Technical discussions related to Audio Signal Processing (digital effects, acoustics, noise reduction, musical signal processing, etc).
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jsnuff1 - Sep 30 2008
I have a mic that outputs its level in decibels (the range this mic outputs seems to be
between around -55db to 0db) and would like to find an algorithm that converts this value ... 
George Bean - Sep 25 2008
I have an application to decode an FSK signal on a VHF carrier. In the past
I would have used an EXAR XR2211A FSK demodulator chip feeding a
microcontroller but I assume with the... 
sauw...@gmail.com - Sep 23 2008
Hi all,
This is more of a theoretical question, since I have much to learn about acoustics. How does a spectrum behave when a target (ex. car, plane) is evaluated at different ... 
sumandari - - Sep 14 2008
Hi, im newbie
sorry for my bad english
how to get octave band spectrum from fft spectrum without using
filter. Can i just average the freq in each bandwidth? Thankyou
--
Re... 
"d.abc51" - Sep 13 2008
Hi all, I'm trying to code a program that can mix together multiple
PCM wav files into a single output. It is essentially a sequencer, and
works fine at points in the sequence wh... 
kaus...@yahoo.com - Sep 13 2008
Hello everybody,
-I often get confused between an audio codec and an audio file format. Doesnt a .mp3 extension necessarily mean that the file has been encoded using 'mp3' encodin... 
daro...@yahoo.fr - Sep 2 2008
Hi !
I have little experience with time filtering ,ad i'd very much appreciate some input. So here's the deal:
I'm trying to enhance audio with some non-linear processing
so bas... 
tung...@yahoo.com - Sep 1 2008
Hi all,
I am working on an Noise Adaptive Playback project to eliminate the noise from a microphone of the USB HeadSet.
My project is working ok with floating point number, now i... 
sauw...@gmail.com - Aug 26 2008
I've got a binary file in 24 bits that I would like to convert to 16 bits and 12..
Does anyone know if there is a simple way of doing it in matlab besides using fread(fid,inf,... 
sneshs - Aug 22 2008
Hello there,
I'm new in digital signal processing.
I'm going to create a linear phase filter with filter lenght N = 512 for audio signal
processing in Matlab. But I'm kind... 
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