Not a member?

# Discussion Groups > Resampling

Comp.dsp is a worldwide Usenet news group that is used to discuss various aspects of digital signal processing.

We found 292 threads matching "resampling"

You are looking at page 1 of 8.

The most relevant threads are listed first

## Resampling irrational or large rational ratios? 0

2005-07-08 15:30:00
Resampling with small integer ratios can be pretty staightforward, like a rate change of 3/2, upsample by 3 -> LPF -> dec by 2. However, it's not as simple when it's a ratio of large integers or some arbitrary new sample rate. I started to read this document on resampling: http://ccrma...

## Re: detune a signal 0

Jerry Avins - 2006-01-14 22:23:00
Hans Fugal wrote: > More context: this will be an audio processing plugin (VST and DSSI), > and the way plugins work is they take a block of n samples and return a > block of n samples. > > > > Resampling doesn't necessarily alter duration. > > > Would you elaborate on that? I t...

## Data resampling 0

Janey - 2004-06-16 00:09:00
Hi All, This query is related to data resampling. I have an ADC whose sampling frequency is 40Mhz now I need two different version of resampled data one being data resampled at 20Mhz while the other being at 22Mhz. Could any one please tell me what would be the best way of doing the resamplin...

## filtering before or after resampling? 0

sujani - 2004-08-19 12:00:00
Hi all, I have two sets of data captured at frequencies of 100 Hz and 120 Hz.In order to comapre the sets of data,I need to resample them to the same frequency.I also should filter the data. I am wondering which order is the best-filtering followed by resampling,or resampling followed by filter...

jon222 - 2007-04-17 10:21:00
Does anyone have Java or C code for a gradual resampling. I already have implementation of NyquistÂ–Shannon sampling theorem, where I can resample to any ratio. I know to calculate the amplitude of the signal at any time. Now I need to know at which time to take the amplitude to achieve the gradual ...

## Looking for math background to resampling 0

pslant - 2003-09-19 09:55:00
Im looking for some mathemathical background for resampling (of images in my case) What I'm after are 3 things specifically: (#2 most important) 1) How can a wider kernel result in a "better" sometimes "sharper" result. that feals very unintuitive to me. for example sinc 16*16 2) In the al...

## Farrow resampling filter 0

2007-09-12 11:51:00
Hi all, Do you have a general structure or a documentation how to design and compute the filter coefficients for a Farrow interpolation resampling filter. It's for a Gardner synchronisation scheme... Thanks ...

## Re: Which Scilab curve fitting approach is most applicable. 0

Richard Owlett - 2007-11-13 07:35:00
mnentwig wrote: > Have a look: > http://yehar.com/dsp/deip.pdf > -mn Just had a chance to glance at the table of contents. It looks worthwhile. From the title page: "Polynomial Interpolators for High-Quality Resampling of Oversampled Audio" by Olli Niemitalo, August 2001. Abstra...

## combined arbitrary resampling and match filtering for timing recovery 0

Awan - 2009-07-03 10:38:00
Hi DSP Guys, Can you please help on polyphase matched filter which is doing arbitrary resampling and timing recovery at the same time ? regards Awan ...

## Resampling libs 0

Ronaldo Souza - 2010-08-16 13:58:00
Can anyone recommend a good resampling lib? I need to handle rather small buffers (100s of Ks of signed 16 bits) but also really long streams, often covering 24+ hours @250sps, so speed is important. BTW, tipically I'll be downsampling from 250sps to 200sps, but there will be other scenarios w...

## easy resampling bandwith question ... 0

sammy davis jr. - 2003-06-25 12:34:00
hi all, let's say i want to resample to 4/5 the input samplerate, i.e. interpolate 4X then decimate 5X. for both steps lowpass filters need to be designed. according to dspguru.com, it is sufficient to 1. determine cutoff freq. for the interp. filter 2. determine cutoff freq. for the dec...

## resampling 0

mikejones - 2006-10-24 09:05:00
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...

## Resampling and Filtering Process 0

hamkanen - 2011-01-05 06:23:00
Hi, i'm designing a resampling process that implement a FIR filter. The resampling rate is 3/2 and the FIR filter has 73 taps , so the step of processes is 1. Upsampling by 3 2. Filtering with 73-taps filter 3. Downsampling by 2 Let's say i have 2000 array of input signal, so the output ...

## Intel's polyphase resampling function 0

Matt Roos - 2004-08-25 14:10:00
Hi, All. I'm trying to use Intel's IPP polyphase resampling functions. They are part of the signal processing set and found in the manual under Speech Recognition Functions. I've found the manual's usage description somewhat lacking. I don't understand it well enough to implement. Can an...

## Low Frequency Noise from Resampling? 0

firstcranialnerd - 2007-02-06 09:32:00
Hi, I am looking at some simulated heart rate data and it is unevenly sampled events, so I resample the data at say 4Hz or 7Hz or thereabouts. When I take the fft of the resampled data (it has been zeromeaned, filtered, hamming windowed etc) I see a huge amount of noise, especially from approximat...

## Live Audio Resampling between the 8000 and 48000 sample rate 0

Emmer - 2005-09-20 11:59:00
Dear All, I hope to do the audio sample rate convertion(resampling) between the ISDN and my device. The ISDN audio sample rate is 8000, 16bits/sample, Mono but my device use 48000, 16bits/sample, Mono. I can do the resampling use some software library, but it is heavy loading. I hope to han...

## Cutting and Resampling 0

kingdavid3 - 2007-03-19 12:33:00
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 y...

## Resampling of irregularly sampled data in 3 dimensions using a convolution kernel (Lanczos) 0

Andy365 - 2011-05-10 10:24:00
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")? Therefor...

## Re: speed, pitch and tempo 0

Chetan Vinchhi - 2006-09-13 11:05:00
gaetanoortisi@yahoo.it wrote: > Chetan Vinchhi wrote: > > > "Time-domain harmonic scaling" is a commonly-used technique to > > achieve this sort of transformations. A google search ought to dig > > out some material. > > > > No interpolation technique is needed to change speed?...

## Re: Async. sample rate conversion for audio - various methods vs fractional delay filters (Farrow) 0

PFC - 2008-06-24 07:06:00
> 3) I'm sure there are other ways to approach the problem, some mixture > of the above two etc... Yep. The new ESS Sabre DAC seems to use a novel approach. This is a DAC which runs on its own clock (low jitter) and it accepts digital audio data in SPDIF or I2S with an asynchronous ...

## Video Port Capture. Interesting Questions. 0

arichard - 2007-06-12 07:10:00
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 comp...

## Re: Announce: Fast real-time-stretching and resampling algorithms (Procrustes/Sylea) 0

DSP team - 2006-05-23 10:50:00
Paul Toritz wrote: > How does your resampler compare to Secret Rabbit Code: > > http://www.mega-nerd.com/SRC/ Let the others say this :) > Secret Rabbit Code gives specs for signal to noise ratio and bandwidth > as well as speed. Our bandwidth is 47.7% of the sampling rate. Y...

## Re: Fast down sample filtering 0

Grant Griffin - 2009-12-10 08:09:00
Eric Jacobsen wrote: > On 12/9/2009 7:54 AM, km wrote: > > Hi All, > > Does anyone have a suggestion for a fast down sampling filter? > > For N:1 reduction, add N consecutive samples together and output that. > Move to the next window of N samples. > > The response will not be...

## Re: New low spectral noise window function 0

dbd - 2010-12-29 11:50:00
On Dec 29, 5:55=A0am, "waywardgeek" wrote: > .. > > I lean toward 'time-aliasing' as the process seems to resemble the > > effect of aliasing in the frequency domain. > > Dale B. Dalrymple > > I see. =A0There are algorithms (WSOLA, PICOLA, TD-PSOLA) that are time do= main > only ...

## downsampling 0

Rob Edgar - 2003-08-11 05:45:00
Hi there, First be gentle! I know very little about this stuff :-) I have been asked to develop a module to emulate whatever SoundForge does to downsample a speech only wav file from 44.1khz to 11.025khz and 8khz when it has the "apply anti-alias filter" set. This has been done for years and ha...

## resample routines accessable from VB 0

d1camero - 2006-11-07 17:44:00
Hi all, I am having a tough time trying to find routines for resampling that will work from VB - any ideas? thanks Don ...

## Fractional Resampling WCDMA gives noise 0

DSPWirelessGuy - 2011-07-29 08:20:00
Dear all, I will be grateful if I can get some light on this. I have Tx Data upsampled from 3.84MHz to 12 MHz (@3.125X. On the Tx side I use a nice RRC filter to perform interpolation and upsample using Matlab inbuild polyphase filter to generate Test vectors. Problem is , on the receive...

## usage of the library doing audio resampling (libresample) 0

Viv - 2010-10-12 03:53:00
Hi all, I really hope this is the right place to post my question because I really need some guidance from some experts. I was told to find out a library, preferably LGPL, but in a worst case scenario also a commercial one would be ok, that would do resampling. Some scenarios where I will...

## Re: Time varying phase correction 0

mnentwig - 2011-08-24 22:51:00
Hello, If resampling on a uniform grid with +/- n samples will do (I doubt it): - fft - zero-pad or remove n samples from the center bins - ifft. Depending on n, this may be quick or slow. - you can zero-pad the input signal to increase resolution. For example, 10Msamples should be still mana...

## Re: Sampling rate conversion 0

robert bristow-johnson - 2009-02-10 17:04:00
On Feb 10, 4:16=A0pm, "cpshah99" wrote: > > I have done something using linear interpolation essentially, the only difference between this and what i am alluding to is that your reconstruction filter has an impulse response of { 1/T*(1 - |t|/T) for |t| ...

## Resampling and Convolution 0

nazmat - 2006-02-16 06:32:00
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: PAL to NTSC conversion software on DSP 0

Mark Robinson - 2005-10-25 08:31:00
PB wrote: > > Does anyone know of a software algorithm implemented on a DSP that > performs PAL to NTSC conversion? Also which DSP? The colour decoding/encoding parts of the job are best handled by the video encoder and decoder chips that you use to digitise the video. Look at the SAA7...

## Re: Help me to understant sampling theory, decimation by integer factor 0

Ron N. - 2008-02-22 15:03:00
On Feb 22, 5:11 am, "hyjeon_0_o" wrote: > Hi, everyone! > > I read "decimation by integer" and "interpolation by integer factors" > in a DSP book. > I'm just wondering why integer factor to downsampling... > > Sometimes, to resample signal by rational factor, people use conve...

## Re: Resampling and Filtering Process 0

Jerry Avins - 2011-01-06 11:36:00
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 ...

## Re: want to understand how Matlab handle image resolution/rate conversion? 0

Erik de Castro Lopo - 2005-02-11 23:04:00
kiki wrote: > > Second question: my understanding of "imresize" is rate conversion. Suppose > I want to upsample to a ratio of 7/5... > > In DSP principle I should first upsample by 7(fill in zeros every other 6 > pixels, and expand the size of image to x7) and then downsample ...

## Re: interesting insight into impulse definition 0

Greg Berchin - 2008-03-12 14:02:00
On Mar 12, 1:46=A0pm, Eric Jacobsen wrote: > Is there a case where it really becomes problematic in a practical > sense? One very simple example comes immediately to mind. A while back I was explaining the concept of resampling to a higher sampling rate (I don't remember whether we've...

## Re: measuruement of peak amplitude of each cycle of sinusoid having varying frequency 0

Ron N - 2011-08-30 21:22:00
On Aug 21, 11:03=A0pm, sanaashau...@hotmail.com wrote: > I am measuring the peak amplitude of each cycle of a sinusoid having > frequency of 125Hz but the frequency varies in the range 100Hz to > 150Hz after around every 3 cycles. I sample the signal for 8ms to > capture 1 complete cycle of ...

## Re: Sample Rate Conversion (Downsampling) 0

Ronald H. Nicholson Jr. - 2004-12-17 23:14:00
In article , Jim Thomas wrote: > Each value of y depends on every filter tap, so I still don't see why it's > polyphase. If this is polyphase, aren't ALL FIRs polyphase? Yes. All this polyphase stuff is just a distraction, *except* for reasons of implementation efficiency. Standar...

## Re: Amplitude Compression 0

bogfrog - 2008-08-05 14:14:00
Thanks for the input, everyone. I'll take some time to digest what's been written. I'm an undergrad student, and even though I've finished implementing the algorithms in this paper, I am not required to do every single robustness test, even though I'd like to. The other stuff is easier (filteri...

## Re: Zero padding fftw 0

Clay - 2010-12-09 14:53:00
> > That depends on your definition of "resolution". =A0Doing an FFT, zero > padding, then doing an IFFT is, IMHO, a valid way to do sample rate > conversion in batch applications. > I've actually read a few articles based on exactly this approach to resampling. It is valid. Clay ...
| 1 | | 3 | 4 | 5 | 6 | 7 | 8 |