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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).

Search Results for "iir"

  

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Converting FIR from IIR   [7 Articles]

itza...@yahoo.co.in - Sep 20 2007
Hi all, I have designed a FIR filter which has 60 DB SNR.Also I designed IIR filter at 60 DB SNR. FIR taps is around 700 whereas IIR is 12. So I want to design IIR.But I want li... Converting FIR from IIR

Generating Chirp/LFM by using IIR filters   [2 Articles]

keta...@gmail.com - Sep 5 2006
Help needed in designing chirp generator and pulse compressor using iir filters (this amounts to using an IDT, I guess). I want to generate chirp using an IIR filter. I want... Generating Chirp/LFM by using IIR filters

Audio IIR filter never settles after the initial error input

gordon_ao - Jun 5 2004
Hello all, A simple question actually. Just found an issue with IIR low pass filter in decimation low pass after the SINC filter in a Sigma Delta A/D system. The one bit inp... Audio IIR filter never settles after the initial error input

Is the use of IIR filters for audio processing is a good idea?.   [3 Articles]

kssubrahmanya - May 6 2005
Hi all, what are the effects of using IIR filter on audio signal?. ... Is the use of IIR filters for audio processing is a good idea?.

abt IIR filters and DSPs

Sameer Kibey - Jul 25 2003
hi all I am interested in implementation of cascaded second order IIR filters. Do i necessarily have to use a floating point DSP for this application? Any serious drawback if a... abt IIR filters and DSPs

IIR audio filtering   [2 Articles]

daro...@yahoo.fr - Sep 1 2008
Hi ! I have little experience with time filtering ,ad i'd very much appreciate some input. So here's the deal: I'm trying to enhance audio with some non-linear processing so bas... IIR audio filtering

resampling of an IIR filter   [3 Articles]

stefanosorrentino - Nov 28 2002
Hello friends. I have an apparently unsolvable problem: I have a given IIR filter (they give me the poles and the zeros). The filter was originally sampled at a given (very high... resampling of an IIR filter

How to design an IIR filter using slope(db/oct) information   [5 Articles]

care...@rediffmail.com - Mar 3 2005
Topic : To design an IIR HPF filter using IPP(Intel Integrated Performance Primitives) library. Current need: to calculate BW or Q from slope. The specification of filter is :... How to design an IIR filter using slope(db/oct) information

Exponenital Filter design   [2 Articles]

gordon_ao - Jan 24 2005
Hello Folks At the moment I am trying to design a one pole simple IIR digital filter to simulate exponential delay effect with simple equations like 1 - e^(-t/T), where T i... Exponenital Filter design

1-Bit Goertzel Algorithm Performance   [2 Articles]

dn...@newelltech.com - May 14 2007
I'm implementing a DTMF detector using the Goertzel Algorithm. I have raw simulations built using Excel (I know...) and am getting some interesting results. I don't see a lot o... 1-Bit Goertzel Algorithm Performance

Overflow in IIR filters

Carsten Borg - Jan 3 2001
I'm implementing a resampling (decimation) filter on a fixed-point platform (c54x) as a cascade of second-order-section IIR filters. Unfortunately my implementation is very sens... Overflow in IIR filters

How to design filter to match both magnitude and phase

gordon_ao - Sep 11 2003
Hello folks, I would really appreciate your insight in terms of how to design digital filters to match both magnitude and phase of a narrow band complex Analog gain(simple 1s... How to design filter to match both magnitude and phase

Audio Equalizer   [2 Articles]

akin...@hotmail.com - May 5 2008
Hi I want build a multiband ( 10 band) 2-channel audio equalizer with FIR or IIR filters. Can you suggest any sample project or ready library for that purpose ? Regards, Aki... Audio Equalizer

About Implementing A Sound Equalizer   [2 Articles]

akin...@hotmail.com - Apr 20 2008
Hello , I want to build a multiband equalizer. I know that I must create bandpass filters for that aim. So should they be FIR or IIR filters and why ? Is there any sampl... About Implementing A Sound Equalizer

FIR & IIR : design,filtering

akin...@hotmail.com - Apr 23 2008
Hello I am using some FIR and IIR filters. I use coefficients from Dsptutor's java applets. First of all , have you ever experienced Dpstutor's designs on projects ? I am us... FIR & IIR : design,filtering

Octave and 1/3-octave bandpass filter bank   [6 Articles]

hank...@in.waw.pl - Apr 8 2005
Hi everyone, My previous post was not submitted, so I decided to write it again. I'm supposed to design a bandpass filter bank (octave and 1/3-octave) for audio measurements ... Octave and 1/3-octave bandpass filter bank

Data types in Audio Signal Processing

akin...@hotmail.com - Apr 20 2008
Hello I want to implement my own audio equalizer. First of all I create FIR& IIR bandpass filters but they process"doubl" or "float" samples. But I get sound samples as sh... Data types in Audio Signal Processing

Howling rejection

Curl - Oct 16 2001
Hello My aim is to set up a howling detector (and canceller) I tried an adaptive IIR notch filter (using LMS), but I have still problems with this system (due to harmonics freq... Howling rejection

Re: Frequency response

Erik de Castro Lopo - Nov 29 2007
rendoddi wrote: > Suppose you have H(z)=B(z)/A(z), the transfer function of a digital > filter. > How do you calculate the ANALYTIC expression of the filter's impulse > r... Re:  Frequency response

fractional-octave band filters via FFT

normanrg - Feb 26 2004
Does anyone out there have experience or info in how to construct fractional-octave bandwidth filters using FFTs? It does not seem to be an accepted method, yet I cannot under... fractional-octave band filters via FFT

Re: Amplitude Estimation

Bernhard Holzmayer - Jun 21 2006
1st of all I'm using 512 taps on a 256 Samples Per Second signal. I know I can eliminate the coefficients that are close to zero and still get pretty much the same response... Re:  Amplitude Estimation

Optimal envelope extraction from an audio signal   [3 Articles]

Brett Carruthers - Apr 27 2006
Hello, I am currently using a digital 1000 point lowpass filter to extract the envelope from an audio signal of which I have removed the negative components (by using the ab... Optimal envelope extraction from an audio signal

kingsbury filter   [2 Articles]

Goh Kean Leong RBMA/ESD - Sep 2 2002
hi all forum yahoo group members, I have few question regarding a type of IIR filter structure that name as "kingsbury filter". This type of filter structure is claim to b... kingsbury filter

RE: Speech Noise

Egler, Mark - Mar 7 2002
Marcio,   If you design a second-order low-pass Butterworth "analog-prototype" IIR filter, which just about any digital filter design package can ... RE:  Speech Noise

Biquad filtering with Fs/Fo > 200   [4 Articles]

dgen...@engmail.uwaterloo.ca - Oct 29 2007
I'm designing a parametric equalizer using a spartan 3e starter board and a TI PCM codec I wired to it, running at 48kHz. I decided to use the biquad IIR filter core from opencore... Biquad filtering with Fs/Fo > 200

RE: AUTOMATIC EQUALIZER

THEOPHILUS MEDEIROS - Jan 28 2005
I thought of using adaptive filtering, but since it's a pahse coerrection. It might not be that easy. I plan on doing this using offline filtering. I just have to calibrate... RE:  AUTOMATIC EQUALIZER

Re: Re: Equalizers   [3 Articles]

Jeff Brower - Mar 3 2008
Gene- > The advantage of the parallel architecture is that it will give you the same filter combining response as an analog > graphic EQ; that is, adjacent filters will inter... Re:  Re: Equalizers

Re: DTMF decoding using fixed-point arithmetic

Amaresh patil - Apr 28 2006
Hi The format of the data needs careful selection. it is not strainght. I would suggest you to first find out the max and min value of delay output data format and current o... Re:  DTMF decoding using fixed-point arithmetic

Re: audio interpolation

Grzegorz Kraszewski - Dec 15 2005
Hello sunil@suni... > Please, can anybody tell how can I remove the hiss noise from the > interpolated audio signal? Here, I am simulating the interpolation using > c-program a... Re: audio interpolation

Re: Guitar Valve Preamp with DSP

Bernhard Holzmayer - Oct 2 2006
On Tuesday 26 September 2006 07:09, williamluisterry wrote: > Hi. I've working with al TMS320C613 DSK building some guitar effects. > I was suscesfully whit all of these but I ... Re:  Guitar Valve Preamp with DSP

Re: Filter Order in Analog and Digital   [6 Articles]

Jeff Brower - Mar 22 2005
Glidden- > Thanks for you email. In the figure, I see the slope (coming down on > the right side) change, just like mine (it just looks like a sag) !!! I will > sent you t... Re:  Filter Order in Analog and Digital

RE: Unknown Filter Type

Egler, Mark - Jan 14 2005
I looked it up myself, and I was wrong. A Volterra filter is not recursive, i.e. it's FIR but with polynomial terms. What Jon has is recursive (IIR) and I haven't found a... RE:  Unknown Filter Type