Sound Source Localisation...HELP!!!!

Started by FordPrefect in comp.dsp13 years ago 12 replies

Hi.. I needed a few tips on 2-D localisation of a sound source using 2 microphones (COLLINEAR WITH THE SOURCE). The source is assumed to be...

Hi.. I needed a few tips on 2-D localisation of a sound source using 2 microphones (COLLINEAR WITH THE SOURCE). The source is assumed to be a single frequency source and in-accessible with the only known characteristics being the ones measured from an oscilloscope. Oh, and I'm a Mechanical Engineer with ABSOLUTELY NO PRIOR EXPOSURE TO DSP so please be kind enough to elaborate those "few tips"...T...


soundcard buffer format

Started by Jonas Rundberg in comp.dsp17 years ago 37 replies

Hi I am developing on an app that records sound from the soundcard and then perform some dsp on the accuired sound. So far I've been...

Hi I am developing on an app that records sound from the soundcard and then perform some dsp on the accuired sound. So far I've been working with 44100 Hz sampling frequency, but need to change to 22050 Hz. I use the win32 sdk multimedia functions to communicate with the soundcard. When I use 44100 Hz I know that the recorded buffer I receive from the soundcard contains values that shou...


ossaudiodev full duplex

Started by Anonymous in comp.dsp17 years ago 3 replies

I'm messing around with trying to record sound reflections off of various objects at different frequencies. I'm using python 2.4 on Fedora Core...

I'm messing around with trying to record sound reflections off of various objects at different frequencies. I'm using python 2.4 on Fedora Core 3 and I have an SBLive card. Basically, I want to send some samples through a speaker and then record from an input source (the line input on the sound card which I have the output of a mixer going into and a mic plugged into the mixer). So I want to...


Sound localisation

Started by And Blah Blah Blah in comp.dsp16 years ago 10 replies

I am doing research into sound localisation for a degree project and how to alter signals to affect their apparent position. I understand how...

I am doing research into sound localisation for a degree project and how to alter signals to affect their apparent position. I understand how and why time delay and amplitude difference between the ears have an effect as well as spectral changes. However I keep finding references to phase differences being an important cue and I don't understand what a phase difference would be in this...


DTMF detection and samples

Started by dkurman in comp.dsp16 years ago 2 replies

Hi. I?m a newbie in this but I need to acomplish a job with this stuff. I?m capturing sound through the sound blaster and then need to detect if...

Hi. I?m a newbie in this but I need to acomplish a job with this stuff. I?m capturing sound through the sound blaster and then need to detect if there were some DTMF. For this I?m using Goertzel. What I can?t understand is about buffering and samples, I mean: Do I have to run the goertzel for N=106 on each sample and sum all the results and then try to detect if there were DTMF or this is processi...


Get up and going with audio sound processing DSP development

Started by Aaron Gray in comp.dsp15 years ago 4 replies

Hi, Whats the cheepest way to get up and going with audio sound processing DSP development. I would like a system that I can connect to a...

Hi, Whats the cheepest way to get up and going with audio sound processing DSP development. I would like a system that I can connect to a computer with USB to program it, and have audio in and out connections, I am a programmer and am familuar with digital and analogue electronics. Many thanks in advance, Aaron


sound elevation detection algorithms

Started by Sylvia in comp.dsp14 years ago 25 replies

i require links about sound elevation detection algorithms using only two microphones.i will appreciate any references,papers,algorithms...

i require links about sound elevation detection algorithms using only two microphones.i will appreciate any references,papers,algorithms etc. thanks Sylvia


Re: digital Sound IF filter must be linear phase?

Started by Anonymous in comp.dsp14 years ago

On May 22, 8:51 pm, cincy...@gmail.com wrote: > On May 22, 5:02 pm, "dtsao" wrote: > > > Hi, > > > I need to build a digital filter...

On May 22, 8:51 pm, cincy...@gmail.com wrote: > On May 22, 5:02 pm, "dtsao" wrote: > > > Hi, > > > I need to build a digital filter for a sound signal. The SIF filter must > > remove the video portion of the channel and be compatible with worldwide > > standards and have good rejection. > > From Matlab, it looks like this requires a tremendous amount of > > coefficient


Re: digital Sound IF filter must be linear phase?

Started by Anonymous in comp.dsp14 years ago 1 reply

On May 22, 8:51 pm, cincy...@gmail.com wrote: > On May 22, 5:02 pm, "dtsao" wrote: > > > Hi, > > > I need to build a digital filter...

On May 22, 8:51 pm, cincy...@gmail.com wrote: > On May 22, 5:02 pm, "dtsao" wrote: > > > Hi, > > > I need to build a digital filter for a sound signal. The SIF filter must > > remove the video portion of the channel and be compatible with worldwide > > standards and have good rejection. > > From Matlab, it looks like this requires a tremendous amount of > > coefficient


JAVA & FFT + Sound Processing & Latency in JAVA

Started by QWERTY in comp.dsp16 years ago 3 replies

Hello all! I'm new on this group. I started following it because I need help for project I'm working on at my faculty. I need to include...

Hello all! I'm new on this group. I started following it because I need help for project I'm working on at my faculty. I need to include an FFT algorithm in my project in order to manage sound signal. Can anyone recommend me a good implementation of FFT in JAVA, because, as I have read about it, there is many of implementations which are not optimised. Code examples about mnaging sou...


Converting from stereo to multichannel format.

Started by Vij Kal VijayaRaghavan Kalyanapasupathy in comp.dsp18 years ago 2 replies

Hello, Can someone point out how to convert from stereo (in PCM 2 channel 16 bit 44.1 Khz) format to 5.1/6.1/7.1 channel sound formats. I...

Hello, Can someone point out how to convert from stereo (in PCM 2 channel 16 bit 44.1 Khz) format to 5.1/6.1/7.1 channel sound formats. I have songs that on CD's. I tried using the Dolby Pro-Logic II setting on my receiver/amplifier to see if surround sound effect was available. It didn't turn out too well. I wanted to see if its possible to try and write a program to convert the CD audi...


Simple Sample Repetition?

Started by Cranisch in comp.dsp8 years ago 6 replies

Ok guys, i am a mechanical engineer...so please forgive me if i get messed up with some DSP terms ;-) I have the following issue: From a test...

Ok guys, i am a mechanical engineer...so please forgive me if i get messed up with some DSP terms ;-) I have the following issue: From a test bench measurement in a speed runup I have obtained an air-borne sound signal with a microphone (sampling rate 48kHz). I need to extract a stationary sound signal for one specific speed out of the "continuous" speed runup. Therefore, i have tried the f...


HELP with Amp Modeling

Started by Wayne Preis in comp.dsp17 years ago 2 replies

I'm hoping that someone here can help me. I'm looking for a library of sounds for a guitar processor. I'm trying to build a battery...

I'm hoping that someone here can help me. I'm looking for a library of sounds for a guitar processor. I'm trying to build a battery powered processor using Analog Device's Blackfin chip. Does anybody have any suggestions on where to begin looking? Thanks so much, WP


FFT

Started by maz_p5 in comp.dsp13 years ago 3 replies

Hi, I am working on calculating the delay between the 2 sound sources arrived on the microphones. The sound sources are not exactly the same...

Hi, I am working on calculating the delay between the 2 sound sources arrived on the microphones. The sound sources are not exactly the same due to the various factors. I have tried cross - correlation but I have unsuccessful. I have heard about frequency cross-correlation using FFT for determining the delay. How do I do that in MATLAB? Can someone please explain with the help of an example cod...


Speech proceesing of Stereo Vs Mono sound?

Started by Anonymous in comp.dsp7 years ago 5 replies

When speech files are given and turned out that some are stereo, am I correct to assume that the two channels are identical ... hence it is...

When speech files are given and turned out that some are stereo, am I correct to assume that the two channels are identical ... hence it is sufficient to just to process one channel and extract features? So in the code, i just use an index and read all the indices that are odd and use the correct byte size for memory allocation? I used Audacity to record a 5sec sound - it turned out to have ...


sound source localization using electret microphone array

Started by matar770 in comp.dsp7 years ago 10 replies

i want to use an electret microphone array to locate a sound sorce (actually its the direction of arrival) , im using Steered Response Power -...

i want to use an electret microphone array to locate a sound sorce (actually its the direction of arrival) , im using Steered Response Power - Phase Transform (SRP-PHAT) algorithm, and i'm confused about a few things: 1.my array consists of five microphones but i dont know which configuration is easier to use to obtain the DOA in 3D, the tetrahydron or the square pyramid ? 2.its my first h...


Filter to remove noise from a corrupted song

Started by ~farah_r~ in comp.dsp14 years ago 7 replies

Hi All, I need a little help in analyzing this question. I am a student taking Intro to DSP class and we have this lab in which we're given a...

Hi All, I need a little help in analyzing this question. I am a student taking Intro to DSP class and we have this lab in which we're given a corrupted sound file from which we have to filter out the noise from it. So, what I did was I plotted out the FFT of the sound file to figure out where the interference was, and found that the interference is at freq. of + and - 44.04 kHz with magnitu...


Human hearing instataneous dynamic rage?

Started by Richard Owlett in comp.dsp13 years ago 24 replies

Subject line probably poorly stated. When dynamic range of human ear is discussed it's usually comparing threshold of pain to weakest...

Subject line probably poorly stated. When dynamic range of human ear is discussed it's usually comparing threshold of pain to weakest detectable sound. I'm more interested in comparing a loud and soft sound being distinguished at the same time. Perhaps it might have also come from test to determine when amplifier distortion becomes detectable. Suggested search terms?


comb filters and fourier transforms for splitting sound into frequencies

Started by ben in comp.dsp16 years ago 26 replies

hello, here's the goal: split sound, interval by interval, into frequency "bins". here's two alternative possible methods i know of to...

hello, here's the goal: split sound, interval by interval, into frequency "bins". here's two alternative possible methods i know of to achieve that: 1. comb filters: look for series of spikes seperated by equally sized gaps. 2. fourier transforms: i'm sure you know much more about those than me. how do you think the two method's results will compare? do (or can) they amount to th...


Audio channel timing

Started by Grant in comp.dsp15 years ago 6 replies

Can someone point me in the right direction please? I need to calculate a unique time "marker" in a hi-fi (20Hz - 20 kHz) audio path during a...

Can someone point me in the right direction please? I need to calculate a unique time "marker" in a hi-fi (20Hz - 20 kHz) audio path during a sample period of a few seconds or so. It doesn't matter what sound event(s) go into the calculation or where the marker is placed with respect to any particular sound events in the audio path, so long as the marker is unique and repeatable. If n...