Real Time Interpolation/Resampling

Started by FatScouser in comp.dsp10 years ago 5 replies

Hi, I have a motion control system comprising a realtime component and a non-realtime component. Simple 2D data (position/time) is being...

Hi, I have a motion control system comprising a realtime component and a non-realtime component. Simple 2D data (position/time) is being streamed from the non-realtime area into a FIFO queue in the realtime area. I have a few things that I'm hoping someone could help me with: The realtime component needs to keep running at all cost. Thus, when data is delayed, my buffer is in danger of runni...


Interpolation matrix

Started by Dave in comp.dsp11 years ago 12 replies

I'm trying to define a couple of common signal interpolation/ resampling methods as a matrix operation, i.e. y = W z where, y is the...

I'm trying to define a couple of common signal interpolation/ resampling methods as a matrix operation, i.e. y = W z where, y is the interpolated data vector of dimension m x 1 W is the interpolation matrix of dimension m x l z is the original data vector of dimension l x 1 I've worked out W for linear and nearest neighbour interpolation (easy) and I was wondering if more c...


Resampling in stages?

Started by Anonymous in comp.dsp7 years ago 7 replies

Is there a reason (founded in DSP theory) to resample in stages instead of going directly from one sample rate to another? If so what is the...

Is there a reason (founded in DSP theory) to resample in stages instead of going directly from one sample rate to another? If so what is the reason? And where can I read about it? For example: When I resample from 8kHz to 48kHz the resulting audio sounds worse compared to the audio I get if I resample in stages from 8kHz to 24kHz to 48kHz. Thanks


Live audio resampling problem!

Started by in comp.dsp12 years ago 3 replies

I am trying to decode audio streams sent from an arm board, unfortunely the clock on the arm board can vary from time to time, as a result the...

I am trying to decode audio streams sent from an arm board, unfortunely the clock on the arm board can vary from time to time, as a result the wave sample rate varies. To keep high audio quality, the delay must be kept constant. Can anyone give me some advice? Do I have to design a resample algorithm whose conversion rate can be changed from time to time?


Resampling of irregularly sampled data in 3 dimensions using a convolution kernel (Lanczos)

Started by Andy365 in comp.dsp9 years ago 2 replies

Hello, I have irregular spaced samples in 3 dimensions: f(xi, yi, zi) that I wish to take the FFT of. Therefore I need to resample the data to...

Hello, I have irregular spaced samples in 3 dimensions: f(xi, yi, zi) that I wish to take the FFT of. Therefore I need to resample the data to a regular grid. Since the data will be Fourier transformed; it will also be good to smooth the data somewhat to avoid Gibbs noise ("ringing")? Therefore I am considering using a convolution kernel to resample the data to a regular grid. A convolutio...


Resampling by multistage decimation (Richard Lyons book)

Started by MRR in comp.dsp8 years ago 7 replies

Hello everybody, I am trying to understand the multistage decimation problem stated by Richard Lyons in his book "understanding digital signal...

Hello everybody, I am trying to understand the multistage decimation problem stated by Richard Lyons in his book "understanding digital signal processing" 1st edition, page 306. It is as follows: - We have an input data arriving at 400 kHz and we want to decimate by 100, but we are going to do this in two stage: 50 and 2. - The original signal bandwidth is something greater than 100 kHz, ...


some suggestions on my octave-band spectrogram analysis in fixed point DSP implementation

Started by Simon in comp.dsp13 years ago 6 replies

i am designing one 1/3 octave band spectrogram analysis I use multirate filter bank to realize that spectrogram. It goes this way 1) first,...

i am designing one 1/3 octave band spectrogram analysis I use multirate filter bank to realize that spectrogram. It goes this way 1) first, from the biggest frequency value, I use three IIR bandpass filter, then calculate the std value. 2) Then decimate the input by 2 through(one 30 order FIR filter, then resampling the input), then use the same three coffeicents to calculate the succ...


streaming resampled buffers seamlessly

Started by sammy davis jr. in comp.dsp16 years ago 3 replies

Hi guys, this problem is driving me nuts over here. I'm working on a streaming resampling method that resamples buffers in sequence, when...

Hi guys, this problem is driving me nuts over here. I'm working on a streaming resampling method that resamples buffers in sequence, when i reach the end of an input buffer i zeropad it (enough for the anti-alisaing filter), generate the output buffer, and i'm done. the problem is that those last zero-samples at the end of my input buffer skew the last two samples of my output buffe...


Video Port Capture. Interesting Questions.

Started by arichard in comp.dsp12 years ago

Dear All, I have some questions about the video Port: As described in "spru629" the video Port is capable among others of acquiring frames,...

Dear All, I have some questions about the video Port: As described in "spru629" the video Port is capable among others of acquiring frames, perform scaling and chrominance resampling. In the configurations of the video port, the threshold to perform a DMA transfer for the chrominance components is automatically defined as half of the threshold defined for the luminance component. My...


Setting sampling rate of Tektronix VSA? Resampling non-integer rates?

Started by Peter Mairhofer in comp.dsp5 years ago 2 replies

Hi, Is anyone using equipment like a VSA for data capture? Any idea how I can set the sampling rate in a Tektronix RSA? Whatever I set (e.g....

Hi, Is anyone using equipment like a VSA for data capture? Any idea how I can set the sampling rate in a Tektronix RSA? Whatever I set (e.g. using SENSe:ACQuisition:BANDwidth or SENSe:IQVTime:FREQuency:SPAN), the sampling rate is always 150 MHz! Frustrating... With an Agilent MSG I am transmitting an LTE signal which is sampled at 30.72MHz and to be received by the Tektronix VSA. (But ...


Sampling frequency offset correction (resampling) - help needed

Started by kobem in comp.dsp9 years ago 5 replies

hi, Can someone explain how I can compensate for sampling clock frequency offset (SCO) using fractional-delay interpolation? In OFDM...

hi, Can someone explain how I can compensate for sampling clock frequency offset (SCO) using fractional-delay interpolation? In OFDM basebnad receiver design book they said that you can use fractional-delay interpolation filter to compensate SCO but of course they forgot to write about. Let's say I have a ramp signal (it will be easier for me to explain what my doubts are as it has amplitude...


Resampling vector from the middle sample (Keeping the symmetry)

Started by Dan25 in comp.dsp5 years ago 2 replies

Hi All, I would like to re-sample a vector using different sample rates. But I would want to keep the symmetry as it is. E.g. h = [ 0.1 0.3...

Hi All, I would like to re-sample a vector using different sample rates. But I would want to keep the symmetry as it is. E.g. h = [ 0.1 0.3 0.7 0.3 0.1]; % Original vector with 5 samples h2 = resample(h,4,5); % Re-sampled vector into 4 samples But the above command doesn't give a symmetric vector as the original one. Is there any way to re-sample keeping the symmetry around the center ...


multirate resampling implementation

Started by chivak in comp.dsp11 years ago 1 reply

Hi , I'm new to multirate filters. I am trying to implement an interp by 3 and decim by 5 multirate filter. Lets say i have 40 taps...

Hi , I'm new to multirate filters. I am trying to implement an interp by 3 and decim by 5 multirate filter. Lets say i have 40 taps in the filter. I have 5 sub-filters with 8 taps each. In the 1st clock cycle i input 2 samples to sub-filters 1 and 4 In the 2nd clock cycle i input 2 new samples to sub-filters 2 and 5 In the 3 clock cycle i input 1 new sample to sub-filters 3...