Questions tagged [interpolation]

Interpolation is a method of constructing new data points within the range of a discrete set of known data points.

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Pixel Shift when using Fourier Downsampling

I am working on a Super-Resolution network, and want to try to use Fourier downsampling, different from the normally used bicubic downsampling as it relates to MRI imaging. I have tried using this ...
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Understanding the Farrow Filter

I found the following diagram of a Farrow Filter online: As far as I understand the purpose of a Farrow filter is to interpolate samples at a fractional ratio. For example: we have an ADC sampling at ...
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fast spherical filter: interpolation

Here is the minimum working examples of fast spherical filter in c++ The reference paper is Fast Spherical Filter There is a bug in the implementation where the ...
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Interpolation of STFT between time and frequency bins

I am trying to identify and remove airplanes in a time series obtained at a radio meteor receiver. Here we have a spectrogram which displays the power content of my signal as a function of frequency ...
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How are the results of a fourier-transform interpolated to cover the entire frequency range?

Say we have an audio signal and we do a Fourier transform with block size 2048. Now the outputs correspond to 1024 different frequencies. A spectrum analyzer however doesn't just display 1024 discrete ...
Potheker's user avatar
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How to show that these two factors are related to frequency interpolation and extrapolation respectively

Consider two sequences $x[n]$ and $y[n]$ being simultaneously sampled from a real-world signal. Let $x[n]$ be a sequence of length $N$ sampled for duration of $T_x$ at rate $f_x$ $n = 0, ..., N-1$ ...
auckydocky's user avatar
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What is wrong with this sinc interpolation? (Zero padding in frequency domain)

I'm trying to pad $N=16$ zeros to the DFT of a $16$ point sinewave, but something is wrong either with my code or with my method. The method is this : I create a new $32$ point DFT where $F'(0),...,F'(...
In the blind's user avatar
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Room Impulse Response Domain of Sparsity

I have been studying the problem of room impulse responses (RIRs) interpolation for a couple of months. I am trying to use compressed sensing to reconstruct (at best) the sound field in the room with ...
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interpolation: why I can see the original sequence in the interpolated data only when (N/M) = integer

I am trying to understand the zero-padding in frequency domain and discovered that original sequence in interpolated data only when (N/M) = integer where N and M are the lengths of output and input. <...
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Transmitting data with QPSK using an SDR

My question assumes that I want to transmit some data using QPSK using and SDR. The SDR will be fed with IQ data and this will transmitted at a sample rate Fs and be up converted to a carrier ...
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LTE PSS Channel Estimation for coarse SSS Equalization

In LTE the Primary Synchronisation Signal (PSS) can be detected by taking a correlation with the known 3 zadoff chu sequences (roots: 25,29,34). Once the peak has been detected the receiver knows ...
Sonny Wandel's user avatar
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Estimating spectrum with regularly missing samples from data

Suppose: $$ s(t) = \sin(2\pi{f_0}t) $$ Suppose I'm sampling the signal with a sample frequency $f_s >2f_0$ . However, every $M$ samples there is a dead-time of $L$ samples. Traditionally, the (...
Sammy Apsel's user avatar
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On FFT, interpolating signal vs extending signal in time

When we interpolate, then FFT the output will have more bins. When we extend the signal in time, Then FFT output will have more bins too but: Interpolation increases max bin frequency but time ...
mohammadsdtmnd's user avatar
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Frequency spectrum of interpolated sine

When interpolating a sine wave in Python, it seems I get a lot of additional frequencies closely around the fundamental. Why is that the case? I would have expected a more or less similar spectrum, ...
MisterFilter's user avatar
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what degree of accuracy does quadratic interpolation add to fft cross-correlation?

I am not a dsp engineer, so I might be missing some obvious background, but I've been doing my best to educate myself. Let me know if there's more info needed. I am using an fft cross-correlation ...
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Real time variable speed audio playback with interpolation

I'm trying to correctly implement the real-time variable speed playback of my buffer, to achieve the pitch shifting. I don't need to stretch to preserve the original length. Buffer is calculated with ...
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How can I use interpolation to align audio [duplicate]

I notice that when I connect two audio clips, there is a jump at the junction. How do I use interpolation to connect these two pieces of audio? Thanks for the answers, I've got an answer from my ...
Killuaisaack's user avatar
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Averaging a Signal with a delayed version of itself

I am working on a problem that requires interpolating between two music files. One music file is very similar to a delayed version of the other. I would like the 'average' of the two signals to sound ...
Mason Wang's user avatar
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Sample rate change using Lagrange interpolator

I'm trying to understand how to make Sample rate change by a non integer factor say $K=1.0012$ In real hardware I will be using cubic polynomial but for now I have chosen a linear interpolator. I have ...
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Correct OFDM oversampling method

I need to oversample my OFDM signal, however I have issues understanding which OFDM oversampling method is the recommended one. First method : Before taking the IFFT, the OFDM subcarriers are padded ...
Nicolate's user avatar
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DSP library for Software Defined Radio

Can anyone recommend a good C library for SDR? To be more specific I just need a function in C to compute coefficients for CIC compensation filter (I need to do it in real time, so Matlab is not an ...
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How can the DCT be used for bandlimited interpolation?

I know the discrete cosine transform (DCT) is used for compression, but can anyone give an example of how to use it for bandlimited interpolation? One way might be zero-padding in the DCT domain and ...
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Why does interpolation with zeros introduce frequency artifacts?

Consider the original signal (just a sine) and the same signal interpolated with zeros: The Fourier transform of them: In this example I inserted only one zero between the samples, as if I sampled ...
Curious's user avatar
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Upsampled signal values

My question is: after upsampling a signal, does the output signal contain the original signal values? The diagrams I saw so far (on wikipedia and different forums) have always shown the original ...
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Resampling in time

I have a huge dataset, way too much to be loaded into memory. I need to downsample the data to a new rate. Normally, this would consist of upsampling by N, applying FIR LPF, then decimating by M to ...
Levi's user avatar
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Multiple questions about how to implement practical resampling

I am learning resampling theory, and for the time being I am specifically interested in downsampling. I have a textbook that is not a dsp textbook but has a section on resampling. The way they put it, ...
Levi's user avatar
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How should I interpolate on a triangle mesh?

I have a triangular mesh (generated from a Lidar point cloud). On this mesh I do point measurements using two instruments. One records M amplitudes of frequencies in the visible range and the other ...
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Does the combined pointsample+bilinear upscaling algorithm have a name?

I have been performing digital image upscaling of pixel art in one of my hobby projects, and the two simplest upscaling algorithms are point sampling or bilinear upscaling. Point sampling exhibits ...
trojj's user avatar
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Are there ways to reduce the smearing of zero-padding interpolated data?

Are there ways to reduce the smearing / spectral leakage of zero-padding interpolated data? I learned that, given a small collection of samples, one can increase the frequency resolution of an FFT ...
Alex Larson's user avatar
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Reconstructing an undersampled signal by cutting off at the signal's maximum frequency

Assume a (continuous) band-limited signal $f$, that is, a signal for which $F(s) = 0$ for all $\lvert s \rvert > p / 2$. If the signal is sampled with frequency $p$, we can reconstruct it by ...
Dan Oneață's user avatar
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Accurate frequency estimation with closely-spaced tones

I have a MIPs-limited DSP platform where I am taking an FFT of a streaming audio signal at regular intervals. I need to find the top 4 amplitude peaks and their corresponding frequency for every frame....
Bob's user avatar
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Extract diagonal area from image

I have a gray scale image of fibres in different orientations. My goal is to mark the area where the fibres have a specific angle and neglect the rest. In the future it should be an automated process ...
Till's user avatar
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Spline interpolation of a binary mask

I have the following binary mask (If it helps you can assume that every it is always 1px wide). I want to travel from one given point to another (from start to end in this case). You are given the ...
Assefa Seyoum's user avatar
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Zero padding versus zero stuffing

Let's say I want to increase the sampling rate of my signal, $f_t$, from a sampling frequency $f_s$ to some multiple $Mf_s$. One way to do this is to add zeros between the samples of $f_t$. This ...
In the blind's user avatar
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1 answer
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Interpolation to create equally spaced value using Python

I am working on a project in which I am required to use Interpolation to create equally spaced position values as the position samples are not equally spaced due to varying train speed. So, the total ...
Aisha Nasir's user avatar
2 votes
1 answer
349 views

Applying Kaiser Window to sinc interpolation

I am trying to apply the kaiser window during sinc interpolation. The following is my sinc interpolation code: ...
Hari's user avatar
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16 point Sinc Interpolation Kernel

I have a signal that is sampled at Fs and wish to apply a "16-point Sinc interpolation kernel". However, I am not sure what the 16-point refers to. Does it refer to having a kernel that is ...
Hari's user avatar
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Multirate Control System Transfer Functions

I'm interested in oversampling the inputs to a digital controller to increase the SNR of the input process variable signal. I've read on this site and in articles like the one below that it is not ...
FooAnon's user avatar
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How to interpolate the peak amplitude of an fft output?

I want to find the peak amplitude of an audio signal in a given frequency area. My problem is that if the signal has a strong resonating frequency between 2 fft bins, then both bins are way below the ...
Potheker's user avatar
5 votes
2 answers
888 views

Odd artifacts after sinc interpolation

I am trying to upsample a signal using sinc interpolation. I have written a way to do this in python. ...
guitardenver's user avatar
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Guided bilinear interpolation: is it a known algorithm?

I have a function that's sampled relatively coarsely along one direction, $x$, and much more finely along another, $y$. Sampling grid is regular. I need to interpolate between all these samples, with ...
Ruslan's user avatar
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Basics of interpolation: is filter or upsample first?

I am studying interpolation filters. I'm under the impression that when talking about interpolating a signal, we filter and then upsample the signal. BUT In an article, Interpolation (Digital Filter ...
AD23's user avatar
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Interpolate & sample period GNU Radio

I've a little question regarding interpolation in GNU Radio: I'd like to do an interpolation (by a factor of 2) and be able to change the sampling period of the samples interpolated but im having a ...
Tira's user avatar
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Fractional Frequency Shifting a Discrete Signal in the Frequency Domain

In our current hardware-based signal processing pipeline, we have a time-domain signal $x[n]$ that we want to frequency shift by $f_0$. To do this, we multiply the signal by a complex exponential in ...
Logan's user avatar
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Plotting the Frequency Response of Zero-Order Hold

I'm trying to find the frequency response and the DC gain, then plot the response and find the cutoff frequency from the graph. I start off with the impulse response $$h(t)= 1 ; 0\leqslant t\leqslant ...
April Crude's user avatar
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Why is this Python implementation of trigonometric interpolation not working properly?

Consider a signal $u_j, j = 0, 1, \dots, N-1,$ sampled on an evenly-spaced grid of points, $x_j$. Define the discrete Fourier transform of $u_j$ by $$U_k := \frac{1}{\sqrt{N}} \sum_{j=0}^{N-1}u_je^{-(...
coba22's user avatar
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Performing Sample-Rate Conversion on Subchunks of an Audio Signal

I have implemented sample-rate conversion (SRC) from scratch in my hobby music player, and it worked fine. However, originally, the music player loaded entire audio files into memory and performed the ...
MulattoKid's user avatar
1 vote
2 answers
660 views

Symbol Timing Recovery with Fractional Sample per Second

I'm trying to write symbol timing recovery loop and taking help from "Digital Communications A Discrete-Time Approach" by Miachel Rice. MATLAB communication toolbox has also implemented the ...
ubaabd's user avatar
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How to implement interpolation by convolution for rotation of an image by an angle which is not a multiple of 90 degrees?

I know that the usual way to perform rotation of an image is to compute the new pixel coordinates by multiplying with a rotation matrix $ \begin{pmatrix}x_{rot} \\ y_{rot} \end{pmatrix} = \begin{...
SheppLogan's user avatar
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Why does upsampling and interpolation by convolution introduce a shift compared to imresize?

For purposes of understanding the process - not for any practical purpose for which I could use imresize - I wanted to show that 2x upsampling followed by convolution with an appropriate kernel (...
SheppLogan's user avatar

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