Parallel Decimation Filter Implementation?

Started by Phil in comp.dsp13 years ago 1 reply

Hi, I'm hoping that someone out there may be able to help me solve this problem. I'm trying to create a decimate by 2 filter that will have...

Hi, I'm hoping that someone out there may be able to help me solve this problem. I'm trying to create a decimate by 2 filter that will have 4 parallel inputs and 2 outputs. The reason for doing this is that I am receiving data into my FPGA at 4 times the clock FPGAs clock rate. This is being done by bringing in 4 samples per clock. I was thinking of using a polyphase filter, but this...


Relationship between FIR frequency response and each polyphase response

Started by gretzteam in comp.dsp5 years ago 19 replies

Hi, I'm thinking of this in the context of decimation (but same thing for interpolation). As an example, say we want to decimate by 8. In...

Hi, I'm thinking of this in the context of decimation (but same thing for interpolation). As an example, say we want to decimate by 8. In the frequency domain, our goal is to design a filter that attenuates the frequencies that will alias when we through away 7 samples. You go off and design an FIR filter that gives you say 60dB worst case in the stopbands. So far so good. When comes time to i...


Sample Rate Conversion (Downsampling)

Started by in comp.dsp15 years ago 24 replies

Hello, For a project which I'm working on, I have a 60Hz sinewave modulated with a square wave at 0.5Hz, modulation is 2.72% The signal...

Hello, For a project which I'm working on, I have a 60Hz sinewave modulated with a square wave at 0.5Hz, modulation is 2.72% The signal comes sampled at 8kHz, I should downsample it to 200Hz, so M = 40. I do that in two stages, first with M1 = 10, then with M2 = 4. I implemented polyphase structure in Simulink, and runs great. By "great" I mean that the output signal, sampled at 200Hz, p...


Interpolation using polyphase FIR filter: Finding coeffisients

Started by MrAlfred in comp.dsp13 years ago 18 replies

Hi, I have a datastream that is sampled 4800 times pr second, but the rf front end I am going to use accepts only input at 250000 samples pr...

Hi, I have a datastream that is sampled 4800 times pr second, but the rf front end I am going to use accepts only input at 250000 samples pr second. Clearly I need a 52,0833 times interpolation. To achieve this I have planned to first interpolate 625 times, and then decimate 12 times. I will do this in steps, for instance in steps of 5 for the interpolation, and 4 for the decimation. I...


Where can I find comparision between symmetric and asymmetric TAP FIR filters?

Started by vijay in comp.dsp11 years ago 3 replies

Newbie Here. I have been told that and we normally go for 3,5,7,9 Tap poly phase filters for image processing.... and 4,6,8 Tap filters...

Newbie Here. I have been told that and we normally go for 3,5,7,9 Tap poly phase filters for image processing.... and 4,6,8 Tap filters should not be used... I am not able to find any reference online to understand the issues with even tap polyphase filters. Can someone point me to a suitable online reference. Regards Vijay


Quadriphase barker sequences

Started by Anonymous in comp.dsp15 years ago 2 replies

Hello, I am looking for a quadriphase Barker sequence (a sequence whose elements are from the set {1,j,-1,-j} and which has a...

Hello, I am looking for a quadriphase Barker sequence (a sequence whose elements are from the set {1,j,-1,-j} and which has a periodic autocorrelation function whose maximum sidelobe magnitude


Help polyphase implementation

Started by bhaskar.nallani in comp.dsp10 years ago 2 replies

Dear all, I want to convert the signal from 44.1 to 48kHz. I am using an pi/160 FIR filter of length around 8k. When I up sample to 147 times...

Dear all, I want to convert the signal from 44.1 to 48kHz. I am using an pi/160 FIR filter of length around 8k. When I up sample to 147 times and filter and down sample to 160, I am getting good results. So for efficient implementation I am implementing the same with poly phase structure. The input signal increments with a time_stride of 147/160 and I need to find the exact phase of th...


How to interpret polyphase coefficients generated in MATLAB

Started by vizziee in comp.dsp11 years ago 24 replies

Hello everyone, I am trying to design a low pass decimator filter in MATLAB. I am supposed to decimate a signal sampled at 200MHz down to...

Hello everyone, I am trying to design a low pass decimator filter in MATLAB. I am supposed to decimate a signal sampled at 200MHz down to 10MHz. The signal bandwidth is 8 MHz and the signal spectra is centred at the sampling frequency. I began with the following code: --------------------------------------------------------------- Fs_adc = 200e6; % ADC Sampling Frequency Fpass1 =...


polyphase, interpolation, timing error

Started by Anonymous in comp.dsp14 years ago 5 replies

I've been following a number of threads over the past couple of weeks regarding fractional timing delays and interpolation, but none of...

I've been following a number of threads over the past couple of weeks regarding fractional timing delays and interpolation, but none of them really address my problem, so here goes.... I have a sampled received signal taken at r(kTs), Ts = sample rate and k is integer. The sampled signal is then fed into a bank of matched filters (MFs) (my system has a min number of 2 MFs and a max of 16 M...


fixed point issue inSRC

Started by srikk in comp.dsp13 years ago 6 replies

Hi, I am implimenting a SRC for both decimation and interpolation using polyphase filter implementation. my question is 1.i take a...

Hi, I am implimenting a SRC for both decimation and interpolation using polyphase filter implementation. my question is 1.i take a sine wave (+1 to -1) of 24 bit and convolving with filter taps of 1.31 format. 2. when i save my result after filtering to 24 bit, i need to truncate the result to 24 bit and save it 3. in the above process i found the output sine wave having a ...


fixed point issue inSRC

Started by srikk in comp.dsp13 years ago

Hi, I am implimenting a SRC for both decimation and interpolation using polyphase filter implementation. my question is 1.i take a...

Hi, I am implimenting a SRC for both decimation and interpolation using polyphase filter implementation. my question is 1.i take a sine wave (+1 to -1) of 24 bit and convolving with filter taps of 1.31 format. 2. when i save my result after filtering to 24 bit, i need to truncate the result to 24 bit and save it 3. in the above process i found the output sine wave having a ...


ripple/Noise floor after decimation

Started by yogesh_gharote in comp.dsp9 years ago 9 replies

I have a ADC system which samples at 160 MHz . I am passing the data through a narrowband FIR filter of bandwidth 2.5 MHz . This also amounts...

I have a ADC system which samples at 160 MHz . I am passing the data through a narrowband FIR filter of bandwidth 2.5 MHz . This also amounts to saying that i can filter & decimate the data sequence (to the input of the filter) by a factor of 32 (80/2.5 = 32) . The FIR is implemented as a polyphase structure. Filter specs are as follows: Input data rate : 160 MHz (160 mega samples...


Polyphase & wavetable playback?

Started by R Jones in comp.dsp15 years ago 4 replies

Hi. I am looking to add "wavetable" playback of sampled music/instruments to an existing project. The design already uses an FPGA & has an...

Hi. I am looking to add "wavetable" playback of sampled music/instruments to an existing project. The design already uses an FPGA & has an audio CODEC which supports playback at 48 kHz. I want to implement pitch-shifting and mimic higher/lower notes by altering the playback rate of the samples. The pitch-shift is in the range of -2x to 2x (-2*sampling_rate to +2*sampling_rate) controll...


A question from Vaidyanathan's "Multirate Systems" book

Started by Rick Lyons in comp.dsp15 years ago 13 replies

Hello Earthlings, I recently ran across some material in Vaidyanathan's "Multirate Systems and Filter Banks" DSP book that discusses a...

Hello Earthlings, I recently ran across some material in Vaidyanathan's "Multirate Systems and Filter Banks" DSP book that discusses a way to improve the computational efficiency of polyphase filters used in non-integer decimation applications. Vaidyanathan's description (starting on page 128, and derived from a 1987 conference paper by C.-C Hsiao) sounds interesting except ...


CIC versus Polyphase FIR

Started by sachinwannabe in comp.dsp12 years ago 4 replies

I am trying to understand the real benefits of using a CIC filter for rate sample change v.s a conventional multi-rate FIR filter. My thoughts on...

I am trying to understand the real benefits of using a CIC filter for rate sample change v.s a conventional multi-rate FIR filter. My thoughts on CIC v/s traditional sampling rate conversion methods below: I understand the multiplier-less advantage of CIC filters. However, because of their spectral content, one invariably needs a compensation filter. The good thing about the compensation filter...


Scilab, sparse matricies, and polyphase filtering

Started by Tim Wescott in comp.dsp11 years ago 6 replies

The subject line says it all. I need to resample some data in Scilab -- I'm getting data sampled at 5MHz that I need to resample to 4.9...

The subject line says it all. I need to resample some data in Scilab -- I'm getting data sampled at 5MHz that I need to resample to 4.9 some-odd MHz. When I was using 20ms long data sets it worked very well to do an FFT, trim bins out of the middle to reduce the frequency by the right factor, then do an IFFT. Now we're doing the "real" algorithm with 1/2 second long data sets, and i...


How to resample fast

Started by s036 in comp.dsp15 years ago 5 replies

I need to write a mixer of different sample-rate input and need resample function. I implement interpolation filter and decimation filter...

I need to write a mixer of different sample-rate input and need resample function. I implement interpolation filter and decimation filter by polyphase decomposition. However, it takes large computation time when convert 11KHz -> 48KHz, or 44KHz -> 48KHz. I saw using a windowed sinc interpolator can do this. Is anybody explained the algorithm ? Or any fast algorithm to do resample except l


multirate filter design

Started by Anonymous in comp.dsp13 years ago 5 replies

Hi, When one designs a fractional rate change filter, when specifying the filter parameters do you design the filter at the interpolated rate...

Hi, When one designs a fractional rate change filter, when specifying the filter parameters do you design the filter at the interpolated rate or the decimated rate? For example if you have an input rate of Fs then interp by 64 decimate by 125 filter and the filter will be implemented in a P/Q polyphase fashion do you design the filter at the 64*Fs rate? Assuming that the signal is alr...


PQMF Filters

Started by coolup in comp.dsp15 years ago 2 replies

When I checked for PQMF filters in Wikipedia , I got the following information. It says that signal in odd subbands is stored frequency...

When I checked for PQMF filters in Wikipedia , I got the following information. It says that signal in odd subbands is stored frequency inverted. Does anybody know how the frequency is inverted? > From Wikipedia, the free encyclopedia A polyphase quadrature filter, or PQF is a filter bank, which splits an input signal into a given number N (mostly a power of 2) of equidistant sub-bands.


Regarding Acoustic Echo Cancellation with subbanding

Started by Aparna Ram in comp.dsp13 years ago

Dear Sir, I am working on Acoustic Echo Cancellation. I have implemented both AEC without subbanding and AEC with subbanding in...

Dear Sir, I am working on Acoustic Echo Cancellation. I have implemented both AEC without subbanding and AEC with subbanding in C+ +. For subbanding of AEC I used Polyphase IIR filter(with 70dB attenuation). AEC without subbanding is working fine. But in the outputs of AEC with subbanding there exists some slight disturbance(but echo is cancelling for all recorded...