## Sampling frequency correction by Interpolation/Decimation technique?

Started by in comp.dsp12 years ago

Dear all, In OFDM system, we difine T is the sampling period at the output of IFFT (at transmiter), T' is sampling period at receiver,...

Dear all, In OFDM system, we difine T is the sampling period at the output of IFFT (at transmiter), T' is sampling period at receiver, SFO=(T'-T)/T At receiver, after CP removal, the (m,n)-th sample of the time-domain received signal is r'(m,n),(include SFO). How can we use Interpolation/decimation technique to correct the r'(m,n) (to get desied signal r(m.n) without SFO) Who can help m...

## Question about Sigma-Delta conversion/decimation

Started by in comp.dsp12 years ago 1 reply

Hi all, My question is related to sigma-delta conversion. I understand the analog side of this, i.e. the modulator, and have developed a...

Hi all, My question is related to sigma-delta conversion. I understand the analog side of this, i.e. the modulator, and have developed a matlab simulation which gives me 1's and -1's output for any input signal. Essentially, this is the high data rate 1 bit output of the sigma delta modulator. Now, I need help understanding the decimation process. Conceptually, I understand what decima...

## ADI decimation function

Started by in comp.dsp11 years ago

Been looking at the ADI library, specifically fir_decima FIR-based decimation filter Synopsis #include float *fir_decima (const...

Been looking at the ADI library, specifically fir_decima FIR-based decimation filter Synopsis #include float *fir_decima (const floatinput[], floatoutput[], const float pmcoefficients[], floatdelay[], intnum_output_samples, intnum_coeffs, intdecimation_index); Description The fir_decima function implements a finite impulse response (FIR) filter defined by the coefficien

## Testing Digital Filter

Started by in comp.dsp14 years ago 2 replies

Hello, I am using Linear feedback shift register(LFSR) to test Digital filter(DF) because LFSR generate white noise and has a flat...

Hello, I am using Linear feedback shift register(LFSR) to test Digital filter(DF) because LFSR generate white noise and has a flat like spectrum. Can anynody suggest when I pass white noise in DF, on what citeria should I judge the filter (Device Under Test) is pass or fail ?? my filter is 3rd order polyphase comb decimation filter with decimation factor of 256. ...

## Order of cascade in IFIR filter

Started by in comp.dsp4 years ago 6 replies

I'm trying to reduce the circuit size of a multi-channel FIR filter used prior to decimation. The multi-channel structure already makes use of...

I'm trying to reduce the circuit size of a multi-channel FIR filter used prior to decimation. The multi-channel structure already makes use of the decimation factor to reduce the overall circuit size - only 1/DecimationFactor outputs for each channel are actually calculated. I want to replace it with an IFIR filter to reduce the circuit size even further, and playing around wi

## Digital Filter testing with LFSR

Started by in comp.dsp14 years ago 1 reply

Hi guys, It is a known fact that LFSR generate flat spectrum, I choosed LFSR to test my 3rd order polyphase comb decimation...

Hi guys, It is a known fact that LFSR generate flat spectrum, I choosed LFSR to test my 3rd order polyphase comb decimation filter with decimation factor of 256. I am wondering what would be the lenght of LFSR ? it should be 4 bit or 8 bit or 10 bit ?? All generate flat spectrum. What characteristics help me in choosing the lenght of LFSR ??. I have read few technical...

## Help me to understant sampling theory, decimation by integer factor

Started by in comp.dsp11 years ago 7 replies

Hi, everyone! I read "decimation by integer" and "interpolation by integer factors" in a DSP book. I'm just wondering why integer...

Hi, everyone! I read "decimation by integer" and "interpolation by integer factors" in a DSP book. I'm just wondering why integer factor to downsampling... Sometimes, to resample signal by rational factor, people use conversion oversampling(integer factor) -> low-passfilter -> downsampling(also integer factor). (This means "Sampling rate conversion by non-integer factors") Howe

## Digiatal Sampling correction Interpolation/Decimation for application in flexible OFDM Modem.

Started by in comp.dsp12 years ago 3 replies

Hi all, My project is Digiatal Sampling correction Interpolation/Decimation for application in flexible OFDM Modem. I'm implementing in Matlab...

Hi all, My project is Digiatal Sampling correction Interpolation/Decimation for application in flexible OFDM Modem. I'm implementing in Matlab the 5_taps optimized interpolator which will improve the performance for long data OFDM bursts. I used control parameters basepoint index (m(k)) and fractional interval (mu(k)) (reference to FM Gardners article "Interpolation In Digital Modems---PART...

## Decimation filter: Output magnitude response

Started by in comp.dsp9 years ago 4 replies

I have a decimation filter which decimates by 100. Input sampling frequency fclk1 = 1.6MHz, Output Sampling freq flck2 = 16kHz. I am...

I have a decimation filter which decimates by 100. Input sampling frequency fclk1 = 1.6MHz, Output Sampling freq flck2 = 16kHz. I am trying to plot the input and output frequency responses. For the input, I use fft function in matlab, and get the magnitudes of my frequency response. To generate the plot X-axis points (i.e., frequencies) I do the following freq = (flck1/fftPtsIn)*(...

## Mixed-Radix 2048 point FFT algorithm

Started by in comp.dsp12 years ago

Hi all! For the 2048 point FFT implementation I have used two 1024 point Radix-4 (Decimation-in-frequency) FFT and for final stage I have used...

Hi all! For the 2048 point FFT implementation I have used two 1024 point Radix-4 (Decimation-in-frequency) FFT and for final stage I have used 2048 point (Decimation-in-time) Radix-2 FFT. But outputs are incorrect. When input is a pulse at x[i] = {1 when i = 1, 0 otherwise} the real part of output is cosine with frequency = 2, since I have expected the cosine with frequency = 1. Likewise fo...

## Filterbanks Basics

Started by in comp.dsp13 years ago 2 replies

Hi All, Could any one please explian the theory behind decimation and interpolation?. Thanks. Thomas.

Hi All, Could any one please explian the theory behind decimation and interpolation?. Thanks. Thomas.

## decimation in time - butterfly

Started by in comp.dsp9 years ago 1 reply

is there a method to memorise the butterfly? If anyone has any tricks? or how does the butterfly diagram come about?? need it for a exam

is there a method to memorise the butterfly? If anyone has any tricks? or how does the butterfly diagram come about?? need it for a exam

## frequency response of cic

Started by in comp.dsp11 years ago 5 replies

Not clearly, why instead of four petals one is drawn only R = 8; % Decimation factor M = 1; % Differential delay N = 9; % Number of...

Not clearly, why instead of four petals one is drawn only R = 8; % Decimation factor M = 1; % Differential delay N = 9; % Number of stages f=0:0.001:0.5; fcic=(abs(sin(pi*M*f)./(eps+sin(pi*f/R)))).^N; plot(fcic)

## OT: Anthropomorphism

Started by in comp.dsp14 years ago 25 replies

There is a lot of anthropomorhpising* which goes on in the discussion group, for example "The DFT assumes N samples at the input" or...

There is a lot of anthropomorhpising* which goes on in the discussion group, for example "The DFT assumes N samples at the input" or "the decimation filter ensures that alias frequencies cannot be seen by the downsampler". I was just wondering if there is an opposite word to anthropomorhism, whereby inanimate characteristics are contributed to humans, eg. "That lecturer has a very low SNR...

## FFT question

Started by in comp.dsp11 years ago 3 replies

I am looking at "Understanding Digital Signal Processing" Second Edition page 137- figure 4-5. It is a full decimation in time butterfly...

I am looking at "Understanding Digital Signal Processing" Second Edition page 137- figure 4-5. It is a full decimation in time butterfly diagram for an 8 point FFT (It seems like everyone here has this book as near as I can tell - ha ha) ------------------------------------------- Here are my thoughts: I know that for a sequence of real numbers that the DFT coefficients: X(7), X(6...

## parallel decimation?

Started by in comp.dsp13 years ago 4 replies

Hello all, I'm writing an application in which I need to decimate the input (44100 samples per second) signal by a factor of 4 and 16. I'd...

Hello all, I'm writing an application in which I need to decimate the input (44100 samples per second) signal by a factor of 4 and 16. I'd like to do this as efficiently as possible, but I'd also like to keep the delay as low as possible - particularly for the divide by 16 branch. If I didn't care about the delay I'd obviously cascade two divide by four decimators to give my two output r...

## New to DSP, need to understand decimation

Started by in comp.dsp13 years ago 5 replies

Hello. I am new to the DSP area. So.. I am just going to throw all my questions out there so you can see how lost I am... and maybe have...

Hello. I am new to the DSP area. So.. I am just going to throw all my questions out there so you can see how lost I am... and maybe have yourself a chuckle :-) I realize my questions are in different areas but I wanted to put it all together. Any help would be greatly appreciated and would point me in the right direction. The problem (at it's minimum): I receive 2 channels bandpass ...

## Which processor for decimation?

Started by in comp.dsp13 years ago 4 replies

Hi, i have to decimate my input data (16 bit resolution) from 10Khz to 10Hz using a 60db/dec or 80db/dec butterworth digital filter, i need to...

Hi, i have to decimate my input data (16 bit resolution) from 10Khz to 10Hz using a 60db/dec or 80db/dec butterworth digital filter, i need to know if this is possible using a standard MCU (for example 16 bit PIC) or is necessary to use a DSP core. Thanks in advance, Emiliano.

## Sigma-Delta A/D Converter

Started by in comp.dsp12 years ago 5 replies

Hi, In Oversampled Sigma-Delta A/D Converter, which is in the demo, why did they use three decimators (4*4*4) rather than one decimator...

Hi, In Oversampled Sigma-Delta A/D Converter, which is in the demo, why did they use three decimators (4*4*4) rather than one decimator of 64? Is there any particular theory behind this for the gradual process of decimation?

## Interpolation and decimation in DFT...

Started by in comp.dsp12 years ago 10 replies

Hi all, I have two blocks of samples of signals. x0, ..., x_{n-1}, and y0, ..., y_{n-1}, where DFT[x]= y. Suppose I want to...

Hi all, I have two blocks of samples of signals. x0, ..., x_{n-1}, and y0, ..., y_{n-1}, where DFT[x]= y. Suppose I want to interpolate {x} by a upsampling rate of 2, how to do that by operating on the {y} sequence, assuming {x} is not given so we can only modify {y}? How about decimating {x} by a downsampling rate of 2? Now, what if I want to interpolate {y} by a upsam...