Audio application problem

Started by PROVENTEK MINDCRAFT AB in comp.dsp15 years ago 19 replies

Hi folks, I'm working with an dsp audio application and I desperately need an algorithm for tone control. The filter is described in my...

Hi folks, I'm working with an dsp audio application and I desperately need an algorithm for tone control. The filter is described in my spec as a Baxendall filter with the axial point of 1kHz and the characteristics is a maximum of +-6dB per octave. So far I have implemented low and high pass second order Butterworh IIR filters with coefficients calculated with Matlab and they work f...


IIR filter design

Started by in comp.dsp15 years ago 4 replies

Hi all ! Thank you for reading this post. I have a small problem. I would like to design an IIR filter for a FPGA and simulate it with...

Hi all ! Thank you for reading this post. I have a small problem. I would like to design an IIR filter for a FPGA and simulate it with octave/matlab to see if it works as it should. So I don't use the included 'filter' function, but I designed it myself. As one can see it in the code provided here, the design works for 2nd order filters, but not for orders 3 and more. Who could help me ?? ...


SPDIF EQ

Started by Dirk Bruere at NeoPax in comp.dsp9 years ago 7 replies

Does anyone do a s/w 3rd octave equalizer that will work on a stereo (non dolby etc) SPDIF output (from a mobo or soundcard)? --...

Does anyone do a s/w 3rd octave equalizer that will work on a stereo (non dolby etc) SPDIF output (from a mobo or soundcard)? -- Dirk http://www.transcendence.me.uk/ - Transcendence UK http://www.theconsensus.org/ - A UK political party http://www.blogtalkradio.com/onetribe - Occult Talk Show


Constant Q spectrogram Vs FFT Spectrogram?

Started by Simon in comp.dsp13 years ago 1 reply

I am a newcomer to Spectrogram analysis. It is said, there are two kind of spectrogram, one is constant Q spectrogram(octave band...

I am a newcomer to Spectrogram analysis. It is said, there are two kind of spectrogram, one is constant Q spectrogram(octave band filter,logrithm of frequency ).The other is FFT spectrogram. I knew a little about FFT spectrogram.But I am not so sure what's the difference between them? advanced Thanks Simon


design a filter from material absorption coefficients

Started by Emile in comp.dsp13 years ago 4 replies

Hi, Can anybody point me to a paper/book/site/algorithm/tool on how to desing a low order filter to model the absorption coefficients of...

Hi, Can anybody point me to a paper/book/site/algorithm/tool on how to desing a low order filter to model the absorption coefficients of materials. I have found that these coefficients usually are given for 6 octave bands (125,250,500,1000,2000,4000Hz). More information on this subject is greatly appreciated. Emile Vrijdags


fastest accurate spectrum analyser routine

Started by robert_temp in comp.dsp3 years ago 13 replies

Hi, I'm implementing a spectrum anaylzer with very limited memory. My intention is to provide 1/3 octave bands from 20hz to 5Khz accurate...

Hi, I'm implementing a spectrum anaylzer with very limited memory. My intention is to provide 1/3 octave bands from 20hz to 5Khz accurate to +-1db with a 10k sampling rate. I have .5K of RAM to play with so you can see my problem! The platform is an atmega328 with around 8k of code space left. What does anyone reckon might work, accuracy can drop & sample rate could be lowered further. ...


practical FFT

Started by Sharan123 in comp.dsp3 years ago 40 replies

Hello, I am having some questions related to FFT while using inbuilt Matlab or Octave FFT functions. These functions seem to take N samples...

Hello, I am having some questions related to FFT while using inbuilt Matlab or Octave FFT functions. These functions seem to take N samples of the signal that needs to be analyzed. Are there any requirement on the length of the input signal that needs to be fed into. A signal can be made up of many frequencies, so, it is obvious that a signal of N length can have any or all of the follow...


circular array beamforming

Started by hassanhamdoun859 in comp.dsp11 years ago

Dear all I am trying to calculate the broadband radiation pattern(beampattern) of a circular array beamformer operating in a given frequency...

Dear all I am trying to calculate the broadband radiation pattern(beampattern) of a circular array beamformer operating in a given frequency range(1 octave) such that the main lobe response is approximately similar for all frequencies i.e the beampattern is frequency invariant I am trying to do that in MATLAB , I need help references matlab documentation or any usefull papers please help...


output of bandpass filter?

Started by BobG in comp.dsp14 years ago 7 replies

Trying to make an octave band analyzer by computing a BP filter, running all the input samples thru it, and summing all the output samples,...

Trying to make an octave band analyzer by computing a BP filter, running all the input samples thru it, and summing all the output samples, recompute filter and repeat for N frequencies. I suspect there is something wrong with getting output ampl by summing all the samples. How do smart people solve this problem??


Problem Using Kiss FFT

Started by Javier Pajuelo in comp.dsp5 years ago 10 replies

I get different results using Kiss FFT and Octave/Matlab fft. Here's the input/output of my...

I get different results using Kiss FFT and Octave/Matlab fft. Here's the input/output of my code: ========================================================= xtmp[2][4] 0 1 2 3 0 (0.2348,-0.5121) (-0.2745,0.0647)(-0.06092,-0.4704) (-0.04771,0.1112) 1 (-0.14,-0.04515)(-0.5469,0.4785) (-0.3687,-0.2009)(-0.04413,0.1298) Input: Output: 0.234797 -0.512116 -0.1706950.026731 -0.27448...


invfreqz matlab/octave function, underlying theory

Started by banton in comp.dsp10 years ago 2 replies

Hello, I would like to write (or find) a modified version of "invfreqz" that gives poles and zeros instead of coefficients. Of course I could...

Hello, I would like to write (or find) a modified version of "invfreqz" that gives poles and zeros instead of coefficients. Of course I could first compute the coefficients and then convert to poles and zeros by root-finding, but I wonder if this isn't an unecessary detour. Can anybody point me to documents that give me some insight into the underlying mechanism applied by "invfreqz". Of c...


negative group delay

Started by bharat pathak in comp.dsp11 years ago 5 replies

Hi, I am seeing some negative group delay at 1000, 2000, 3000, 4000, 5000 Hz. The example filter listed below with matlab/octave...

Hi, I am seeing some negative group delay at 1000, 2000, 3000, 4000, 5000 Hz. The example filter listed below with matlab/octave code. any clue why this is happening? interesting part is when I feed 1000 hz sine wave sampled at 10000 hz I don't see the behaviour of negative group delay. Regards Bharat %-----------------------------------------------------...


Re: wavwrite problems

Started by Randy Yates in comp.dsp12 years ago

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...

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 > not really a problem, but may be noteworthy. > > 2. wav


How much trouble can a NEWBIE get into doing SIMPLISTIC filter in frequency domain

Started by Richard Owlett in comp.dsp15 years ago 8 replies

[ has background related to [NEWBIE question] How do I filter a digitized signal ] I have a real [only] input {voice in .WAV format) I...

[ has background related to [NEWBIE question] How do I filter a digitized signal ] I have a real [only] input {voice in .WAV format) I do a Scilab fft on it. I multiply the resulting frequency domain by a "filter function" *CASE 1* Value of filter is either 1 or zero depending on frequency The filter will have same value over intervals of at least 1/12 octave There will be mult...


Reduce Ringing in FIRs

Started by Randy Yates in comp.dsp3 years ago 22 replies

Gentle comp.dsp readers, A requirement just popped up in one of my projects to redesign an FIR lowpass filter so that it doesn't create so...

Gentle comp.dsp readers, A requirement just popped up in one of my projects to redesign an FIR lowpass filter so that it doesn't create so much time-domain ringing on spikes. I had originally used octave's firls to design a linear-phase lowpass filter (Fs = 625 Hz, Fp = 100 Hz, Fs = 150 Hz), but its response has got a lot of wigglies. Any thoughts/papers/google searches/bickering/smart r...


Sample data log code for DSP?

Started by aojkim in comp.dsp9 years ago 24 replies

Hello, I am an intern for a lab, and I recently installed P25M from Innovative Integration on a pc. I was told to use Octave, which I found...

Hello, I am an intern for a lab, and I recently installed P25M from Innovative Integration on a pc. I was told to use Octave, which I found easily. My next step, however, is this: "Get a sample code for data logging using DSP" I AM STARTING COMPLETELY FROM SCRACTH. PLEASE HELP ME. Answers such as "read a book first," "read basics first" will not help because I was told to "get a sampl...


prototyping and migrating code to production: MATLAB, C, C++, asm

Started by robert bristow-johnson in comp.dsp5 years ago 25 replies

okay guys, what do you do, if you start with an alg that works pretty good in MATLAB/Octave/MathCad/Mathematica/whatever, and you wanna make...

okay guys, what do you do, if you start with an alg that works pretty good in MATLAB/Octave/MathCad/Mathematica/whatever, and you wanna make production code out of it. let's pretend the production code will be C. what do you do? rewrite it yourself, line by line? use http://www.mathworks.com/products/matlab-coder/ ? something else? (i guess i would also be interested in hearing...


IIR audio filterbank

Started by Robert Adams in comp.dsp8 years ago 19 replies

I am looking for a way to design an audio IIR filterbank. I need to split an audio signal into roughly 1/3 octave bands with 6th-order...

I am looking for a way to design an audio IIR filterbank. I need to split an audio signal into roughly 1/3 octave bands with 6th-order or 8th-order IIR bandpass filters and then sum them back together again with a ripple of +/- 1 dB. I can do a "seat-of-the-pants" design that works well until I approach Nyquist where it kind of falls apart. I have very limited MIPs so I can't do anything very...


Using Multirate algorithm for doing pitch shift effect ?

Started by Eudes in comp.dsp14 years ago 18 replies

Hi, I'm trying to implements a pitch shift effect (up / down a .wav sound 44100Hz 16bit to different semitones in the range of around 1...

Hi, I'm trying to implements a pitch shift effect (up / down a .wav sound 44100Hz 16bit to different semitones in the range of around 1 octave). My problem is that I need to do it on about 15 sample simultaneously in real time. The classic pitch shift algorithms use too much of processor time. As my samples are looped, I don't care about the lenght modification of my samples. That's why, ...


IIR constant Q bandpass filters

Started by loopy in comp.dsp13 years ago 1 reply

Hi, I would like to apply constant Q IIR filters to a music signal of 44100Hz where each filter matches the frequency of the western well...

Hi, I would like to apply constant Q IIR filters to a music signal of 44100Hz where each filter matches the frequency of the western well tempered tuning system. This equates to 12 tones per octave. At the moment I have achieved this using large custom FIR filters in Matlab and it takes 5 minutes to process 1.1s of data! I would like to apply IIR filters and end up with an envelope of the...