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Microphonr beamforming

Started by lali...@gmail.com in Audio Signal Processing14 years ago

Hi all, I am working on Multichannel wiener filter and spatial prediction in application to binaural hearing aids. I have both the filter...

Hi all, I am working on Multichannel wiener filter and spatial prediction in application to binaural hearing aids. I have both the filter coefficients which will supress the noise or interfering source at any angle ( 20 deg to 340 deg, target source is assumed at 0 deg and in the range of 20 deg on both sides of target source there is no interfer). My problem is to evaluate both the algorithms ...


STFT window size causes AM output from sine wave input.

Started by camb...@yahoo.co.uk in Audio Signal Processing14 years ago

Hi all, I'm writing a time-varying equalisation algorithm using an overlap-add STFT with a Hann window at 50% overlap. It will be taking...

Hi all, I'm writing a time-varying equalisation algorithm using an overlap-add STFT with a Hann window at 50% overlap. It will be taking 'real-world' audio signals, however before I began EQ testing, I tested everything straight-through ('flat' zero EQ). I tried various basic signals and by pure chance the first one I tried was a sine wave that was periodic with the window size. Due the 50...


need help - Wiener filter

Started by hend...@web.de in Audio Signal Processing14 years ago 2 replies

Hi, I'm currently working on my bachelors thesis. I examine the noise reduction performance of the wiener filter in various transform...

Hi, I'm currently working on my bachelors thesis. I examine the noise reduction performance of the wiener filter in various transform domains. I began with the FFT and the KLT (Karhunen?Loève Transform) domain. For each block I transform the block with the fft/klt and then compute the filter coefficients, filter the signal, inverse transform and then write it out. My problem is now that


Pitch Detection - Goto 'PreFest', Yeh and Roebel, and Klapuri Implementations

Started by b.ev...@hud.ac.uk in Audio Signal Processing14 years ago

Hi All, I'm a long time reader, first time poster! As part of my MSc by research I wish to test 3 pitch detection algorithms alongside my...

Hi All, I'm a long time reader, first time poster! As part of my MSc by research I wish to test 3 pitch detection algorithms alongside my own. The 3 algorithms are: M.Goto's PreFest method as detailed here - http://citeseerx.ist.psu.edu/viewdoc/download?doi=10.1.1.58.1309&rep=rep1&type=pdf Yeh and Roebel's method as detailed here - http://articles.ircam.fr/textes/Yeh09b/ and Klapuri...


Testing existing state-of-the-art VAD algorithms

Started by kash...@gmail.com in Audio Signal Processing14 years ago 1 reply

Hi all Are there implementations of the standard VAD methods which can be used to evaluate their performance when given a wav file in the...

Hi all Are there implementations of the standard VAD methods which can be used to evaluate their performance when given a wav file in the presence of noise? Does Matlab, Labview or any other give us the ability to test ? This is for my Master's thesis and any insight would be helpful ? Once, i have evaluated the existing VADs performance on the wav files i can work towards devising a ...


Hydrophone - turbulence measurements

Started by e.ta...@uq.edu.au in Audio Signal Processing14 years ago 1 reply

Hi all, I have some wave (.wav) files from measurements using a hydrophone in an agitated tank. From a FFT I can see peaks at the impeller...

Hi all, I have some wave (.wav) files from measurements using a hydrophone in an agitated tank. From a FFT I can see peaks at the impeller frequency (1000rpm ~ 17 Hz) and its harmonics. However, I would like to separate the "audio pressure fluctuation" from the "fluid turbulent pressure fluctuation". Is it possible? Is there any kind of filter I should use? Extra information: I am proces...


Comressive Sampling

Started by Mona in Audio Signal Processing14 years ago 1 reply

Hi everyone, So I actually do not have a special problem with something; I just wanted to have a productive chat with someone that shares my...

Hi everyone, So I actually do not have a special problem with something; I just wanted to have a productive chat with someone that shares my interest in Compressed Sensing! I am actually working on my thesis on compressive sensing of speech. So far everything is working nice practically; but I'm having difficulties proving the theory to myself. I know it's a new topic, but that actually shou...


HRIR interpolation

Started by maartendeprez in Audio Signal Processing14 years ago 2 replies

Hi everyone, I'm quite new to dsp, programming a tool to optimize surround music for listening on headphones, stereo speaker systems, etcetera....

Hi everyone, I'm quite new to dsp, programming a tool to optimize surround music for listening on headphones, stereo speaker systems, etcetera. (Such tools probably already exist, but i couldn't find any free software that does what i want.) The best HRIR measurements i could find are those of the Listen project, and with the help of those, i reached some, to my mind, astonishing results that enc...


temporal masking

Started by iulia in Audio Signal Processing14 years ago 2 replies

Hello, I want to implement in matlab the calculation of the masking threshold for an audio file. I read a lot about the frequential masking but...

Hello, I want to implement in matlab the calculation of the masking threshold for an audio file. I read a lot about the frequential masking but I didn't find a lot of bibliography about temporal masking. It seems to be somethings simple but I realy don't understand how it works and how I integrate this in the global masking threshold. Could you please help me with this... anything will be be...


Simple dsp

Started by franz_as_tux in Audio Signal Processing14 years ago 2 replies

Hi @ all! I've a question: I'm working on a project for thesis, involved in automatic pitch shifting for voice/ guitar ( yes it's very...

Hi @ all! I've a question: I'm working on a project for thesis, involved in automatic pitch shifting for voice/ guitar ( yes it's very different..i have to decide valutating how many time I can spend in this work, but this is not the question :) ) In my university we have dsk6711 or dsk5402/5510 c30 and so on but I can't bring @ home for the time i had to work with. Buying one from spectrum digit...


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