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?

Hi, I am trying to extract MFCC coefficients for variable sample rate audio feed 1.Input Sample rate is dynamic and is not known in front (...

Hi, I am trying to extract MFCC coefficients for variable sample rate audio feed 1.Input Sample rate is dynamic and is not known in front ( an arbitrary media file ) though it is fixed per session and doesn’t change over time. 2.All Input feeds are being down sampled to a predefined sampling-rate ( say 8Khz or 16Khz ) 3.Re-sampling is done in the following way: a.2nd order Lowpass

Hello. Generally we do up sampling by inserting zeros and passing through Image rejection filter. Instead of inserting zeros we can duplicate...

Hello. Generally we do up sampling by inserting zeros and passing through Image rejection filter. Instead of inserting zeros we can duplicate the samples and pass through a filter. the advantage of going for second mathod is i can choose a less order filter compared to the first methode. similarly is it possible to do the same methode for sampling rate conversion by non integer factor. if so wh...

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...

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...

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

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...

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...

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...

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...

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...

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 ...

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...