Why do RTA DEQs use pink noise?

Started by Impoliticus in comp.dsp14 years ago 3 replies

I'm working on a project for a DSP lab class and as you may have guessed from the topic, the project involves programming a real time analyzer...

I'm working on a project for a DSP lab class and as you may have guessed from the topic, the project involves programming a real time analyzer DEQ. Perhaps I should have asked this in the planning/research stage, but why do RTA equalizers, such as the DEQ2496, flatten the frequency response of the loudspeakers with respect to pink noise? I understand that pink noise has equal energy per octave, b...


Re: Newbie Matlab Digital Filter / Feedback Question

Started by Jerry Avins in comp.dsp8 years ago

On Thursday, January 13, 2011 8:44:24 AM UTC-5, j26 wrote: > I am trying to use a filter to smooth time-series data. ... > I have recently...

On Thursday, January 13, 2011 8:44:24 AM UTC-5, j26 wrote: > I am trying to use a filter to smooth time-series data. ... > I have recently started experimenting with the butter and cheby2 filter > design functions in Octave. Those are IIR filters. > I have a filter that smooths the input data > well (low-order / low cutoff frequency), but sometimes when the input > changes abruptly


Newbie Matlab Digital Filter / Feedback Question

Started by j26 in comp.dsp8 years ago 22 replies

I am trying to use a filter to smooth time-series data. Although it's been a good 5 years since my last signal and systems / control systems...

I am trying to use a filter to smooth time-series data. Although it's been a good 5 years since my last signal and systems / control systems course, I have recently started experimenting with the butter and cheby2 filter design functions in Octave. I have a filter that smooths the input data well (low-order / low cutoff frequency), but sometimes when the input changes abruptly, the output cannot...


Interpolated Constant-Q Equalizer

Started by in comp.dsp11 years ago 36 replies

Hello, We are implementing DSP based 31 band 1/3 octave audio graphic equalizer on programmabele DSP. The customer specifications demand...

Hello, We are implementing DSP based 31 band 1/3 octave audio graphic equalizer on programmabele DSP. The customer specifications demand for an Interpolated Constant - Q topology. For the implementation we are basing our design on Rane's Dennis Bohn paper on AES ("Constant-Q Graphic Equalizers", JAES Volume 34 Number 9 pp. 611-626; September 1986). Although the topologies suggested on the ...


fftshift equivilent in C

Started by Graham in comp.dsp16 years ago 3 replies

Does anyone have an equivilent to the matlab/octave/sci.py 'fftshift' routine that is written in C? I am using FFTW directly to compute rank 1...

Does anyone have an equivilent to the matlab/octave/sci.py 'fftshift' routine that is written in C? I am using FFTW directly to compute rank 1 and 2 FFTs but need to 'shift' the zero frequency component. -- FFTSHIFT Shift zero-frequency component to center of spectrum. For vectors, FFTSHIFT(X) swaps the left and right halves of X. For matrices, FFTSHIFT(X) swaps the first a...


help needed for Butterworth crossover response

Started by gangadhar.m in comp.dsp12 years ago 9 replies

Hi all, I implemented a butterworth crossover with the matlab (code given below). The code is working perfectly for a 4th order butterworth...

Hi all, I implemented a butterworth crossover with the matlab (code given below). The code is working perfectly for a 4th order butterworth crossover(i.e. 24 dB/Octave). Th added response for the lowpass and high pass filters of the butterworth crossover gives 3 dB increase at the corner frequency which is expected. But the same is not obtained when i use 2nd and 3rd order butterworth fi...


scilab help

Started by bharat pathak in comp.dsp11 years ago

hello, does anyone know how to lock the zoom button in scilab. for example: x = rand(1,100); plot(x) I want to zoom...

hello, does anyone know how to lock the zoom button in scilab. for example: x = rand(1,100); plot(x) I want to zoom again and again in various parts of the plot without clicking always the radio button called "zoom" on the top left of the plot. The radio button in active only for one zoom. unlike octave in which i could continously zoom to any area usi...


Some terms are confusing...

Started by Anonymous in comp.dsp11 years ago

Hi, all I am new in this field and don't understand some terms in standards. Recently I've read standards about Octave filters and Sound...

Hi, all I am new in this field and don't understand some terms in standards. Recently I've read standards about Octave filters and Sound level meter and I don't understand the definition of "level linearity error". Could someone explain "level linearity error" to me? It would be nice to have an example or how to test or find it. Thank you very much. Tim


Re: wavwrite problems

Started by david bateman in comp.dsp12 years ago

On May 27, 5:05 am, Randy Yates wrote: > I wrote: > > I'm using octave 2.9.9 on > > > [yates@localhost inp]$ uname -a > > Linux...

On May 27, 5:05 am, Randy Yates wrote: > I wrote: > > I'm using octave 2.9.9 on > > > [yates@localhost inp]$ uname -a > > Linux localhost.localdomain 2.6.20-1.2948.fc6 #1 SMP Fri Apr 27 19:18:54 EDT 2007 x86_64 x86_64 x86_64 GNU/Linux > > > and I'm experiencing two problems with wavwrite(): > > > 1. The parameter order changed between version 2.9.8 and 2.9.9. This is >


Higher upsampling with minimum phase downsampling produces more aliasing

Started by jungledmnc in comp.dsp5 years ago 20 replies

Hi, I'm programming a sound generator, based on wavetables. I have 8192 point wavetable. I create several band-limited "subwavetables" by...

Hi, I'm programming a sound generator, based on wavetables. I have 8192 point wavetable. I create several band-limited "subwavetables" by taking DFT, zeroing high octave(s) and IDFT. For generating particular pitch a choose a wavetable, which has all harmonics until 20k. Sound good so far, way better than just upsampling the original non-band-limited wavetable. The harmonics that exceed 22k ...


Good software list to enhance productivity of DSP designers.

Started by bharat pathak in comp.dsp11 years ago 18 replies

Hello All, So far we had discussions on listing good books on DSP and also some free books available online. Let us start this...

Hello All, So far we had discussions on listing good books on DSP and also some free books available online. Let us start this discussion thread, on which free softwares DSP designers use to enhance their productivity? Like for example octave/scilab/rlab or others. Please also mention that if the software runs only on windows /linux or both, and some useful features of ...


Hilbert transformation using real valued input FFT

Started by Anonymous in comp.dsp6 years ago 4 replies

Hello, I would like to use the hilbert transformation to get an amplitude envelope of a real valued (=measured) signal. Wikipedia describes it...

Hello, I would like to use the hilbert transformation to get an amplitude envelope of a real valued (=measured) signal. Wikipedia describes it nicely: https://en.wikipedia.org/wiki/Analytic_signal#Applications I implemented my tests in matlab/octave and it's easy: \n Hilbert transformation of x \n Shifts the signal by pi # function H=hilbert(x) N=length(x); X = fft(x);


RLB Weighting

Started by ikerr in comp.dsp9 years ago 1 reply

Greetings, I have been reading through the ITU-R...

Greetings, I have been reading through the ITU-R BS.1770-1 (http://webs.uvigo.es/servicios/biblioteca/uit/rec/BS/R-REC-BS.1770-1-200709-I!!PDF-E.pdf) document and have been trying to reproduce the frequency response of the RLB weighting using the supplied filter coefficients. However, when I use the supplied filter coefficients to construct a Bode plot in octave, the resulting plot attenuates ...