Thank you for your answer!The 6 samples delay was just an example to illustrate the question. The real latencies I am dealing with are on the order of...
Hello,I'm trying to figure out the best way to compute the delay from a room response filter. This room response filter was measured by playing white noise from...

Re: Track delays between two speech signals over time

Reply posted 3 years ago (03/01/2018)
I ended up using a normalized cross-correlation and the variation in delays from frame to frame (in valid frames where there should not be any delay variation) is...

Re: Track delays between two speech signals over time

Reply posted 3 years ago (02/27/2018)
I ignore noise frames by simple mean squared energy computation. But otherwise I compute the cross-correlation over the entire signal, but while looking at the output...

Re: Track delays between two speech signals over time

Reply posted 3 years ago (02/27/2018)
the python code looks like this:import os, sys, fnmatch, mathimport numpy as npimport matplotlib.pyplot as pltfrom scipy.io import wavfile as wffrom scipy import...

Re: Track delays between two speech signals over time

Reply posted 3 years ago (02/27/2018)
Yes, the EC does adapt slowly, and I'm looking at a converged section without interfering double talk to ensure that the filter is stable. I should have added that...
Hello,I'm trying to examine the delay between two speech signals over time by using cross-correlation over small chunks. I'm doing this in python with the approach...

Re: Proof of Z-transform of M-Fold Decimator/Compressor

Reply posted 4 years ago (10/21/2016)
Thank you Professor Smith!I think I understand what you described, but I'll need to spend more time on it and follow Tim's approach (wait for the rest of my brain...

Re: Proof of Z-transform of M-Fold Decimator/Compressor

Reply posted 4 years ago (10/21/2016)
Hi Tim,Thanks, I get it now. Yes, I like the approaches given in the books because they provide a very good intuition on the spectral behavior.Regards,Srikar
Hello,An M-Fold decimator is given by,y(n) = x(nM)Y(z) = Sum_-inf_to_inf [ x(nM) * z^-n ]Pretty much every book that shows the derivation defines a comb sequence...
In general, yes, to avoid aliasing you need to pre-filter. But this identity is simply to show that a cascade of L-interpolate and M-decimate is interchangeable...
Dr Mike and Rick,Thank you so much for providing useful and clear explanations. I can now clearly see why this identity is true. Yes, this is precisely the problem...
If you cascade an M-fold decimator and an L-fold interpolator, it is said to be equivalent to interchanging the order only if  L and M are relatively prime (GCD...

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