Polyphase decimating filter - not working - why?

Started by Tamara in comp.dsp12 years ago 18 replies

Hello...I created a polyphase decimating filter based on F.Harris's book shared register design. This is a Decimation by 2, 100 tap, Fs...

Hello...I created a polyphase decimating filter based on F.Harris's book shared register design. This is a Decimation by 2, 100 tap, Fs =100MHz, Low pass 20MHz - stop band 24MHz filter. Each MAC is 28-bit length, 14-bit ADC input, and the filter output is truncated. The filter looks good at first, starts attenuating at 20MHz, and then cuts off about 23MHz.. but when I keep increasing fr...


Polyphase Filterbanks for Symbol Timing Synchronization

Started by mir_aculous in comp.dsp9 years ago 1 reply

Hi, I am currently reading fred harris' papers mentioned below to implement a timing synchronization in my software DSSS receiver. A. Loop...

Hi, I am currently reading fred harris' papers mentioned below to implement a timing synchronization in my software DSSS receiver. A. Loop Control Architectures for Symbol Timing Synchronization in Sampled Data Receivers B. Polyphase Fitlerbanks for Symbol Timing Synchronization in Sampled Data Receivers Although, I get the big picture behind the idea of implementing timing synchronization...


44.1 khz to 48 khz using polyphase

Started by nomi in comp.dsp14 years ago 9 replies

Hi, I am trying to do a 44.1 khz to 48 khz resampling in Matlab, but am coming across a few problems. For this conversion to take place...

Hi, I am trying to do a 44.1 khz to 48 khz resampling in Matlab, but am coming across a few problems. For this conversion to take place the poly phase filters come out like 480/441=(4 x 4 x 10)/(7 x 7 x 3). I have been trying to figure this thing out on paper, like the for the first stage I get 44100 x (4/7), now as I understand it I have to take a 44.1 khz signal upsample it by 4, appl...


Polyphase interpolation followed by decimation, do I get original data?

Started by Mr. Ken in comp.dsp13 years ago 6 replies

I am using an 80 tap matched filter to interpolate a signal by 10 times, which splits the coefficients into 10*8taps. After that, I will decimate...

I am using an 80 tap matched filter to interpolate a signal by 10 times, which splits the coefficients into 10*8taps. After that, I will decimate the result by 10 using same filter. Do I get back original data (ignore the delay)?


Digital Power Control

Started by Rob Gaddi in comp.dsp4 years ago 70 replies

Got an application coming up for a high-efficiency, multikilowatt polyphase power supply that needs scarily fast responses to external...

Got an application coming up for a high-efficiency, multikilowatt polyphase power supply that needs scarily fast responses to external requests. I'm thinking about maybe throwing a C2000 at the problem. It could really simplify keeping my phases in alignment and getting the feedforward through all the paths to match. TI sure makes it look easy, almost as if it's their job to do so. A...


Polyphase Filters

Started by commsignal in comp.dsp7 years ago 4 replies

Hello all, Can someone explain to me the following in Fred Harris' book Multirate Signal Processing for Communication Systems? 1. Chapter 6...

Hello all, Can someone explain to me the following in Fred Harris' book Multirate Signal Processing for Communication Systems? 1. Chapter 6 (page 143 - Separating the Aliases): The inputs to the M-path filter are not narrowband, and phase shift alone is insufficient to effect the destructive cancellation over the full bandwidth of the undesired special contributions (and the next sentence re...


Polyphase filter outputs

Started by nikolatesla20 in comp.dsp10 years ago 10 replies

Hi, I am almost a complete newbie when it comes to DSP. I've been studying various internet articles a lot over the last week but I still have...

Hi, I am almost a complete newbie when it comes to DSP. I've been studying various internet articles a lot over the last week but I still have some questions that I don't understand. What I'm trying to accomplish is to resample an audio signal to change it's pitch. I don't care if the length changes, I just want to get pitch changes. Since some pitch changes may be somewhat small (notes) I n...


Sampling rate conversion

Started by Praveen in comp.dsp14 years ago 9 replies

Hello, I am designing a sample rate converter from 8khz to 8.4 khz, which gives an upsampling rate of 21 and downsampling rate of 20. I...

Hello, I am designing a sample rate converter from 8khz to 8.4 khz, which gives an upsampling rate of 21 and downsampling rate of 20. I am implementing it using polyphase filters. I have got a filter with cut-off frequency of 21. This means I have 21 filters each of them shifted and upsampled by 21. And finally I cascade the output and downsample by 20. is this the right way of doing ...


combined arbitrary resampling and match filtering for timing recovery

Started by Awan in comp.dsp10 years ago 1 reply

Hi DSP Guys, Can you please help on polyphase matched filter which is doing arbitrary resampling and timing recovery at the same time...

Hi DSP Guys, Can you please help on polyphase matched filter which is doing arbitrary resampling and timing recovery at the same time ? regards Awan


FIR interpolator

Started by Vladimir Vassilevsky in comp.dsp11 years ago 21 replies

Recently I run into the problem with a basic task: design of a polyphase set of filters for the interpolation of a signal. The input is the...

Recently I run into the problem with a basic task: design of a polyphase set of filters for the interpolation of a signal. The input is the sampled signal; the output should be the interpolated values spaced at 1/10 of the sample. So, I designed the LPF at 10 x sample rate by the Parks-McClellan algorithm, and then decimated it into 10 subsets of the coefficients. But, although the...


SNR estimation

Started by Tom Derham in comp.dsp14 years ago 1 reply

I need to estimate the SNR (or, even better, the actual signal and noise powers separately) at the output of a matched filter. I have seen...

I need to estimate the SNR (or, even better, the actual signal and noise powers separately) at the output of a matched filter. I have seen various implementations for communications systems using maximum likelihood or based on particular types of signal coding. However my application is a radar type one - the signal transmitted is known (e.g. chirp or polyphase pulse), but after the matche...


Multirate filters

Started by maverick.gvs in comp.dsp11 years ago 1 reply

hi all, I am working on polyphase filter bank design. I want some pointers on (i)how to avoid aliasing from other bands? (ii)what are the...

hi all, I am working on polyphase filter bank design. I want some pointers on (i)how to avoid aliasing from other bands? (ii)what are the important steps while considering the design of prototype filter? (iii)How to avoid the amplitude distortion? (iv)How to design filter banks with different passband widths? thanks in advance vanamali


FMDemod, Phase noise and Polyphase

Started by Yip in comp.dsp13 years ago 10 replies

Hi All, I am trying to debug a FM demodulation noise problem for a FPGA prototype. The output in general looks ok, but the THD+N of a clean...

Hi All, I am trying to debug a FM demodulation noise problem for a FPGA prototype. The output in general looks ok, but the THD+N of a clean 60% modulated signal is only in the mid-40's dB (noticable even by ear), and that is mainly due to a high noise floor. The FM architecture is using the differentiator approach dAtan(Q/I)/dt with [1 -1] as the differentiator. (Phase wrap-around at +/-p...


Parallel Decimation Filter Implementation?

Started by Phil in comp.dsp12 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...


Question on channelizer

Started by gongdori in comp.dsp7 years ago 11 replies

Hello, I'm a newbie in DSP world and got several questions on channelizer. I recently studied polyphase channelizer and weighted...

Hello, I'm a newbie in DSP world and got several questions on channelizer. I recently studied polyphase channelizer and weighted Overlap-add(WOLA) filter bank. It seems that the only benefit WOLA has is that the number of channels can be not related to the decimation factor. Why is it so beneficial? When each channel is moved to baseband, isn't it desirable to have the minimal sampling freque...


testing a digital filterbank

Started by rameshk in comp.dsp9 years ago 2 replies

Hi All, I tried browsing topics on filterbanks on this site so as not to repeat my question. But haven't found any posts that is similar to my...

Hi All, I tried browsing topics on filterbanks on this site so as not to repeat my question. But haven't found any posts that is similar to my problem. I have a 16-channel digital filterbank based on polyphase decomposition. My objective is to find the absolute delay a signal undergoes in one of these filters. To test this, a 10-us square pulse was amplitude modulated by a 50 MHz carrier...


Re: I'm confused ;) was[Re: ADC limitations for bandpass/IF sampling]

Started by Bhaskar Thiagarajan in comp.dsp14 years ago 2 replies

"Joerg" wrote in message news:qJ_nf.33643$q%.17135@newssvr12.news.prodigy.com... > Hello Steve, > > > This form of polyphase sampling...

"Joerg" wrote in message news:qJ_nf.33643$q%.17135@newssvr12.news.prodigy.com... > Hello Steve, > > > This form of polyphase sampling is much loved by those of a masochistic > > tendency, for the pain and misery it can cause. :-) It is a nightmare to > > get multiple ADCs like this to track over time and temperatures. > > > > Nah, there are situatio


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...


Halfband filter True or false?

Started by jfgagne in comp.dsp14 years ago 25 replies

Hello everyone, I would like to confirm that even with a large number of coefficients it is not possible to make a Halfband filter with narrow...

Hello everyone, I would like to confirm that even with a large number of coefficients it is not possible to make a Halfband filter with narrow transition bandwidth? Ex.: Fpass=7MHz(-1dB tolerated), Fstop=8.125MHz with Astop=80dB using Fsampling = 32.5 MHz I tried a few software tools and I couldn't calculate this filter. It seems to me that I have to make it using a polyphase FIR. My go...


FIR filtering using a lookup table

Started by Anonymous in comp.dsp16 years ago 5 replies

Hi, Currently I am using a polyphase FIR filter to perform an interpolation. To gain performance I have in mind to redesign the original...

Hi, Currently I am using a polyphase FIR filter to perform an interpolation. To gain performance I have in mind to redesign the original filter with the principle of a lookup table. Does anyone know where I can find some helpful information of how to implement this principle? Regards, Ellegaard