Back in Apr 25, 2004, MC Canzee posted a query in comp.soft-sys.matlab that said (in part): > i want to resample frame-based. > Therefore i...

Back in Apr 25, 2004, MC Canzee posted a query in comp.soft-sys.matlab that said (in part): > i want to resample frame-based. > Therefore i need a filter, that returns filterstates. > Like e.g. [y,zf] = filter(b,a,x,zi) does. > But due to resampling process this filter should be multirate to be > efficient, like e.g. upfirdn(); > The problem with upfirdn is that it does not return filte

I am reading a full length (duration of the speech) audio wave file sampled at 44.1KHz, 16 bit stereo. I want to cut a 15 second segment from...

I am reading a full length (duration of the speech) audio wave file sampled at 44.1KHz, 16 bit stereo. I want to cut a 15 second segment from the audio and resample it to 16000 Hz. So far I managed to do the resampling. Please help on how to cut 15 second length portion of the audio. Thank you [s, fs]=wavread('test.wav); %%cut to 15 seconds s=resample(s, 16000, fs); %downsam...

How do you get rid of resampling artifacts? The P/Q fixed-point resampler algo I am working on takes as input a signal which has been...

How do you get rid of resampling artifacts? The P/Q fixed-point resampler algo I am working on takes as input a signal which has been upsampled by P (zero insertion) and low-pass filters the upsampled signal (fc = min[1/P,1/Q] ). In the final step every Qth filter output sample is kept. The artifacts seems to get worse when P/Q is much larger than 1...for example when P/Q = 6. The low-pass...

Hi all, I am having a tough time trying to find routines for resampling that will work from VB - any ideas? thanks Don

Hi all, I am having a tough time trying to find routines for resampling that will work from VB - any ideas? thanks Don

Hi all, How can i convulve a data signal with a sampling rate of 100GHz with an impulse response of sampling interval of 0.167ns and the samples...

Hi all, How can i convulve a data signal with a sampling rate of 100GHz with an impulse response of sampling interval of 0.167ns and the samples are not evenly spaced.Thank you all. Nazmat

Rick, Hamkannen also needs to know that FIR filters are just an easily analyzed special case, and that all filters, even purely analog ones,...

Rick, Hamkannen also needs to know that FIR filters are just an easily analyzed special case, and that all filters, even purely analog ones, exhibit beginning and ending transients. Jerry

How does one detune a signal? My understanding is that you resample, but resampling involves changing the duration as well, so after processing...

How does one detune a signal? My understanding is that you resample, but resampling involves changing the duration as well, so after processing a block you would have a different number of samples than you started out with. Then what do you do? pad/truncate?

Hi, I need to downsample a bunch of signals, all of which have very different properties (the Nyquist criteria will not be fulfilled after...

Hi, I need to downsample a bunch of signals, all of which have very different properties (the Nyquist criteria will not be fulfilled after the downsampling). My question is how to proceed in the best possible way. All processing is off-line but has to be fairly fast. Case 1: An analogue signal (continuous amplitude and time) has been sampled and needs to be downsampled. I have no problems here...

Hi I need to resample a 32Khz signal into a 8Khz signal, and then back again to 32Khz after it has been processed and am looking for some...

Hi I need to resample a 32Khz signal into a 8Khz signal, and then back again to 32Khz after it has been processed and am looking for some guidance. I am using a dsPIC and have the following two library functions available to me, FIRDecimate and FIRInterpolate, I am also using the dsPICfd software to create my filters. After reading many tutorials on doing this, I am still a bit disorienta...

Hi, I am trying to do a 44.1 khz to 48 khz resampling in Matlab, but am coming across a few problems. For this conversion to take place...

Hi, I am trying to do a 44.1 khz to 48 khz resampling in Matlab, but am coming across a few problems. For this conversion to take place the poly phase filters come out like 480/441=(4 x 4 x 10)/(7 x 7 x 3). I have been trying to figure this thing out on paper, like the for the first stage I get 44100 x (4/7), now as I understand it I have to take a 44.1 khz signal upsample it by 4, appl...

I've got a single channel of 16-bit/44.1kHz audio that I'd like to resample to 16-bit/88.1kHz. I'll start with this since I believe that it...

I've got a single channel of 16-bit/44.1kHz audio that I'd like to resample to 16-bit/88.1kHz. I'll start with this since I believe that it might be easier than dealing with uneven mutiples if I chose another rate (like 48kHz for example). Please let me know if my understanding is correct: 1. First I need to zero pad each sample. I assume that I can just write a simple program to do thi...

Hello, I have a signal sampled at 16kHz, and I would like to detect when any of the frequency components within 0.5Hz-10Hz is above...

Hello, I have a signal sampled at 16kHz, and I would like to detect when any of the frequency components within 0.5Hz-10Hz is above predefined limit (say -10db). Because of high sample rate I expect that the resampling/decimating input signal will be necessary. Doing FFT just for yes/no answer seems like an overkill. I was considering bandpass filter with magnitude comparator. Ar...

Hello, When I want to interpolate by 1.5 I have to upsample by 3 and downsample by 2 to preserve the bandwidth of the original signal. If the...

Hello, When I want to interpolate by 1.5 I have to upsample by 3 and downsample by 2 to preserve the bandwidth of the original signal. If the input signal is N samples long the current implementation requires a temporary 3N long buffer. Are there smarter implementations where the 3N long buffer can be avoided? Thanks.

DSP Gurus: I'm familiar with the classic interpolate-filter-decimate approach to multirate DSP. I've heard that the chirp-z transform can be...

DSP Gurus: I'm familiar with the classic interpolate-filter-decimate approach to multirate DSP. I've heard that the chirp-z transform can be used to resample a signal also. Can anyone provide additional information or references (other than Google search results)? Thanks, OUP

Hello all, I have two audio devices which differ slightly in sample rate (i.e. 8000 Hz and 7999 Hz). I would like to have those streams in...

Hello all, I have two audio devices which differ slightly in sample rate (i.e. 8000 Hz and 7999 Hz). I would like to have those streams in the same sample rate, and the maximum tolerable delay is about 5-10 ms. Which methods exist, and where can I find some good documentation about this? The solution is to be implemented on a regular PC. Grateful for any pointers, This message was se...

I have a DPSK modulated signal that is sampled at 8kHz and a demodulator that works at 7.2kHz, which is a multiple of the bit rate. The base...

I have a DPSK modulated signal that is sampled at 8kHz and a demodulator that works at 7.2kHz, which is a multiple of the bit rate. The base tone is either 1200 or 2400 Hz. I'm looking for an effective way of demodulating the input signal. The method I'm currently considering consists of interpolating using a FIR to 72kHz with a LPF with cutoff frequency of about 3-4kHz, then decimating ...

I had to resample a signal with requirement of minimal processing delay. So I made a Lagrange polynomial extrapolator to predict the signal on...

I had to resample a signal with requirement of minimal processing delay. So I made a Lagrange polynomial extrapolator to predict the signal on the duration of +1 sample into the future. That is straightforward and it works good enough for the job. However what could be the other options for extrapolation of the Nyquist bandlimited signal? What is an optimal solution for this case? ...

I've come to realise that in order to get a resampling component that I have developed to integrate properly in my application, I will need...

I've come to realise that in order to get a resampling component that I have developed to integrate properly in my application, I will need to implement a buffer so that I can look ahead of the current buffer (at the expense of introducing a bit of latency). I assume that a FIFO would be the best way to implement this buffering. The question is: how efficient would the STL implemen

with the newer flash recorders I now have higher resolution and sample rate options then my older 44.1/16 PCM I realize I can record at lower...

with the newer flash recorders I now have higher resolution and sample rate options then my older 44.1/16 PCM I realize I can record at lower safer levels with the 24bit resolution but I was wondering if sampling at 96 or 88 and resampling to my final level of 44.1 after post processing has any advantage. -- Phil

I'm pretty new to the world of frequency domain as proven by this question: Is there any correlation between the quality of resampler required...

I'm pretty new to the world of frequency domain as proven by this question: Is there any correlation between the quality of resampler required to make aliasing "imperceptible" and the magnitude of sampling rate change? I'm currently playing with various resampling algorithms, running some 44.1kHz content through them (with lots of high and low frequencies) and I noticed increasing the samp...