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Questions tagged [resampling]

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Theoretical considerations for Using Finite Arrays to Represent Samples of Signals of Infinite Length

Discrete signals can come from sampling continuous-time signals. A continuous-time signal can be of infinite support. For instance, a sin wave. In terms of mathematical description of signals and ...
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1answer
32 views

What is the meaning of spectral contamination in image processing?

I resize the Multispectral image to perform IHS-to-RGB transformation. I used the Nearest Neigbour Interpolation method when resizing. While researching this issue, I saw that this is the best option ...
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1answer
49 views

Streaming windowed sinc interpolation/resampling: trying to understand a Rust implementation

I'm working on a fork of the Rust dasp library, which is intended to be a DSP toolkit that abstracts over samples/frames/signals, and contains a number of functions ...
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1answer
57 views

Python: Resample without equal

I would like to resample (downsample) a signal using python in order to get an even spacing and fill gaps. About the signal: It consists of a vector for y (amplitude) and x (timestamps) Very slow; $...
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1answer
125 views

Multiple choice on sampling and aliasing

I found some multiple choices in a well known book . The problem is that I don't get the answers in some and I want to do so. Question 1: The signal x(t) with Fourier transform $X(j\omega) = u(\omega)-...
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1answer
51 views

Remove noise introduced while converting a Delta-Sigma Modulated (DSM) signal into a pulse-code modulated (PCM) signal?

I'm writing a tool that converts a dsdiff audio file (dsd64) into a flac audio file (DSM -> PCM). The project is both a programming exercise and a signal processing exercise; I'm learning a lot ...
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1answer
54 views

What are the 'phases' in this 2D scaler filter

I am pretty new to DSP and have been looking at how image scaling is done in hardware. I and came across a 2D scaler core for an FPGA here. I have read up a bit about polyphase filters, but am ...
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2answers
58 views

draw sines at higher freq

I'm trying to plot some sine waves (code example in plain js here). When freq is "low" (freq = 10 hz in that case), the plot is quite nice: The problem ...
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1answer
84 views

Apply filter before or after Downsampling?

Context: I'm trying to do formant estimation. My first attempt didn't work out all that well so I'm trying to follow closely Praat since their formant estimation is remarkably accurate. Question: I ...
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0answers
699 views

How do I change the sampling rate without changing the speed and pitch in Python?

I am building a voice recognition algorithm, and the use case specifies that it should be able to identify the targeted speaker from various audio sources, which means that it must be able to process ...
2
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1answer
237 views

How to implement sinc interpolation

I'm trying to write my own high quality audio sample rate converter. I barely know anything about signal processing though so I need help. From what I understand I need to sum together normalized sinc ...
0
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1answer
37 views

How to transform a signal to go through specific points?

I have a 1d signal obtained using a Fourier based resample method (TDIFDZP) for which the resampled points don't necessarily go through the original samples. I want to transform the upsampled signal ...
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0answers
30 views

Resampling with factor linearly moving from one value to another: calculate expected target length

I am resampling an audio (1D) signal, using a resampling factor that moves linearly from startFactor to endFactor; say the input ...
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1answer
79 views

Oversampling a signal

I’ve built a circuit with an analogue sensor (70 Hz bandwidth) connected to a 24 bit ADC, which is oversampling the signal at 10 kHz. Nyquist sampling theorem implies I should sample by at least 140 ...
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0answers
35 views

Resampling assessment

I'm trying to figure out which of the resampling methods is best in my job. Comparing the original image vs. bilinear, bicubic, bicubic spline, lanczos. For now I have considered metrics like MSE RME ...
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1answer
82 views

Upsampling implementation

I have problem with upsampling implementation in the following way: My code: ...
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1answer
93 views

Upsampling in Python

I'm trying to perform upsampling in Python. Given the signal 's' and factor of upsampling N, I wrote the following code: ...
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2answers
212 views

Does resampling cause phase shifts?

I have a signal sampled at 40MHz, I would like to resample it to 37MHz. The signal is not periodic, I did resampling with Matalb resample function and it doesn't cause phase shift (as far as I ...
1
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1answer
61 views

How to correct for delays in a PAM-X signal with orthogonal CDMA?

I have a receiver which receives data bits at a rate of 5kbit/s. The number of data bits per burst is low (say, 16 bits). These databits are obtained after despreading the incoming signal using an ...
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2answers
173 views

Downsampling issues

I have this data that need to be downsampled so that it can be fed into a classifier. The issue is that I need to downsample from 1000Hz into 20Hz. If you see the graph below, the blue line is the ...
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2answers
120 views

Fourier Transform of sampled reflectance measurements

Goal: Calculate the complex index of refraction ($\hat n = n +jk)$ from reflectance measurements. Data: Reflectance measurements for various materials were taken with a FTIR (Nicolet™ iS50 FTIR, ...
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2answers
79 views

Rectify signal basic question

When rectifying a signal by squaring it, is it necessary to upsample by a factor of 2? Why is this?
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1answer
104 views

Upsampling question- Aliasing/imaging problem

I'm trying to study vibrations in a non-stationary shaft by upsampling the original signal. The original sample rate is at 25.6kHz, and it is being upsampled to 36kHz. In the FFT of the upsampled ...
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3answers
89 views

Interpretation of sample rate

I'm a little confused about the sample rate: If I have an audio file that was recorded with a sample rate of 48kHz, this means that there are 48,000 values per second in this file, right? So when I ...
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1answer
41 views

Obtaining a high sample rate digital approximation to an analog filter from lower frequency measurements

Consider a set of dense, but generally irregularly-spaced frequency response measurements of some real low-pass analog filter. Denote the maximum frequency for which a frequency response measurement ...
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2answers
50 views

Resampling of DTFT

I have a constant digital signal that is 1 for every sample and of length 4. 4 point DFT coefficients are $$[4,0,0,0]^T$$ Obviously. I wonder, if I resample the DTFT such that samples are taken at $$...
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0answers
47 views

Composition of interpolation and decimation matrices

I understand that interpolation is a linear transformation of a signal vector that combines interleaving the elements of the input vector with zeros followed by a filtering operation to remove any ...
0
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1answer
224 views

How to Emulate Sample Frequency Offset for OFDM in Matlab?

I am studying the effect of sampling frequency offset on OFDM, and in order to emulate SFO, I have to slightly change the sampling rate of the signal. In order to have a +5ppm offset, I have to change ...
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0answers
28 views

resample interpolation decimation filters algorithm

I'm creating a program in order to perform Resample, Interpolation and Decimation Frequency. ...
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0answers
54 views

Weird resampling artifacts of a periodic signal

I am trying to grab time series data from a database storing data on a propritary format. When i chose a resolution which is not a multiple of the power of two of the sampling resolution i get these ...
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1answer
25 views

Decimating by last interpolating factor? (Multi-stage)

Let's say I have a multi-stage interpolator at the transmitter, which interpolates by 2, 4 and 8 (that is, 64). When decimating (at the receiver) why should I start decimating by 8, then 4, then 2, ...
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1answer
461 views

How to prevent distortion after down-sampling?

I am new to audio signal processing, but in one of my machine learning projects that I am working on, I need to down-sample my audio somewhere in the middle of my neural network. Therefore, I need a ...
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0answers
50 views

Interpolation FIR filter Remez design criteria

I want to upsample an audio signal by a factor of 8. The technique I was going to use was to upsample by zero stuffing and then interopolate using a low pass anti-aliasing FIR filter designed using ...
0
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0answers
109 views

Delay of non-polyphase vs polyphase FIR filters

I implemented a polyphase filter function for interpolating and decimating. It works great (I compared it to what I'd get by interp/decim by inserting L-1 zeros and I get the same results) except for ...
2
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3answers
962 views

Resampling and removing high frequency noise?

I am currently working on a simple sampler that will allow me to load in a wav file and use my MIDI keyboard to play the loaded wav sample at the frequency according to the note played. Now I need ...
0
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1answer
167 views

Upsampling and pulse duration

How does upsampling factor relate to pulse duration? If I want to do pulse shaping and I have rcosdesign(beta,span=8,sps=4) should I upsample by ...
3
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1answer
285 views

Optimal trade-off between oversampling and filter length

For some sampling-frequency-preserving operations on Nyquist–Shannon sampled signals, such as: a shift a.k.a. translation, and differentiation by applying a derivative filter a.k.a. gradient filter, ...
1
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1answer
35 views

Method to rescale signals to mean length

I have a set of signals of varying lengths. I have provided an example of the same below - Their lengths vary between 186 to 202, with a mean length of 197. I am looking to rescale them to the mean ...
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0answers
26 views

Converting the equalizer coefficients from VSA at 2 sample/Symbol to 1.5 sample/Symbol

I was looking for some feedback on conversion of equalizer/filter coefficients at different sampler rate. Problem Statement: We have built a transmitter system whose output is at 1.5 sample/symbol (...
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3answers
578 views

Absolute convergence of periodic sinc interpolation

An $N$-periodic complex discrete-time sequence $[x_0, \dots, x_{N-1}]$ can be resampled to an $M$-periodic sequence $[y_0, \dots, y_{M-1}]$ with $M>N$, using sinc interpolation: $$\begin{align}y_m ...
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0answers
139 views

Why do I have to scale filter coefficients during bandlimited sample rate conversion?

I am writing an example for how to perform sample rate conversion using windowed sinc and polyphase filters. Here is the C++ code as a github gist. So far, this works, and I think I have understood ...
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0answers
30 views

delayed result in sinc upsampled signal

I have implemented an infinite kernel sinc interpolation. For some signals I notice an artifact, visible at the right end of the plot. I am wondering: is it a bug in my code or a phenomenon usual for ...
0
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1answer
573 views

Extrapolate a 2D array using Fourier Transform

I need to extrapolate a given 2D array to a larger domain, keeping the spatial frequency. This is the original field: (the data file in numpy npz format and a Jupyter notebook to plot it can be found ...
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0answers
60 views

How should I modify my filter coefficients for the filter to be unaltered, when I change my sample rate

I am guessing this is a commonly encountered problem, but I am lost on how to really solve it. I have FIR filter coefficients that are used to match the frequency response of data sampled at 3 kHz. I ...
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1answer
369 views

Best way to resample an OFDM signal without interpolating + filtering?

I am working on a custom transceiver with a sampling rate of 3.84 MHz (256 OFDM subcarriers at 15 kHz, although 56 subcarriers are null). I would like to resample the signal to 5 MHz, in order to use ...
1
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1answer
234 views

Is it possible to lose half the samples, and yet see the full spectrogram as the original signal?

Consider an speech audio signal sampled at 16,000 samples per second. Plot its spctrogram. If we delete every other sample from input signal, we get y[n] = x[2n] for all n. It seems that if we plot ...
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1answer
124 views

Having Glitches trying to use a Sinc Function Interpolation on a Signal, by chunks of 1024 samples

I am trying to interpolate a signal using a third party lib that performs Sinc Function Interpolation on a signal. It works well if I input all the samples at once but introduce phase distortion if I ...
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0answers
155 views

Matching two signals with small time drift

I have two time-domain signals sampled at 100 Hz that were measured using two different oscillators and therefore have a time drift between them. I have two synchronization points, one at the start ...
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1answer
1k views

The condition in order to not lose information after upsampling and downsampling [closed]

If you have a signal x[n] and it is interpolated by N and then decimated by M, how do you ensure minimum distortion in x[n] from the resampling process?
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1answer
1k views

Resampling from variable to fixed-rate [closed]

I have a gyroscope which reports x,y,z values to my embedded device. The rate that one sample appears is variable and has a jitter of around 10ms. However I later want to have a log file with a fixed ...