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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

FIR in audio processing

Rajmathi S - Jan 10 2010
Hi Rammya,   1. Why we choose IIR filter design for practical application, especially for processing audio signal?   We can design an IIR Filter of far less order than an FIR f... FIR in audio processing

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

FIR in audio processing   [4 Articles]

ramm...@ymail.com - Jan 9 2010
hi... I'm new in dsp field.Just a fresher working for Audio product based company. i have some queries please help me. 1. why we choose IIR filter design,for practical applica... FIR in audio processing

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

Implementing IIR filter

samu...@yahoo.com - Jan 8 2010
Hi, I have just designed an IIR filter in Matlab and I need to implement it in a dsPIC, but I think there is some mistakes in my implementing procedure! I have a 4th order filter ... Implementing IIR filter

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

Re: reinventing the 1/3 octave wheel   [3 Articles]

David Reaves - May 14 2009
Bill, Given enough coefficients, FIR filters can replicate just about ANY curve. But, when creating banks of octave-related (or 1/3 octave) filters, IIR filters are more ... Re: reinventing the 1/3 octave wheel

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

slope of a filter

ramm...@ymail.com - Jan 28 2010
hi all Thanks to all who ever respond to my previous queries. I want the technical counterpart or derivation regarding 5the slope of filter 1. why 1st order filters have a 20... slope of a filter

90 degrees phase shifter

palm...@hotmail.com - Nov 26 2009
Hi, I need to make a 90Âș phase shifter for a digital crossover audio system. I've heard something about Hilbert Transform but I would like to know if it is the best way to do a ... 90 degrees phase shifter

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

Allpass filters design with a desired phase response

palm...@hotmail.com - Dec 27 2009
Hi, I know, IIR allpass filters are used for phase equalization Can anybody help me about how I can get the coeficients for a desired phase response? Thanks ... Allpass filters design with a desired phase response

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

Chamberlin state variable filter transfer function   [3 Articles]

kt...@am3d.com - Mar 29 2009
Hi, I am comparing coefficient quantization errors for second order IIR filter structures (for fixed point implementation). I have found the Chamberlin state variable structure to... Chamberlin state variable filter transfer function

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