ITD algorithm equation for sound source localization

Started by Sylvia in comp.dsp13 years ago 4 replies

Hi In the ITD algorithm for sound source localization,there is an equation that needs to be solved for angle of arrival a = d sin(theta) +...

Hi In the ITD algorithm for sound source localization,there is an equation that needs to be solved for angle of arrival a = d sin(theta) + d*theta where d is radius of head and a is additional distance that sound has to cover to reach distant ear(compared to the near ear). My question is,if know 'a' and 'd',how we can implement the above equation to have unique value of theta(angle of ar...


how does one change the timbre of a sound

Started by jeanbaptiste36 in comp.dsp15 years ago 6 replies

I was wondering how one changes the timbre of a sound without changing the pitch. Igor

I was wondering how one changes the timbre of a sound without changing the pitch. Igor


spectral accumulation using phase vocoder

Started by Anonymous in comp.dsp3 years ago 2 replies

Hi! I've made a simple freeze effect with a phase vocoder that can 'freeze' the sound by repeatedly converting the same spectral frame of that...

Hi! I've made a simple freeze effect with a phase vocoder that can 'freeze' the sound by repeatedly converting the same spectral frame of that sound to the time domain, incrementing the phases each time with the phase difference calculated from that frame and the previous frame. This works fine. What I would like to do though, is take another freeze frame out the incoming sound a


speakers for single source sound detection

Started by Sylvia in comp.dsp13 years ago 6 replies

In single source sound localization,which speakers are better to use in reverberant environment?Point source speakers or normal speakers.i...

In single source sound localization,which speakers are better to use in reverberant environment?Point source speakers or normal speakers.i would appreciate any list of standard speakers used in sound localization.


Timestretching with Granular Synthesis: Keeping the pitch constant?

Started by Joe Bloggs in comp.dsp12 years ago 4 replies

I have made a program that timestretches WAV sound files using granular synthesis. It has three variable input values: % stretch factor, grain...

I have made a program that timestretches WAV sound files using granular synthesis. It has three variable input values: % stretch factor, grain length (in samples) and grains per second. All seems to be working fine except for one problem I'm having with the pitch of the stretched sound. If, for example, I stretch a sound to 200% with a grain length of 1000 samples and 41 grains per seco...


Sound Capturing

Started by kiplring in comp.dsp17 years ago 1 reply

Is there anyway to capturing the speaker-out-sound( not from microphone)? Any Library, or Win-API welcomed. Thanks.

Is there anyway to capturing the speaker-out-sound( not from microphone)? Any Library, or Win-API welcomed. Thanks.


Dual Microphone recordings for sound localization lab exercise

Started by Robert Rozman in comp.dsp12 years ago 5 replies

Hi, I'd kindly for pointers if such binaural recordings are available to public somewhere ? I'd love to have recordings from two...

Hi, I'd kindly for pointers if such binaural recordings are available to public somewhere ? I'd love to have recordings from two microphones (spaced from each other) and several different positions of sound source... I'd use it for lab exercises... Thanks in advance, regards, Rob.


something went wrong

Started by khurram712 in comp.dsp12 years ago

hi everyone.. i tried spectral subtraction method for removal of noise from speech signal but the sound that i hear now seems more like sound...

hi everyone.. i tried spectral subtraction method for removal of noise from speech signal but the sound that i hear now seems more like sound of "Alvin and Chipmunks".can someone diagnose the problem this is my code for the program.i hope you could help out clc %function [ss]=specsub1(si,fs) [s,fs]=wavread('west.wav'); %specsub(noi,fr); ti=16e-3;% desired frame increment (16 ms) o...


Sample Rate conversion

Started by HardySpicer in comp.dsp8 years ago 30 replies

My A/D samples at 33.33kHz and after processing I am using the sound card to output the audio result. The sound card only accepts...

My A/D samples at 33.33kHz and after processing I am using the sound card to output the audio result. The sound card only accepts 44.1khz, 22.05kHz etc. I am thinking of going 33.1 to 20.05. this doesn't have to be spot on, just approx since it is only for listening to. What is the best way - say 333/220 and use euclids algorithm?


OT Clock Speeds

Started by Anonymous in comp.dsp17 years ago 2 replies

Maybe someone here is knowledgable re clock usage in a Windose environment... 1- If I capture my (high-end) sound card ADC output to disk,...

Maybe someone here is knowledgable re clock usage in a Windose environment... 1- If I capture my (high-end) sound card ADC output to disk, is it paced by the card's clock or the computer's clock? In other words, the 44100 samples/second is who's "second"? 2- If I play a wav file is it paced by the sound card's clock or the computer's clock? 3- If I play a CDDA disk (with no sound ...


reading sound samples from text for lpc-10 in c language ???

Started by emperor84 in comp.dsp14 years ago 1 reply

my lpc-10 c program works on an ".au" sound file but my aim is to run the program from a txt file where the sound samples are settled in...

my lpc-10 c program works on an ".au" sound file but my aim is to run the program from a txt file where the sound samples are settled in the begining part of lines like that : 14183 12593 29486 25712 -9985 12288 16384 -5889 -12033 -22273 -30465 -9985 18432 12288 12288 20480 8192 4096 4096 -12033 ............... but i can't solve this problem, the .au file is opened that with ...


Behind-the-scenes software equalizer

Started by Anonymous in comp.dsp15 years ago 18 replies

Hello, I recently pulled a pair of old stereo speakers out of my basement to use with my PC. The only markings that might identify them I can...

Hello, I recently pulled a pair of old stereo speakers out of my basement to use with my PC. The only markings that might identify them I can find on them are "MLI 691H Hi-Fi Sound Monitor" Anyways, they don't really have that great of a sound quality, and I was wondering if there are any software equalizers out there that would allow me to tweak various frequencies. I was hoping this ...


Dumb question from a newbie...

Started by John Oyler in comp.dsp13 years ago 2 replies

I've been tossing the idea around of trying to design a device to generate music and sound effects electronically. In particular, I want to be...

I've been tossing the idea around of trying to design a device to generate music and sound effects electronically. In particular, I want to be able to unplug the sound chip from an old home computer, and plug in a small board that would upgrade it from the beeps and buzzes that was all it was capable of, into something a bit snazzier. Now, I've got a good handle on most of it, I've got an f...


Calculating sound amplitude in dB from a WAV file

Started by Anonymous in comp.dsp15 years ago 16 replies

Hi everyone, I suppose this is an easy question for some, but I'm trying to calculate in dB, the amplitude of a recorded WAV sound file. Is...

Hi everyone, I suppose this is an easy question for some, but I'm trying to calculate in dB, the amplitude of a recorded WAV sound file. Is this possible? How can it be done? Are there any libraries/APIs that do this? Thanks, - Olumide


Use DSP features of sound card

Started by fakufaku in comp.dsp12 years ago 2 replies

Hi, I was wondering if it is possible to access and use the dsp features of sound cards in order to get some hardware acceleration for...

Hi, I was wondering if it is possible to access and use the dsp features of sound cards in order to get some hardware acceleration for computationaly intensive algorithms like fft or filtering. I was also thinking of graphic cards. Did anyone try this before ? Thanks a lot. Robin


IVR

Started by anuradha_12 in comp.dsp11 years ago 11 replies

hi guys Im new to DSP and stuff.I'm trying to create an IVR for my final year project. I'm planing to use Goertzel algorithm for this.This is a...

hi guys Im new to DSP and stuff.I'm trying to create an IVR for my final year project. I'm planing to use Goertzel algorithm for this.This is a non TAPI windows based IVR which uses live sound stream to detect DTMF tones. Following APIs will allow me to access to the sound card WaveInOpen() waveInPrepareHeader() waveInAddBuffer() waveInStart() waveInUnprepareHeader() waveInClose() D...


voice changer

Started by python in comp.dsp14 years ago 6 replies

is it possible to change someone's voice to sound like a particular person's voice? e.g make my voice sound like George Bush(accent, pitch, length...

is it possible to change someone's voice to sound like a particular person's voice? e.g make my voice sound like George Bush(accent, pitch, length of words and all). i'm thinking of doing this as a university project but i am completely new to this field of dsp. are there algorithms for doing this? please advice me.


fundamental - simple question - on audio samples

Started by Srini in comp.dsp14 years ago 5 replies

what exactly is the physical significance of the samples in a recorded sound stream like say in a wav file. i realize it is a result of...

what exactly is the physical significance of the samples in a recorded sound stream like say in a wav file. i realize it is a result of AD conversion of a signal but beyond that i am drawing a blank. is this the amplitude of the sound wave? can someone point me to any reference. my own searches have led to lot of conceptual understanding but not specific. thanks srini


phase corrector / filter with arbitrary phase and amplitude / designer tools

Started by in comp.dsp11 years ago 18 replies

Hi I am creating the active bass trap. It is device to correct room acoustic similar to...

Hi I am creating the active bass trap. It is device to correct room acoustic similar to this: http://www.bagend.com/bagend/downloads/ETrap.pdf http://www.bagend.com/bagend/ETrap.htm In few words - It receives sound from room via microphone, filters by pass band filter (tuned on resonance frequency of room) and transmitting via speaker sound in opposite phase. This sound wave is opposite t...


sound visualization (fft, spectrum, phase, panning, etc)

Started by dial...@gmail.com in comp.dsp15 years ago 2 replies

Hi friends, This is my first post to the list. I have been developing some sound visualization software in linux using c/c++ & openGL for quite...

Hi friends, This is my first post to the list. I have been developing some sound visualization software in linux using c/c++ & openGL for quite some time now, and I am having difficulty. So i've been trudging through the net looking for examples to do the different items i have mentioned below. I have made some good progress, and found a few examples for FFT's and such. And having my own pr...