Interpolation using polyphase FIR filter: Finding coeffisients

Started by MrAlfred in comp.dsp11 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...


FFT upsampling filter

Started by blueskull in comp.dsp4 years ago 19 replies

Hi, all: New member here. I did a search, but it seems there is no information related to this topic. What I am doing now is I am trying to...

Hi, all: New member here. I did a search, but it seems there is no information related to this topic. What I am doing now is I am trying to upsample some data points to a much higher sample rate, and to do that, I need a low pass filter after zero stuffing. There are 2 ways to reduce calculation complexity, AFAIK: a. To use a polyphase filter at input sample rate, or b. To use a FFT fil...


Problem with IFIR in SRC

Started by Govind in comp.dsp13 years ago 4 replies

Hello, I have implemented polyphase SRC code for ratio of 3/2 (8KHz 12KHz). I am using 396 tap filter, which gives 1584000 [ =...

Hello, I have implemented polyphase SRC code for ratio of 3/2 (8KHz 12KHz). I am using 396 tap filter, which gives 1584000 [ = (24000(Fs)*396)/(3*2)] MAC operations per second (MPS). Now I want to reduce the computations by making use of IFIR method. But I find that the implementation becomes too complicated to be put in a formula that can be implemented on Matlab. Can you please giv


Quadriphase barker sequences

Started by Anonymous in comp.dsp14 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


Relationship between FIR frequency response and each polyphase response

Started by gretzteam in comp.dsp3 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...


How to interpret polyphase coefficients generated in MATLAB

Started by vizziee in comp.dsp10 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 & wavetable playback?

Started by R Jones in comp.dsp14 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...


How to resample fast

Started by s036 in comp.dsp14 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


A question from Vaidyanathan's "Multirate Systems" book

Started by Rick Lyons in comp.dsp14 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 ...


Scilab, sparse matricies, and polyphase filtering

Started by Tim Wescott in comp.dsp10 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...


multirate filter design

Started by Anonymous in comp.dsp11 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...


polyphase, interpolation, timing error

Started by Anonymous in comp.dsp13 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.dsp12 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.dsp12 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 ...


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

Started by vijay in comp.dsp10 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


PQMF Filters

Started by coolup in comp.dsp13 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.


Help polyphase implementation

Started by bhaskar.nallani in comp.dsp9 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...


CIC versus Polyphase FIR

Started by sachinwannabe in comp.dsp11 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...


DFT\MDFT Filter Bank + Proccessing

Started by JacobG in comp.dsp8 years ago 6 replies

Greetings, I am dealing with the task of channelizing an input signal to M different channels (for instance 32 channels), so the different...

Greetings, I am dealing with the task of channelizing an input signal to M different channels (for instance 32 channels), so the different channels may be processed and synthesized back together. The prototype filter should bring a 40db stop band attenuation. I starting by simulating a uniform dft filter bank, where the input signal is commutated into the prototype filter polyphase filters...


Polyphase filter bank with nonuniform frequency bins and offsets

Started by bmcgee22 in comp.dsp11 years ago 2 replies

I am trying to design a configurable analysis/synthesis filter bank with nonuniform bins and offsets. My subband sizes are fixed at Fs/32,...

I am trying to design a configurable analysis/synthesis filter bank with nonuniform bins and offsets. My subband sizes are fixed at Fs/32, Fs/64, Fs/128 and Fs/256 the bandwidth. I need to be able to support any combination of these subbands starting at any Fs/256 offset. My original approach was a two stage approach that divided the bandwidth into 32 subbands, then split each of those into 1, ...