Hi, I'm currently an undergrad who is evry new to beamforming and matlab, would greatly appreciate is someone can help me in this. After...

Hi, I'm currently an undergrad who is evry new to beamforming and matlab, would greatly appreciate is someone can help me in this. After calculating the complex weights through MVDR, they appeared in the form of w (8X1 matrix with complex values). (example) w = 0.1250 0.1079 - 0.0630i 0.0614 - 0.01089i -0.0019 - 0.1250i -0.0646 - 0.1070i -0.1098 - 0.0598i -0.1249 + 0.0037i ...

Hi All! I am not good at math. But I wanna to know the name of the curve that is used in CoolEdit to draw the wave samples. I thought...

Hi All! I am not good at math. But I wanna to know the name of the curve that is used in CoolEdit to draw the wave samples. I thought it is the Bezier Curve, but it seems not to be. Thanks!!

Hi, I have an audio application where I can selectively filter an input signal by choosing 1 of 16 Cauer filters with rather steep rolloff, 80db...

Hi, I have an audio application where I can selectively filter an input signal by choosing 1 of 16 Cauer filters with rather steep rolloff, 80db (designed using MATLAB ellip(), cascaded SOS, but running in my own C-code). These filters cover the entire Nyquist range in equally-spaced, non-overlapping bands and perform very nicely. My problem is that I would like to combine one or more filters...

Hello, is there any way to get near-realtime audio spectrum info faster than FFT ? For example- VST graphic 7 band EQ,with level meters for each...

Hello, is there any way to get near-realtime audio spectrum info faster than FFT ? For example- VST graphic 7 band EQ,with level meters for each band..

I have tried to use biquad filters but it eats a lot of cpu. The filter bands should be very narrow with +/-5hz precision. any help? and...

I have tried to use biquad filters but it eats a lot of cpu. The filter bands should be very narrow with +/-5hz precision. any help? and also, can anybody post some ideas for fast and easy soft-clipping?

Hi All, I would like to to know to go about rectifying the problem of "Inner matrix dimensions must agree." The problem occur when I try...

Hi All, I would like to to know to go about rectifying the problem of "Inner matrix dimensions must agree." The problem occur when I try to multiply a real time sound signal with the hanning window at 320. Would like to know how to go about to change or modify the problem when using Matlab. Thanks!~

Hi all, I would like to know how to apply the hanning window to an audio signal that is sampled at 16KHz?And how do we set the duration of the...

Hi all, I would like to know how to apply the hanning window to an audio signal that is sampled at 16KHz?And how do we set the duration of the hanning window to be 20ms for the whole signal? Thanks!!

Hello, Also, the values of b0 and b1 (or -b1, for the HPF) depend on the frequency= response that you need to achieve. On a Signals &...

Hello, Also, the values of b0 and b1 (or -b1, for the HPF) depend on the frequency= response that you need to achieve. On a Signals & Systems' book you will f= ind out methods to obtain them, or use MATLAB (or equivalent tool) to obtai= n them. Regards, =20 Jaime Andr=E9s Aranguren Cardona j...@ieee.org j...@computer.org ----- Original Message ---- From: Amit

Hi, I am doing a low-pass and high-pass 7-tap moving average filter filter on Xilinx Virtex II Pro FPGA. I am recording a sound, storing it...

Hi, I am doing a low-pass and high-pass 7-tap moving average filter filter on Xilinx Virtex II Pro FPGA. I am recording a sound, storing it in DDR RAM and filtering it and playing it back. But, the output sounds like noise for both low-pass and high-pass filters. Could someone please let me know what I could be doing wrong? Also, how can I calculate the cut-off frequencies for a seven...

Chris Moore wrote: Please see my article AN-11. It gives a simplified method (Impulse Invariance) of designing with first order filters: >...

Chris Moore wrote: Please see my article AN-11. It gives a simplified method (Impulse Invariance) of designing with first order filters: > http://www.sevenwoodsaudio.com/AN11.pdf Best, Chris Jaime Andres Aranguren Cardona wrote: > Hello, > > Also, the values of b0 and b1 (or -b1, for the HPF) depend on the > frequency response that you need to achieve. On a Signals & Systems'