Frame-based resampling in MATLAB?

Started by Jerry Wolf in comp.dsp13 years ago 4 replies

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


Cutting and Resampling

Started by kingdavid3 in comp.dsp13 years ago 2 replies

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 to get rid of resampling artifacts ?

Started by John McDermick in comp.dsp9 years ago 14 replies

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...


resample routines accessable from VB

Started by d1camero in comp.dsp13 years ago

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


Resampling and Convolution

Started by nazmat in comp.dsp14 years ago

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


Re: Resampling and Filtering Process

Started by Jerry Avins in comp.dsp9 years ago

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


detune a signal

Started by Hans Fugal in comp.dsp14 years ago 7 replies

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?


Best resampling approach for different types of data?

Started by chrah in comp.dsp10 years ago 20 replies

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...


Resampling for multirate system help

Started by billysdomain in comp.dsp13 years ago 6 replies

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...


44.1 khz to 48 khz using polyphase

Started by nomi in comp.dsp14 years ago 9 replies

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...


Resampling Questions - Newbie requests your help

Started by Please Respond Here in comp.dsp16 years ago 4 replies

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...


passband frequency detection

Started by Korenje in comp.dsp12 years ago 2 replies

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...


Resampling

Started by Jens in comp.dsp9 years ago 3 replies

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.


[Q] How can the chirp-z transform be used in resampling?

Started by One Usenet Poster in comp.dsp13 years ago 6 replies

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


Arbitrary asynchronous (plesiochronous?) resampling in "real time"

Started by snappy in comp.dsp14 years ago 32 replies

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...


DPSK resampling

Started by Unbeliever in comp.dsp14 years ago 4 replies

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 ...


Resampling with minimum delay

Started by Vladimir Vassilevsky in comp.dsp8 years ago 12 replies

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? ...


DSP newbie:FIFO buffers in C++

Started by Chris Sperry in comp.dsp16 years ago 3 replies

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


audio sampling rate question

Started by Phil in comp.dsp12 years ago 11 replies

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


Perceived aliasing in small ratio SRC

Started by somenoob in comp.dsp13 years ago 15 replies

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...