Generating Chirp/LFM by using IIR filters

Started by keta...@gmail.com in Audio Signal Processing13 years ago 1 reply

Help needed in designing chirp generator and pulse compressor using iir filters (this amounts to using an IDT, I guess). I want to generate...

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 to send a pulse to the iir filter and expect an output which is a chirp signal starting at f0 and ending at f1 in time duration t0. Also, I want to use the iir filter to generate pulse when fed with the above chirp. ...


Converting FIR from IIR

Started by itza...@yahoo.co.in in Audio Signal Processing12 years ago 6 replies

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...

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 linear phase. How can I achieve linear phase using IIR filters. Thank you, Abhijith


FIR in audio processing

Started by Rajmathi S in Audio Signal Processing10 years ago

Hi Rammya, ? 1. Why we choose IIR filter design for practical application, especially for processing audio signal? ? We can design an IIR...

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 filter giving us similar desired response. This greatly reduces the hardware resources required. On the flip side, IIR Filters have a non-linear phase response, which if acceptable, the filter is best designed as IIR. Als...


FIR & IIR : design,filtering

Started by akin...@hotmail.com in Audio Signal Processing11 years ago

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...

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 using fir and iir filtering functions from Paul Embree's book. I am building a PC\Windows application using Directshow and Winapi. In my application , I use these filtering functions with coefficients I already got , bu...


How to design an IIR filter using slope(db/oct) information

Started by care...@rediffmail.com in Audio Signal Processing15 years ago 4 replies

Topic : To design an IIR HPF filter using IPP(Intel Integrated Performance Primitives) library. Current need: to calculate BW or Q from...

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 : Cutt-off frequency : in Hz Slope : in dB/oct We are having an IPP library which has IIR filter implementation function. This function has following parameters: pTaps:- Pointer to the array containing the taps. The


Is the use of IIR filters for audio processing is a good idea?.

Started by kssubrahmanya in Audio Signal Processing14 years ago 2 replies

Hi all, what are the effects of using IIR filter on audio signal?.

Hi all, what are the effects of using IIR filter on audio signal?.


abt IIR filters and DSPs

Started by Sameer Kibey in Audio Signal Processing16 years ago

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...

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 fixed point processor is used instead? (the input to this filtering operation is 24 bit audio data samples). looking forward to some replies, Sameer.


Implementing IIR filter

Started by samu...@yahoo.com in Audio Signal Processing10 years ago

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...

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 with 2 section.. I wanna filter an audio signal so I use ADC to sample the signal and then... Can someone show to me the correct implementation steps? thanks


IIR audio filtering

Started by daro...@yahoo.fr in Audio Signal Processing11 years ago 1 reply

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...

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 basically I have an audio extract that i cut into 50% overlapping frames. Subsequently, i window each frame by a smoothing window. then I bandpass each frame F with an IIR filter (order N) using the function 'filter' pr...


FIR in audio processing

Started by ramm...@ymail.com in Audio Signal Processing10 years ago 3 replies

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...

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 application,especially for processing audio signal. 2. Why not FIR? 3. Depending on the response there are many IIR filter(butterworth,bessel etc).Is there any sub division for FIR? 4. What are the disadvantage of FIR filte...


resampling of an IIR filter

Started by stefanosorrentino in Audio Signal Processing17 years ago 2 replies

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...

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) frequency. I also have a signal, which is sampled at a much lower rate. When I filter my signal, I have to resample it at the same (high and unuseful) frequency of the filter, in order to have a correct filtering.


Overflow in IIR filters

Started by Carsten Borg in Audio Signal Processing19 years ago

I'm implementing a resampling (decimation) filter on a fixed-point platform (c54x) as a cascade of second-order-section IIR filters....

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 sensitive to overflow, unless I down-scale my input signal quite a lot (and thereby loose a great deal of signal-to-noise ratio). My question is as follows: Is there a specific second-order-section implementation (Direct


Exponenital Filter design

Started by gordon_ao in Audio Signal Processing15 years ago 1 reply

Hello Folks At the moment I am trying to design a one pole simple IIR digital filter to simulate exponential delay effect with simple...

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 is a known time constant. I think an one pole IIR filter will do it: Y[n] = alpha/[1-(1-alpha)Z^(-1)] can simulate the delay effect when alpha < 1. But I am quite pullzing how the alpha will be determined. i'd ap


Audio IIR filter never settles after the initial error input

Started by gordon_ao in Audio Signal Processing15 years ago

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...

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 input code from Sigma Delta modulator is 1010101010... forever which is supposed to produce all 0 PCM code. However, SINC decimation after the modulator filter went through a period of transient stage (with a few non-


Audio Equalizer

Started by akin...@hotmail.com in Audio Signal Processing11 years ago 2 replies

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...

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, Akin Ocal


About Implementing A Sound Equalizer

Started by akin...@hotmail.com in Audio Signal Processing11 years ago 1 reply

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...

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 sample code in C\C++ for building an equalizer ? Best Regards, Ak?n Öcal


slope of a filter

Started by ramm...@ymail.com in Audio Signal Processing10 years ago

hi all Thanks to all who ever respond to my previous queries. I want the technical counterpart or derivation regarding 5the slope of...

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 dB/decade (or 6 dB/octave) slope. 2. Any website that describe in detail about FIR and IIR Filters. with regards rammya


90 degrees phase shifter

Started by palm...@hotmail.com in Audio Signal Processing10 years ago

Hi, I need to make a 90º phase shifter for a digital crossover audio system. I've heard something about Hilbert Transform but I would like to...

Hi, I need to make a 90º 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 phase correction. Does anybody know if I can implement a 90º phase shifting with cascade IIR biquad sections? Thanks in advence


Howling rejection

Started by Curl in Audio Signal Processing18 years ago

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...

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 frequencies, and fixed-point implementation) I read : "howling rejection, typically done by shifting the frequency of the signal as it goes through the canceller." (.. through an AEC) But i didnt find any docum


Data types in Audio Signal Processing

Started by akin...@hotmail.com in Audio Signal Processing11 years ago 1 reply

Hello I want to implement my own audio equalizer. First of all I create FIR& IIR bandpass filters but they process"doubl" or "float"...

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 shorts ( 16bit) from both Directshow and Waveout API. How can I handle data type conversion between floaats and shorts ? They say that i must divide shorts by 32768 to get floats. But how should I put my processed flo...