Hilbert Transform Calculator

Paste samples, set rate, and choose processing options. See Hilbert output with clear numerical summaries. Download results, compare cases, and validate spectra easily now.

Hilbert Transform Inputs

Used for time axis and instantaneous frequency.
Reduces spectral leakage before FFT.
Limits on-page rows; downloads store up to 5000.
Ignored when the generator is enabled.
Generator settings
Useful for testing amplitude, phase shift, and noise.
Processing options
These change the signal before the transform.
Moving average length in samples.
If smaller than N, auto expands upward.
Reset
Tip: For a clean envelope and phase, enable mean removal and phase unwrapping.

Example data table

Ten samples from a 5 Hz sinusoid at 100 Hz sampling.
it (s)x
00.00000.000000
10.01000.309017
20.02000.587785
30.03000.809017
40.04000.951057
50.05001.000000
60.06000.951057
70.07000.809017
80.08000.587785
90.09000.309017

Formula used

The continuous Hilbert transform of a real signal \(x(t)\) is the principal-value integral:

H\{x\}(t) = \(\frac{1}{\pi}\) P.V. \(\int_{-\infty}^{\infty} \frac{x(\tau)}{t-\tau} d\tau\)

For discrete samples, this calculator forms the analytic signal using an FFT-based frequency filter:

Z[k] = X[k] · M[k], where M[0]=1, M[k]=2 for 1≤k<N/2, M[N/2]=1 (even N), else 0.

The inverse FFT gives \(z[n]\) where the imaginary part is the Hilbert transform: \(z[n] = x[n] + j\,H\{x\}[n]\). The envelope is \(|z[n]|\), and phase is \(\angle z[n]\). If enabled, instantaneous frequency uses the unwrapped phase derivative: \(f[n] = \frac{\Delta \phi[n]}{2\pi}\,f_s\).

How to use this calculator

  1. Set the sampling rate to match your measured data.
  2. Paste samples, or enable the sinusoid generator.
  3. Choose preprocessing: mean removal, detrend, and window.
  4. Enable phase unwrapping and instantaneous frequency if needed.
  5. Press Compute to view results above the form.
  6. Use CSV or PDF buttons to export the computed table.
Note: For best behavior, use a sampling rate well above the highest frequency.

Hilbert transform guide

1) What the calculator returns

For a real input sequence x[n], the tool builds the analytic signal z[n] = x[n] + j y[n], where y[n] is the discrete Hilbert transform. From z[n] it reports envelope |z[n]|, phase ∠z[n], and optional instantaneous frequency derived from phase change and the sampling rate.

2) Why FFT filtering works

The discrete Hilbert transform can be implemented by suppressing negative-frequency content and doubling positive-frequency content in the spectrum. After multiplying the FFT bins by the mask M[k], the inverse FFT produces a complex-valued signal whose imaginary part is the quadrature component.

3) Sampling rate and usable bandwidth

Accurate envelope and phase require a sampling rate comfortably above the highest frequency present. As a rule of thumb, aim for at least 5–10 samples per cycle of the dominant oscillation. For example, a 5 Hz waveform is well-resolved at 100 Hz (20 samples per cycle).

4) Preprocessing choices

Mean removal prevents DC offsets from dominating the analytic signal. Linear detrending reduces slow drift that can distort phase. Windowing (Hann, Hamming, Blackman) lowers spectral leakage when the record does not contain an integer number of cycles, improving envelope stability for narrowband signals.

5) Edge effects and record length

Hilbert transforms are non-local, so endpoints are sensitive to truncation. Expect the first and last few percent of samples to show larger phase and envelope bias, especially with sharp transients. Increasing duration, applying a smooth window, or zero-padding (larger NFFT) can reduce visible ringing.

6) Envelope interpretation

The envelope |z[n]| tracks amplitude modulation for narrowband signals. In vibration and acoustics it highlights slow amplitude changes; in communications it approximates AM demodulation. For broadband impulses, the envelope is still useful but may blend multiple components into a single magnitude trace.

7) Phase unwrapping and instantaneous frequency

Raw phase is wrapped to (−π, π]. Unwrapping removes 2π jumps, enabling a smooth derivative. Instantaneous frequency is computed as Δφ/(2π)·fs and can be noisy; the smoothing window averages the derivative over multiple samples to emphasize trends rather than sample-to-sample jitter.

8) Practical validation

A quick check is a pure sinusoid: the Hilbert transform should be a 90°-shifted cosine/sine pair, the envelope should be approximately constant, and the instantaneous frequency should cluster near the set tone. Add controlled noise to evaluate robustness and tune smoothing length.

FAQs

1) What is the Hilbert transform used for?

It creates a quadrature companion signal, enabling analytic-signal methods such as envelope detection, instantaneous phase, and instantaneous frequency. It is common in vibration analysis, radar/sonar, biomedical time series, and communications.

2) Why does the result look unstable at the ends?

Endpoint samples suffer from truncation because the transform is non-local. Use longer records, apply a smooth window, and interpret the first/last few percent cautiously. Zero-padding can also reduce visible ringing.

3) Should I remove the mean or detrend?

Yes when offsets or drift exist. Mean removal prevents DC bias from inflating the envelope. Detrending helps when slow ramps are present, improving phase behavior and making instantaneous frequency more meaningful.

4) When should I choose a window?

Use a window when your data segment does not align with whole cycles or when leakage contaminates nearby frequencies. Hann and Hamming are good general choices; Blackman offers stronger sidelobe suppression with more mainlobe widening.

5) What does zero-padding (larger NFFT) change?

It increases the FFT grid density and can smooth the reconstructed analytic signal, but it does not add new information. It may reduce visual artifacts and help with interpolation, especially when analyzing narrowband signals.

6) Why is instantaneous frequency noisy?

It depends on a phase derivative, which amplifies noise and sharp transitions. Enable phase unwrapping and use a smoothing window. For highly broadband or impulsive signals, interpret instantaneous frequency as a qualitative indicator.

7) How can I sanity-check the output?

Try a clean sinusoid: the envelope should be nearly constant and the frequency estimate should match the tone. Then add small noise to see how much smoothing is needed for stable trends.

Hilbert transform article

1) Analytic signal in practice

This calculator constructs the analytic signal z[n] = x[n] + j* y[n], where y[n] is the discrete Hilbert transform. From z[n] it reports the envelope |z[n]| and the phase arg(z[n]). These outputs are commonly used to estimate amplitude modulation, track phase, and compute instantaneous frequency.

2) FFT method and filter mask

The discrete transform is computed with an FFT: the spectrum X[k] is multiplied by a mask that doubles positive-frequency bins, keeps DC (and Nyquist for even lengths), and zeros negative frequencies. After the inverse FFT, the imaginary part becomes y[n].

3) Sampling rate and time axis

Enter a correct sampling rate f_s to interpret time and frequency. The output time step is dt = 1/f_s and the instantaneous frequency option uses f[n] = (f_s/2pi) * dphi/dn. For example, a 5 Hz tone sampled at 100 Hz should produce a near-constant 5 Hz estimate away from edges.

4) Edge effects and padding

FFT processing assumes the record repeats, so sharp start or end points can leak into the envelope and phase. Windowing reduces this leakage, while a larger NFFT (zero-padding) refines the frequency grid and can make the quadrature estimate smoother. Padding does not add new information, but it can improve interpolation of the analytic signal.

2) FFT method and filter mask

The discrete transform is computed with an FFT: the spectrum X[k] is multiplied by a mask that doubles positive-frequency bins, keeps DC (and Nyquist for even lengths), and zeros negative frequencies. After the inverse FFT, the imaginary part becomes y[n].

3) Sampling rate and frequency units

Set the sampling rate carefully because time step is dt = 1/fs. Instantaneous frequency is reported in hertz using the phase difference f[n] = (dphi/(2*pi))*fs. For example, a 5 Hz tone sampled at 100 Hz should yield a stable 5 Hz estimate away from edges.

4) Edge effects and padding

FFT processing assumes the record repeats. If your signal is not periodic within the window, discontinuities at the ends can leak energy and distort the envelope and phase near the boundaries. Zero padding increases frequency sampling and can smooth plots, but it does not remove boundary bias.

5) Preprocessing improves stability

Mean removal suppresses a large DC component that can dominate the analytic magnitude. Linear detrending reduces slow ramps that inflate low-frequency bins. Windowing (Hann, Hamming, or Blackman) reduces spectral leakage, which is useful when the record length is short or contains sharp transitions.

6) Envelope and amplitude modulation

The envelope |z[n]| tracks slowly varying amplitude. A common test is an AM signal such as x(t) = (1 + 0.3*cos(2*pi*2*t))*cos(2*pi*20*t). With a sampling rate above 40 Hz, the envelope should follow 1 + 0.3*cos(2*pi*2*t) after transients.

7) Phase, unwrapping, and instantaneous frequency

The phase of z[n] can jump by plus or minus pi when it crosses the branch cut. Unwrapping removes these discontinuities so the phase derivative is meaningful. Instantaneous frequency is sensitive to noise, so the optional moving-average smoothing can reduce jitter.

8) Validation and typical uses

Validate using a pure sinusoid: the Hilbert transform should be the same sinusoid shifted by 90 degrees, and the envelope should be constant. In practice, analytic envelopes are used in vibration diagnostics, seismology, radar, and biomedical time series to highlight bursts and estimate local oscillation rates.

FAQs

1) Why does the envelope look wrong near the start or end?

FFT processing assumes the record repeats. End discontinuities create leakage and bias the Hilbert pair. Try windowing, detrending, longer records, or ignore a short edge region.

2) Does zero padding change the actual transform?

Padding increases FFT length and interpolates the frequency grid. It can make curves smoother and reduce wraparound artifacts, but it does not add new information or fix boundary discontinuities.

3) What sampling rate should I use?

Use a sampling rate comfortably above the highest frequency of interest. As a rule of thumb, 5 to 10 samples per cycle improves envelope and phase stability and reduces instantaneous-frequency noise.

4) When should I enable phase unwrapping?

Enable it when you need a smooth phase trace or instantaneous frequency. Without unwrapping, phase jumps by plus or minus pi can create large spikes in the derivative.

5) Why remove the mean or detrend?

A strong DC offset and slow ramps concentrate power at very low frequencies and can dominate the analytic magnitude. Removing them often yields a cleaner envelope and more interpretable phase.

6) Can I use this on non-stationary signals?

Yes, but interpret results locally. Sudden transients and wideband content can make instantaneous frequency unstable. Smoothing and appropriate windowing help, and longer records reduce variance.

7) How can I sanity-check the results quickly?

Test with a clean sinusoid. The Hilbert output should be the same sinusoid shifted by 90 degrees, the envelope should be constant, and the instantaneous frequency should match the tone value away from edges.

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