Signal Processing Stack Exchange
Signal Processing Stack Exchange

Questions tagged [frequency-domain]

278 questions
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How to circularly shift a signal by a fraction of a sample?

The shift theorem says: Multiplying $x_n$ by a linear phase $e^{\frac{2\pi i}{N}n m}$ for some integer m corresponds to a circular shift of the output $X_k$: $X_k$ is replaced by $X_{k-m}$, where the subscript is interpreted modulo N (i.e.,…
endolith
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When should I calculate PSD instead of plain FFT magnitude spectrum?

I have a thirty-second speech signal that was sampled at 44.1 kHz. Now, I would like to show what frequencies the speech has. However, I'm not sure what would be the best way to do that. It seems sometimes one calculates the absolute value of a…
spede
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Determining the noise floor of a signal in the frequency domain

Is there an accepted way of determining the noise floor of a signal by looking at it in the frequency domain? Is it a matter of averaging all the bins, or median, or some more complex calculation like those described in the question below? What's…
Dan Sandberg
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What is the type of noise in this image?

what reason could make the noise/effect in this image? Below is the original Thanks!
raboof
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Number of DFT (FFT) Points Required for a Specific Frequency Resolution for an Oversampled Signal

I have a bandpass signal centered at 2 MHz and bandwidth of 50 kHz (the signal frequency varies from 2 MHz - 25 kHz to 2 MHz + 25 kHz). This signal is being sampled at 10 MHz. I want a frequency resolution of 100 Hz in FFT. I want to know: How many…
Vinod
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What's the meaning of a complex zero/pole?

I have been studying signal processing and control for a while now, and I use Laplace and Fourier transforms almost everyday. Also another tools such as Nyquist or Bode plots. However, I had never thought of this until today: what is the physical…
Tendero
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How to Apply Window Functions in Frequency Domain?

I am trying to apply a window function to smooth my signal and finally obtain the peaks of the signal. Yet, for spectral leakage I apply window function in time domain. After taking the Fourier amplitude spectrum ı want to further smooth the signal.…
madraperla
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Continuous Time Signal and Discrete Time Signal - Connection Between Periodicity and Discretness

I know that all periodic continuous time signal have discrete spectral representations, but are all discrete spectral representations periodic in continuous time? Also, can all periodic signals be represented by a fourier series?
Dave
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Estimation of Amplitude, Frequency and Phase of Linear Combination of Harmonic Signal Beyond the Leakage Resolution of DFT

How can I find a rough ( as accurate as possible) Amplitude of each frequency when there is spectral leakage. Currently, I am dealing with a system that contains special leakage which seems unavoidable as I am measuring a real signal with any…
Jacob wood
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The Proper Way to Do Sinc Downsampling (DFT Downsampling) for Uniformly Sampled Discrete Signals with Finite Number of Samples

Given a discrete signal $ \left\{ x [ 0 ], x [ 1 ], \ldots, x [ N - 1 ] \right\} $ what would be the correct way to downsample it in the frequency domain (Sinc interpolation)?
David
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Idea for Noise Level Estimation / Automatic Thresholding in the Presence of Peaks

I have the fft of some signal, and want a rough estimate of the noise level in order to choose an appropriate threshold for our peak detection algorithm. In general, the fft contains mostly noise with a handful of peaks (which are usually pretty…
Felix G
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Amplitude of the Signal in Frequency Domain different from Time Domain

How does the Amplitude of the Signal is changed when taking the FFT of a Signal, Have a look, The amplitude had changed from 2 to 30, and Here is my code for generating the above output, f=1e3; %Frequency of Wave A=2; %Amplitude Fs = 1e6; …
Sufiyan Ghori
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Doppler shift in time domain?

I have some clarifications required in this topic So initially I know that to shift a signal $X(f)$ by $f_0$ we need to take its inverse fourier transform and multiply it with a factor of $\exp(−j2\pi f_0)$. Now as I understand it, in the frequency…
Avinash Suresh
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How do real-time convolution plugins process audio so quickly

Okay, so in Logic Pro I can load up a Space Designer plugin (convolution reverb) with an impulse that's 9.1 seconds long, turn my mic on, and get real-time convolution reverb as the mic records incoming audio. My question is: How could this be…
Aaron Dawson
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Estimate the Filter Coefficients of 1D Filtration (Convolution)

I have an output signal $y$ which is an input signal $x$ convolved $\star$ with an impulse response function $h$ with some added noise $n$ : $$y(t) = h(t) \star x(t) + n(t)$$ I know the input signal $x$ and output signal $y$ and would like to…
Leo
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