Is there a software FSK encoder API for PC using no addtl hardware?

Started by Tomer in comp.dsp16 years ago 18 replies

Hi All, We need an API module to allow us to send data using the FSK (Frequency Shift Keying) modulation. This module is to run on a PC...

Hi All, We need an API module to allow us to send data using the FSK (Frequency Shift Keying) modulation. This module is to run on a PC and may use no additional hardware except for the built in sound card. The module will allow us to convert ASCII characters to their FSK sound and play that on the PC sound card. The inputs to the module: a string of characters. The output of the mod...


spectral accumulation using phase vocoder

Started by Anonymous in comp.dsp2 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


Using Sound card Creative Audigy 4 pro for multiple microphone input

Started by salrome in comp.dsp12 years ago 6 replies

Hi My goal is to take sound data from multiple microphones into MATLAB workspace. I have got the sound card Creative Audigy 4 pro for...

Hi My goal is to take sound data from multiple microphones into MATLAB workspace. I have got the sound card Creative Audigy 4 pro for multiple microphone input.I would be glad if some one has info regarding 1.How to connect multiple microphones to this card,which terminals/ports will be used. 2.How can I read the audio data from the microphones through the sound card in MATLAB(for ex...


speakers for single source sound detection

Started by Sylvia in comp.dsp12 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.dsp11 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...


OT Clock Speeds

Started by Anonymous in comp.dsp16 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 ...


Sample Rate conversion

Started by HardySpicer in comp.dsp7 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?


Dumb question from a newbie...

Started by John Oyler in comp.dsp12 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...


Behind-the-scenes software equalizer

Started by Anonymous in comp.dsp14 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 ...


ITD algorithm equation for sound source localization

Started by Sylvia in comp.dsp12 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...


voice changer

Started by python in comp.dsp13 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.


something went wrong

Started by khurram712 in comp.dsp11 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...


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

Started by emperor84 in comp.dsp13 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 ...


Dual Microphone recordings for sound localization lab exercise

Started by Robert Rozman in comp.dsp11 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.


Calibrating speakers

Started by Michel Rouzic in comp.dsp14 years ago 20 replies

I just got a new set of 2.1 speaker for my PC for christmas, my main problem with it is that it outputs a very altered sound, mostly the bass,...

I just got a new set of 2.1 speaker for my PC for christmas, my main problem with it is that it outputs a very altered sound, mostly the bass, they are way too loud. I firstly thought it would be easy fixing that with some software equalizer but no, the sound is still very weird. I thought about playing a delta function through my speakers and record it with a microphone to look at the fre...


Calculating sound amplitude in dB from a WAV file

Started by Anonymous in comp.dsp14 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


Compressor-limiter detection

Started by Rob Vermeulen in comp.dsp15 years ago 6 replies

Folks, I want to be able to detect commercial breaks :) So it seems that (in the Netherlands at least) on TV, the sound of commercials is...

Folks, I want to be able to detect commercial breaks :) So it seems that (in the Netherlands at least) on TV, the sound of commercials is broadcasted through some massive compressor-limiter while "regular" broadcasts such as movies and newsbreaks have "normal" sound. I think someone has already come up with this so I'd like to know how to be able to detect this. I've been thinking/hypot...


Use DSP features of sound card

Started by fakufaku in comp.dsp11 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


fundamental - simple question - on audio samples

Started by Srini in comp.dsp13 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


How to test sound freq?

Started by Burgos in comp.dsp11 years ago 8 replies

First of all: Hello, everybody! I have a problem that I can't figure myself, so I'm kindly asking you for help. My application consists of two...

First of all: Hello, everybody! I have a problem that I can't figure myself, so I'm kindly asking you for help. My application consists of two parts: the first one, which is generating clean sin tone for a given frequency, and the second one, which is used to detect signal freq. I have used FFT to find frequency of input sound, but it seems to expensive for me, because all I need is to test...