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Channel Correlatoin Function & Noise Varaince

Started by PhyLyrDude in comp.dsp15 years ago

In OFDM Channel Estimation, we firstly get a raw channel estimate from the pilots and then perform interpolation for the remaining subcarriers. ...

In OFDM Channel Estimation, we firstly get a raw channel estimate from the pilots and then perform interpolation for the remaining subcarriers. A popular cited method is the 1D Wiener method which basically looks for a set of MMSE weights. The formula requires the correlation of the channel in the each subcarrier and the noise variance. Most text will assume that this is known. Being new to th...


Frequency interpolation

Started by balaji in comp.dsp20 years ago 9 replies

Hai, I am in need of a frequency interpolator algorithm, which can find the arbitrary points "beteween the frequency bins". When I was...

Hai, I am in need of a frequency interpolator algorithm, which can find the arbitrary points "beteween the frequency bins". When I was going through the discussions in this group, I found few mails related to that but many of them are related to finding a particular peak frequency in the Spectrum. But my interest is representing a "N-point" spectrum as a "M-point" spectrum so that when t...


FFT re-scaling or interpolation

Started by Evan Olcott in comp.dsp19 years ago 11 replies

Hello everyone. Have an interesting problem. I have an FFT of an audio signal and I want to interpolate the bins into another frequency...

Hello everyone. Have an interesting problem. I have an FFT of an audio signal and I want to interpolate the bins into another frequency set... instead of a linear set of frequencies, I'd like the frequencies I get (in the end) to be based on an exponential/logarithmic formula, based on the twelfth root of 2... What I'd like to do is take the FFT results and re-interpolate (or re-sc...


Interpolation

Started by Anshu in comp.dsp17 years ago 6 replies

Hi I want to interpolate a complex signal given at points x1,x2,x3....xn to get signal at points x1+a,x2+a....xn+a. 1) To interpolate...

Hi I want to interpolate a complex signal given at points x1,x2,x3....xn to get signal at points x1+a,x2+a....xn+a. 1) To interpolate complex signal, I am interpolating the magnitude and phase separately and later combining them. Is this an appropriate method? 2) I am trying to get FIR filter coefficients using Spline method. The equation that i am using is: y(i+a)= ((a^3)/6)*(...


Beginning Interpolation

Started by Stacy in comp.dsp17 years ago 9 replies

Given: x(n) which is a sequence sampled at Fs1 = 8000 samples/sec. Find: Interpolate x(n) up to Fs2 = 48k samples/sec. L = 6. ...

Given: x(n) which is a sequence sampled at Fs1 = 8000 samples/sec. Find: Interpolate x(n) up to Fs2 = 48k samples/sec. L = 6. Solution? 1) zero pack x(n) by placing L-1 = 5 zeros after every sample. 2) design anti-imaging filter. Using Matlabs fir1, b = fir1(N, Fs) 3) Run an FIR filter on x(n*L) using b. Question: Based on fir1(N, Fs), i) what is the recomm...


matched filters and interpolation

Started by Muzaffer Kal in comp.dsp20 years ago 1 reply

Hi, I'm back on a project which I've been working on at times. The piece I am working on right now is digital clock recovery which works ok...

Hi, I'm back on a project which I've been working on at times. The piece I am working on right now is digital clock recovery which works ok but I'd like to see if I can improve the performance. The current implementation is a feed-forward timing detector and an interpolator for data recovery with a farrow type interpolator. The piece I am trying to add is a matched filter. I only have two sa...


Sine Wave autocorrelation, interpolation of phase

Started by eduardoG26 in comp.dsp7 years ago 11 replies

Hallo. For the purpose of measuring complex impedances i need to compare the phase of two copies of a sinewave over a number of periods. The...

Hallo. For the purpose of measuring complex impedances i need to compare the phase of two copies of a sinewave over a number of periods. The sinewave is generated in the same µC. Frequency is known and stable. The original and the shifted signals are sampled by a double synchr. ADC. The phase shift is the base for calculation of the complex Z of a load. I have implemented an (auto-)correlatio...


Re: Interpolation

Started by Eric Jacobsen in comp.dsp16 years ago

On Tue, 01 Apr 2008 14:41:06 -0600, jim wrote: > Eric Jacobsen wrote: > > The more fundamental issue as I understand it, is whether...

On Tue, 01 Apr 2008 14:41:06 -0600, jim wrote: > Eric Jacobsen wrote: > > The more fundamental issue as I understand it, is whether or not a > > filter would change the original, uninterpolated input samples in the > > output, i.e., are the interpolated samples distinguishable from the > > uninterpolated. In a "decimating" filter the issue that an > > anti-alias


FIR Reduction Routine C64x from TI?

Started by No Never in comp.dsp18 years ago

Dear all, Is there an assembly or linear assembly routine for a FIR reduction or interpolation filter for C64x from TI ? Thanks for...

Dear all, Is there an assembly or linear assembly routine for a FIR reduction or interpolation filter for C64x from TI ? Thanks for any hints, Wolfgang


High pass polyphase filter

Started by Piergiorgio Sartor in comp.dsp15 years ago 4 replies

Hi all, I'm trying to design an high pass polyphase filter. The approach is to take the 5 tap high pass FIR, oversample it by a factor N...

Hi all, I'm trying to design an high pass polyphase filter. The approach is to take the 5 tap high pass FIR, oversample it by a factor N (usually 128 or 256), then subsample it again into the different phases. The oversampling was done by polynomial interpolation and by zero-padding and low-pass. I mean, the two methods were tried, not altogether. In both cases, the results are dis...


Delay by Less than a sampling interval

Started by SammySmith in comp.dsp15 years ago 16 replies

Hi all, Is it possible to delay digital data, by a fraction of the sampling interval. i.e. if fs=1/Ts, where fs is the sampling frequency and...

Hi all, Is it possible to delay digital data, by a fraction of the sampling interval. i.e. if fs=1/Ts, where fs is the sampling frequency and Ts the sampling interval. My understanding is that it can be done with interpolation, but that would require a higher clock. Is it possible without using a higer clock? Regards, Sam


Please help me in understanding "Decimation filter"

Started by hswnetin in comp.dsp20 years ago 17 replies

Hi All, I thank all of you, for your support, which I got previously. I am hardware design engineer and working on sigma delta ADC. And my...

Hi All, I thank all of you, for your support, which I got previously. I am hardware design engineer and working on sigma delta ADC. And my role is to design decimation filter, basically I have to design a blocks which converts single bit to multi bits output. I surveyed lots of literature. I referred "An Economical Class of Digital Filters for Decimation and Interpolation" by Eugine B H...


Timing recovery by oversampling in QAM systems

Started by Calabi_yau in comp.dsp17 years ago 1 reply

Hi, A basic question about QAM. I've noticed that there are timing recovery systems that calculate the 'correct' sample by means of an...

Hi, A basic question about QAM. I've noticed that there are timing recovery systems that calculate the 'correct' sample by means of an interpolation guessing wich one of this intermediate samples is the most suitable instead of closing a loop and recovering and using the recovered clock as a receiver clock.. It seems to me quite interesting but I have a doubt: If we don't synchronize the...


Fir Interpolation question for dummy

Started by Jack1962 in comp.dsp13 years ago 8 replies

Hi everyone. I have a signal sampled at 100Hz for example: [1 2 3 4 3 2 1] I have to interpolate to 200Hz to send a DAC, so I adding zero...

Hi everyone. I have a signal sampled at 100Hz for example: [1 2 3 4 3 2 1] I have to interpolate to 200Hz to send a DAC, so I adding zero between the samples in this way: [1 0 2 0 3 0 4 0 5 0 4 0 3 0 2 0 1]. Now I want to filter with a low pass Fir filter to have a passband frequency of 20Hz. I am using Scopefir, my question is, I have to use 100Hz or 200Hz as sampling frequency in the d...


Sampling Frequency effect on OFDM

Started by OFDMnewuser in comp.dsp10 years ago 1 reply

Hi, I am doing a baseband simulation for IEEE 802.11 g. The standard uses 64 IFFT length (3.2?s) and 20 Msamples per second. I think this result...

Hi, I am doing a baseband simulation for IEEE 802.11 g. The standard uses 64 IFFT length (3.2?s) and 20 Msamples per second. I think this result in one sample per symbol. I am wondering what the structure will be if I use 40 Msample per second, i.e will the IFFT size change, or interpolation will be used? or what is the effect of sampling frequency on the baseband simulation? Best Regards,


Resampling algorithm for limited hardware

Started by bitrex in comp.dsp9 years ago 1 reply

Suppose I have a waveform stored in flash memory on a fairly limited microprocessor, say an AVR 8 bit or something, and I'm sending it out to...

Suppose I have a waveform stored in flash memory on a fairly limited microprocessor, say an AVR 8 bit or something, and I'm sending it out to analog via one of the high-speed PWM channels. Is there a interpolation algorithm of some type that will allow bandlimited frequency transposition of this waveform in real time over a significant range on such limited hardware? Links to any examp...


Image/video processing/enhancement tool

Started by Image in comp.dsp16 years ago 5 replies

Is there a free tool for Matlab based (or other) for doing image/video enhancement: 1. Zooming in using various interpolation (bi-cubic...

Is there a free tool for Matlab based (or other) for doing image/video enhancement: 1. Zooming in using various interpolation (bi-cubic etc.), 2. image enhancement using spatial correlation, 3. image enhancement using succeeding images series. Thanks, IM


Fractional Resampling WCDMA gives noise

Started by DSPWirelessGuy in comp.dsp13 years ago 15 replies

Dear all, I will be grateful if I can get some light on this. I have Tx Data upsampled from 3.84MHz to 12 MHz (@3.125X. On the Tx side I...

Dear all, I will be grateful if I can get some light on this. I have Tx Data upsampled from 3.84MHz to 12 MHz (@3.125X. On the Tx side I use a nice RRC filter to perform interpolation and upsample using Matlab inbuild polyphase filter to generate Test vectors. Problem is , on the receiver side, I do not use polyphase filter. I use RRc filter at the begining then downsample using 3,3,3...


easy start of interpolation question.

Started by Roderik Emmerink in comp.dsp21 years ago 1 reply

Hi, I want to determine if a next found point is near to the previous found point and so determine of I should count it or not. In my...

Hi, I want to determine if a next found point is near to the previous found point and so determine of I should count it or not. In my application it often happens that their are wrong points in my signal. When I have the first points when measuring I can determine where the next point could be. So, my problem is, how to determine the first points (when there is no history yet). I hope some...


LS channel estimation for LTE

Started by Sri2424 in comp.dsp16 years ago

Hi Pals, I am developing a least square channel estimation in matlab. I will brief u what i am doing. Let me know if i am doing anything...

Hi Pals, I am developing a least square channel estimation in matlab. I will brief u what i am doing. Let me know if i am doing anything wrong. At the receiver, the frequency domain channel estimation is done by performing a simple division operation i.e., received pilot by transmitted pilot. Once the frequency domain estimates are found we will perform FD interpolation. we are using ...