Questions tagged [resampling]

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Role of oversampling within resampling / pitch modification

I'm working on a pitch modification plugin involving resampling, and want the quality to be as high as possible. I currently use windowed sinc interpolation and lowpass filter where required. I ...
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1answer
30 views

Downsampling a signal containing artifacts

I have 2 kHz-sampled signal that I would like to downsample to 256 Hz. The original signal contains artifacts (= time intervals where data has too high/too low values. I know boundaries of the ...
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1answer
50 views

Resampling pitch shifter

I’m trying to build a very basic resampling pitch shifter which reads samples from disk. I only want to change -+6 semitones without keeping the original sample length. I already have all the classes ...
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1answer
48 views

Does resampling to low number of samples lead to loss in high frequency content?

I am building an one dimensional Convolutional Neural Network and wish to feed it raw audio but it turns out that the number of samples is very larger. The audio was sampled at 48 kHz and the duration ...
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2answers
58 views

Frequency response of discrete time system involving interpolation and resampling

I am working on a problem towards calculating frequency response of discrete time system(does interpolation followed by resampling) which looks like: x(n) is fed to linear interpolator(h(t) = ...
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3answers
40 views

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
34 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
59 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
83 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
156 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
70 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
59 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
100 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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789 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 ...
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1answer
267 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 ...
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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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32 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
90 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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37 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
85 views

Upsampling implementation

I have problem with upsampling implementation in the following way: My code: ...
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1answer
124 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
238 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 ...
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1answer
64 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
187 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
124 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
82 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
133 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
95 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
44 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
51 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
48 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 ...
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1answer
242 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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29 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
57 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
493 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 ...
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0answers
123 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 ...
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3answers
1k 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 ...
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1answer
174 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 ...
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1answer
298 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, ...
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1answer
36 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
27 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
586 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
143 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 ...
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1answer
637 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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65 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
401 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 ...