Apologies for multiple posting. Dear colleague, Registration for the DAFx-06 conference is open. Detailed information and registration form...

Apologies for multiple posting. Dear colleague, Registration for the DAFx-06 conference is open. Detailed information and registration form can be found at: http://www.dafx.ca/dafx06_registration.html Note that the deadline for early registration is August 7, 2006. The three-day program will be available soon. There will be 64 papers presented (no parallel sessions): - 39 or...

Dear all, I am testing an inband DTMF detection algorithm at the moment.Since I am new in the area, I'd appreciate some clarification on some...

Dear all, I am testing an inband DTMF detection algorithm at the moment.Since I am new in the area, I'd appreciate some clarification on some of the specifications in ITU-T recommendation Q.24. 1. Maximum Non-operation signal duration: I understand that the recommendation requires the detector not to respond to any signal whose duration is shorter than this number. However, what should...

Hi, I am using FFTW to obtain freq values from an audio signal.The input and output arrays are the same size. However, I need more precision (i...

Hi, I am using FFTW to obtain freq values from an audio signal.The input and output arrays are the same size. However, I need more precision (i need to detect frequencies which aren't whole numbers) therefore as i understand it i need a larger output array than my input one. i am using the real-to-real fftw_plan_r2r_1d function as follows: fftw_plan_r2r_1d (numSamples, in, out, FFTW_R2HC, ...

I have a question regarding filtering and amplifying certain frequencies in an audio file. I have a file that has regular audio in the audible...

I have a question regarding filtering and amplifying certain frequencies in an audio file. I have a file that has regular audio in the audible range, and a superaudible tone mixed in that operates at 19KHz. I need to extract the 19KHz signal from the regular audio, amplify it to around 3.5 V, and convert it to a square wave (high when signal is present, low when it is not). I am currently ...

Dear List: I was reading the book, "Speech Coding and Synthesis" by Kleijn & Paliwal. In page 445, from the table we see that for scalar...

Dear List: I was reading the book, "Speech Coding and Synthesis" by Kleijn & Paliwal. In page 445, from the table we see that for scalar quantization of the reflection coefficients, it costs 4 bits per parameter for 1 dB distortion. Similarly, in page 447, from the table we see that for LAR scalar quantization, it costs 3.2 bits per parameter for 1 dB distortion. The above numbers are for ...

Dear List: I want to do some fast experiments with the MPEG Psychoacoustic Model - 2. Is there any source from where I can download the...

Dear List: I want to do some fast experiments with the MPEG Psychoacoustic Model - 2. Is there any source from where I can download the routines? I just need a simple program, where I give the input audio signal, and I get the masking threshold out. Any help in this regard will be highly appreciated, and also acknowledged. Best Regards, ~Arijit

I'm working on a project to interface to an EEG device for reading brain-waves. I'm completely new to this type of thing but I've learned...

I'm working on a project to interface to an EEG device for reading brain-waves. I'm completely new to this type of thing but I've learned alot since I started a couple weeks ago. I first started doing a bandpass by calculating an FFT and zeroing out the signal then doing the IFFT...lol it sorta worked for a while. Anyway now I'm using a windowed sinc routine to calculate FIR coefficie...

Hi all, I'm trying to estimate the impulse response of the room by playing a 4 sec white noise speech file from one end of the room and...

Hi all, I'm trying to estimate the impulse response of the room by playing a 4 sec white noise speech file from one end of the room and recording it at the other end. I'm told to estimate the impulse response of the room by using Recursive Least Squares method. Please help how to get this done, any help in this regard will be greatly appreciated. Thanks, Abhishek Abhishek A...

Hello, I am currently using a digital 1000 point lowpass filter to extract the envelope from an audio signal of which I have removed the...

Hello, I am currently using a digital 1000 point lowpass filter to extract the envelope from an audio signal of which I have removed the negative components (by using the absolute value of the signal). I am using this resulting envelope to detect whether notes have occurred or not. I would like to investigate better filter's responses to extract this envelope from an audio signal. ...

Nasim- > i followed ur discussion abt two q15 number multiplication and addition, > i have one doubt on two q15 format...

Nasim- > i followed ur discussion abt two q15 number multiplication and addition, > i have one doubt on two q15 format multiplication. as u told that in multiplication > of two q15 format we hav two sign bit : > So we are left with, > > > > 0 I's 2 S's 14 F's > > > > So now you left shift it by 1 to get rid of sign, > > > > 0 I's 1 S 15 F (lsb fraction bit being 0) > i am agr