Hi y'all, I'm working on developing a TDOA system with a desired time resolution of 1ns (~1ft). However, the hardware I'm using restricts me...

Hi y'all, I'm working on developing a TDOA system with a desired time resolution of 1ns (~1ft). However, the hardware I'm using restricts me to 1-2MSps for each signal being correlated. Typical correlation signals are in the 430MHz band and occupy a bandwidth in the ballpark of 128kHz. The problem I've encountered is that as much as I can simulate and successfully correlate delayed signals i...

Hi i am simulating a baseband OFDM system Digital down converter. I have a problem regarding decimation and interpolation. In the last stage of...

Hi i am simulating a baseband OFDM system Digital down converter. I have a problem regarding decimation and interpolation. In the last stage of the DDC, i am doing low pass filtering of the data. My problem is when i make a sharp transition filter i.e. ( Fpass = 250e3; % Passband Frequency Fstop = 300e3; % Stopband Frequency the OFDM constellation is scattered. But...

Eric Jacobsen wrote: > > On Tue, 01 Apr 2008 19:37:07 -0600, jim > wrote: > > > > > > > Eric Jacobsen wrote: > > > >...

Eric Jacobsen wrote: > > On Tue, 01 Apr 2008 19:37:07 -0600, jim > wrote: > > > > > > > Eric Jacobsen wrote: > > > > > > If the process requires an AA filter to reduce bandwidth they > > > > "necessarily" will. > > > > > > Back to my previous example with a solitary tone in a unity gain > > > passband. If the decimation rate is integer and the samples are no

Hello all I have a complex transfer function e.g. like this: Y=[0.1+0.2i, 0.3+0.77i, 0.11+0.22i] corresponding to a frequency vector e.g....

Hello all I have a complex transfer function e.g. like this: Y=[0.1+0.2i, 0.3+0.77i, 0.11+0.22i] corresponding to a frequency vector e.g. X = [1, 3, 5] Now I want to have the frequencies between the points of my oryginal vector e.g. X1=[1.2, 2.5, 4.2, 4.8, 4.9] And I want to find a corresponding vector Y1 of complex values. The question is whether I can perform an interpolati...

Ron N. ha scritto: > Phase is meaningless without a defined reference point. > Therefore your phase is likely not wrong, just referenced...

Ron N. ha scritto: > Phase is meaningless without a defined reference point. > Therefore your phase is likely not wrong, just referenced to > a different location or with a different polarity than you expect. > > I usually flip the sign of alternating fft results to move the 0 > phase reference to the center of my fft aperature. This way, > any phase interpolation isn't done with refer

Can anyone tell me what were the improvements in the MELPe's synthesis vs. MELP's? (they are different, i.e. different parameter interpolation...

Can anyone tell me what were the improvements in the MELPe's synthesis vs. MELP's? (they are different, i.e. different parameter interpolation and synthesis. But what is the key advantage/improvement?) How does that improve quality?

Hi! I am trying to implement a Fractional Delay Interpolator using a Farrow Structure as described by Gardner in the article...

Hi! I am trying to implement a Fractional Delay Interpolator using a Farrow Structure as described by Gardner in the article "Interpolation in Digital Modems -Parts I/II ". The basic underlaying equation is : y(kTi) = SUM(over m) {x[mTs].h[kTi-mTs]} ---- EQ (1) Where Ts is the input sample time and Ti is output sample time, both incommensurate to each other. The author then defines ...

hello, i'm trying to program a wavetable-based synth in c++. I have some waveforms and different envelopes for them and want to switch...

hello, i'm trying to program a wavetable-based synth in c++. I have some waveforms and different envelopes for them and want to switch between settings while playing. I use a 4096 samples buffer. My question is, how do i now implement anti-aliasing? I thought of multiplying envelope & waveform, and then lowpass and the resample with linear interpolation at a lower sample rate. Is it right, tha...

Dear All, Recently I had posted a thread where I asked about SNR penalty due to linear interpolation. I managed to solve that problem but there...

Dear All, Recently I had posted a thread where I asked about SNR penalty due to linear interpolation. I managed to solve that problem but there are some more doubts. My receiver has adaptive doppler correction and adaptive DFE using LMS. Now, the filter update constants that I had kept intially are as follows: Training mode: mu1=0.01 -> for Feedforward filter mu2=0.003 -> for

Hi...the working of bandpass signal is a bit confusing for me... for example... for a given raw signal,complex,nonzero over Wa= 10khz and...

Hi...the working of bandpass signal is a bit confusing for me... for example... for a given raw signal,complex,nonzero over Wa= 10khz and Wb=15khz and given sampled digital signal x(n)=Xa(nTs) Need your help to deterine the min fs such that raw signal xa(t)can be recovered from its sample x(n). AFter that need to determine the interpolation formula for xa(t) for this min fs.

The example in the upfirdn() - link below - is as follows. L = 147; M = 160; % Interpolation/decimation factors. N = 24*M; h =...

The example in the upfirdn() - link below - is as follows. L = 147; M = 160; % Interpolation/decimation factors. N = 24*M; h = fir1(N,1/M,kaiser(N+1,7.8562)); h = L*h; % Passband gain = L Fs = 48e3; % Original sampling frequency-48kHz n = 0:10239; % 10240 samples, 0.213 seconds long x = sin(2*pi*1e3/Fs*n); % Original signal, sinusoid @ 1kHz y = upfirdn(x,h,L,M); % 9408 samples, still...

Hello comp.dsp readers, I am working on a plugin that involves estimation of spectral peaks (frequency and magnitude - basically the whole...

Hello comp.dsp readers, I am working on a plugin that involves estimation of spectral peaks (frequency and magnitude - basically the whole thing is phase-vocoder based). I first tried to use parabolic interpolation (by taking the maximum of the parabola to calculate the frequency and amplitude of sinusoids I want to track). In most cases that works sufficiently well, but when sinusoi...

To promote some interesting discussion I thought I would raise the following question: Possibly as part of a sequence of processing, to...

To promote some interesting discussion I thought I would raise the following question: Possibly as part of a sequence of processing, to interpolate a sampled sequence by 2 is it correct to keep the original samples and derive the in-between samples, or replace the original samples in the process of interpolating the signal. Dirk Bell DSP Consultant

Would someone be kind enough to check the argument below. I'm a bit uncomfortable with the conclusion. The equations are in LaTeX format. Some...

Would someone be kind enough to check the argument below. I'm a bit uncomfortable with the conclusion. The equations are in LaTeX format. Some signal, f(t), is bandlimited such that we can represent it with samples every T seconds. f(t) = \sum_{n=-\infty}^{\infty} f(nT) sinc(\frac{\pi t}{T} - nT) The linearity property of the Fourier transform gives that F(w) = \sum_{n=-\infty}^{\inft...

I am planning to use a Lagrange interpolator in the context of fractional delay filter to change the sampling rate by a factor between 1 and 1.5...

I am planning to use a Lagrange interpolator in the context of fractional delay filter to change the sampling rate by a factor between 1 and 1.5 (lower). My signal bandwidth is about 0.23 when sampling frequency fs is normalized to 1. The signal has 12 bits resolution and I would like to maintain this resolution at the output of the interpolator. How can I determine the degree of the Lag...

Here is the code, two stages differetiator, and two stage integrator. It's very common but how do I plot the frequency response? TIA! d =...

Here is the code, two stages differetiator, and two stage integrator. It's very common but how do I plot the frequency response? TIA! d = cos(2*pi*(1:1e3)/1e2); e = zeros(length(d)*10, 1); db = [1 0 -1]; da = [1]; c0 = filter(db, da, d); c1 = filter(db, da, c0); ib = [1]; ia = [1 -1]; e(1:10:end) = c1; c2 = filter(ib, ia, e); c3 = filter(ib, ia, c2)/10/10; figure; ...

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...

Hi,every one. I read some matlab program (Voicebox - estnoisem.m)and inverse the line to be interloation logo. But I can not tell which kinds...

Hi,every one. I read some matlab program (Voicebox - estnoisem.m)and inverse the line to be interloation logo. But I can not tell which kinds kind of interloation it is. y = xi - (xi*xj)/(x-xj) * (yi-yj)/(xi-xj) Whick kinds of interpolation is it? Anyone can tell it? Thank you. The refered matlab line is as follows. % What is the priciple algo? Where it is described? m=dmh(i,2)+(qi*qj...

I need advice on resampling. I sampled a signal at 512*50Hz. Then by using numerical differntion with lagrange interpolation, i found the...

I need advice on resampling. I sampled a signal at 512*50Hz. Then by using numerical differntion with lagrange interpolation, i found the frequency of the fundamental. I need to resample the siganl at a frequency of 512*fs Hz where fs is the off nominal frequency of the fundamental (usually slightly greater than 50Hz). how do I go about implementing it in code?

Hello, I've been studying polyphase decomposition, decimation, interpolation and fractional decimation. I think I have a handle on what I'm...

Hello, I've been studying polyphase decomposition, decimation, interpolation and fractional decimation. I think I have a handle on what I'm doing, but there's one thing I don't understand... Do you have to have greater than an L*Mth order filter in order to use polyphase decomposition to put the compressor and expander in their most efficient places (first and last, respectively)? For exa...