## Forums audiodsp

## z-plane representation of single order all-pass filter

inI'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...

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 clear that as omega increases (increasing frequency) in s = j * omega, that the length of the vectors that go from the y-axis to the mirror imaged poles and zeroes will be the same length (per pole-zero pair). Howeve...

## Filtering out dither noise from 63Hz sine wave

inI have 24 bit ADC data with dithering noise on it. My original signals into the ADC were quite small. The dithering noise therefore makes out a...

I have 24 bit ADC data with dithering noise on it. My original signals into the ADC were quite small. The dithering noise therefore makes out a large part of the amplitude of the ADC data. I want to remove the noise to recover as much as possible of the actual signals. I find lots of websites that say that the dithering noise can be filtered out but no description of how this can be done. Unfortu...

## Audio pcm width conversion: 16 to 24 bit

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...

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: Is there any other way to do 16 bit to 24 bit pcm width conversion apart from left shift by 8 (multiply by 256). Thanks

## DSP Resources for Musicians - Suggestions Required

Hi all, I'm looking for some solid resources on digital signal processing for musicians with a basic math background, specifically dealing with...

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) Mathematical symbols commonly found in DSP theory 1) An explanation of positive and negative values in the time domain / Complex numbers 2) Fourier series 3) Z-transform 4) Basic filter design Thanks very much for ...

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

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Subj= ect: =CE=B4(=CF=89/ T) =3D T.=CE=B4(=CF=89)=20=20=20=20=20 I'm sorry that this email turned out long. My apologies that I couldn't fig= ure out my exact question. So only managed to disperse all the information = I could that hopefully revolves around it.=20=20=20=20=20=20 To start with, I wish to understand the dif...

## SPL computation, Prms computation

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...

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 microphone. I want to get the sound pressure level. According to "Engineering Acoustics" and "Engineering Noise Control"(Bies and Hansen), which was referenced by Wikipedia as well http://en.wikipedia.org/wiki/Sound_pressure...

## Spectral envelope normalization (speech processing)

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...

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 there are no lips. The vocal tract filter h[n] is an all pole filter of order 10. So, in frequency domain we have X(w)=H(W)U(W). I computed with the modified periodogram the PSD of X(w) and with freqz the PSD of H(w) ...

## Sampling Rate Conversion

inHi experts, I want to convert audio sampled at 192KHz to 44.1kHz. It involves interpolation by 147 and decimation by 640....

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 using fda tool, with FIR equiripple method. The parameters I gave are fs - 192000*147 fpass - 16000 fstop - 22050 Apass = 0.4 Astop = -80dB But the fda tool gets hanged after some time. Is there any alternative...

## Request for Guidance on Optimal DSP for Audio Application

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...

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 series from 2011 on using the Freescale DSP 56000 series devices. The capability I wish to implement is as follows : 1. Accept the I2S input signal. 2. Implement a circuit that allows the DSP to be programmed whilst in s...

## Strumming problem.

inHi. I have a very accurate frequency detector that works pretty much perfectly with my guitar. I want to implement certain features that...

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 program displays-> checks for validity) my exact problem in an example can be explained like this: if i strum open E (82hz) once and my program demands an open E followed by an open E it will count both of these no