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Discussion Groups | Audio Signal Processing


Technical discussions related to Audio Signal Processing (digital effects, acoustics, noise reduction, musical signal processing, etc).

  

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Audio pcm width conversion: 16 to 24 bit
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"Ume...@gmail.com [audiodsp]" - Oct 7 2014
I want to convert 16 bit audio pcm samples to 24 bit pcm samples so that when I process them in SRC or any other post processing algorithms it gives better SNR. My question is: ... Audio pcm width conversion: 16 to 24 bit

DSP Resources for Musicians - Suggestions Required
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"roc...@btinternet.com [audiodsp]" - Sep 23 2014
Hi all, I'm looking for some solid resources on digital signal processing for musicians with a basic math background, specifically dealing with the following concepts: 0) Ma... DSP Resources for Musicians - Suggestions Required

z-plane representation of single order all-pass filter
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"fau...@yahoo.com [audiodsp]" - Sep 3 2014
I'm used to the s-plane representation of an all-pass filter, where the pole(s) are on the left-side of the y-axis, and the zeroes are mirror imaged across the y-axis. It's fairly... z-plane representation of single order all-pass filter

ref: pg 174, Oppenheim, Schafer and Buck, Discrete-Time Signal Processing, second ed.
-2

"Kri...@yahoo.co.in [audiodsp]" - May 14 2014
Subject: δ(ω/ T) = T.δ(ω) I'm sorry that this email turned out long. My apologies that I couldn't f... ref: pg 174, Oppenheim, Schafer and Buck, Discrete-Time Signal Processing, second ed.

SPL computation, Prms computation
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wmta...@dcs.upd.edu.ph - Apr 9 2014
Hi I have a microphone(with preamp) connected to a microcontroller - I am able to get samples now, which could be converted to pressure since I have the sensitivity of the micropho... SPL computation, Prms computation

Spectral envelope normalization (speech processing)
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kout...@hotmail.com - Feb 13 2014
Dear forum, I created a synthetic speech signal x[n] of 256 samples assuming that the source signal u[n] is an impulse train (not a glottal flow signal) and also I assumed that ... Spectral envelope normalization (speech processing)

Sampling Rate Conversion   [2 Articles]
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prat...@gmail.com - Dec 26 2013
Hi experts, I want to convert audio sampled at 192KHz to 44.1kHz. It involves interpolation by 147 and decimation by 640. I'm designing filter coefficients us... Sampling Rate Conversion

Request for Guidance on Optimal DSP for Audio Application
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rich...@btinternet.com - Oct 9 2013
Hi, I am designing an audio DAC and my intention is to use a DSP to interface with the CD player to receive the optical or coaxial signal. I have been examining the Elektor seri... Request for Guidance on Optimal DSP for Audio Application

Strumming problem.   [3 Articles]
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exactlybest - Sep 26 2013
Hi. I have a very accurate frequency detector that works pretty much perfectly with my guitar. I want to implement certain features that involve note validity (user plays a note... Strumming problem.

fftw problem   [2 Articles]
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bauzane - Sep 26 2013
I'm an Italian university student. I'm trying to use FFTW libraries. I have problems but I don't know in which forum I may ask help so I try here. I wrote the following C program ... fftw problem
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