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Resampling irrational or large rational ratios?

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

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

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?

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...

Gradual resampling

Does anyone have Java or C code for a gradual resampling. I already have implementation of Nyquist–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 r...

Looking for math background to resampling

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

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.

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...

easy resampling bandwith question ...

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

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...

Intel's polyphase resampling function

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?

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

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

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...

Re: speed, pitch and tempo

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)

> 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.

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)

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...

downsampling

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

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

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

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

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

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

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: Sample Rate Conversion (Downsampling)

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

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: Using Multirate algorithm for doing pitch shift effect ?

> rjb said: > could be a simple scaling issue. do you understand the interpolation or > sample rate conversion problem as commonly presented in a DSP context? > because if you don't mind the length modification, then that's all the > problem you have and that isn't so hard. Hi rjb, t...

Re: FFT Questions.....

Hi, > that can be absolutely determined. Is there an algorithm that performs > a second fft of the same original signal, but over only the frequency > bands that the current transforms resolution can't determine? resampling is insertion zeros in original consequence, up to needing resolution....

Re: Step response from adding impulse responses together

i don't know if u have come across this already. Technical report university of cambridge computer lab Image Resampling. Neil Anthony Dodgson August 1992. Regards Bharat Pathak bharat@arithos.com Arithos Designs www.Arithos.com ...

Frame-based resampling in MATLAB?

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 multira...

Re: Different Resampling Rates for Segments of a Sound File

Jon, I have wondered whether the blips/pops were at the boundaries, but have not investigated enough to determine this or not. I suspect that they are not, but if they are I may well have to implement something like you describe. I think Erik is just trying to help me understand how to use ...

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

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 ...

Re: Interpolation/FIR/etc... =]

> > Farrow is usually used for variable rate interpolation And the reference says: .. "We have rederived the Farrow filter which supports continously variable resampling". But if you have too little time, too loose specs and/or too much memory, a large polyphase filter and rounding to the near...

audio sampling rate question

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...

Re: Pitch

play it faster? what are you trying to do this with? are you talking about pitch-scaling or resampling? try http://www.dspdimension.com or http://www.mega-nerd.com for useful resources on each /chris Lars Tegborg wrote: > > Hi > > Does anyone know how to pitch up sound to double...

Re: Huge resampling factors

stilghar wrote: > I've to resample a signal from 16.3676 MHz to 4.096 MHz. If I'm not wrong > I should interpolate by 10240 and then decimate by 40919. But, these are > huge numbers! Is there any other way of doing it? Depends on how much tolerance you have for the output sample rate. 256/...

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

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...

Re: Sine wave look up table

Thanks Jerry and Clay. I'm going to take a closer look at this. For small FFTs, a table works great. For larger ones, I've traditionally used CORDIC, and have been looking at lower latency and other implementations that use the fast embedded multipliers, including using a small table comb...

Resampling

Dear, I want to resample a real signal from for instance 1024 -> 1027 samples. I prefer not to do it in the temporal domain, in order to avoid large upsampling and downsampling steps. I have read that one possibility could be zero padding in frequency, but for this example I must use a IDFT and...