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Rmd 070a399 Dave Tang 2026-02-02 Introduction to the Fourier Transform

What is the Fourier Transform?

The Fourier Transform is a mathematical technique that decomposes a signal into its constituent frequencies. Named after French mathematician Jean-Baptiste Joseph Fourier, it transforms data from the time domain (or spatial domain) into the frequency domain.

If you hear a musical chord (when you play three or more notes simultaneously to create a harmonic sound), the Fourier Transform can tell you which individual notes make up that chord and how loud each note is.

Why is it useful?

The Fourier Transform is fundamental in many fields:

  • Signal processing: Filtering noise from audio or images
  • Bioinformatics: Analysing periodic patterns in DNA sequences, spectroscopy data
  • Image processing: JPEG compression, edge detection
  • Physics: Quantum mechanics, optics, acoustics

The intuition

Any periodic signal can be represented as a sum of sine and cosine waves of different frequencies. The Fourier Transform finds the amplitude and phase of each frequency component.

Let’s start with a simple example: creating a signal from known frequencies.

  • Hertz is a unit that measures frequency: how many times something happens per second.
  • 100 Hz means something is happening 100 times per second.
# Create a time vector (1 second of data sampled at 100 Hz)
sample_rate <- 100
duration <- 1
t <- seq(0, duration, by = 1/sample_rate)

# Create a signal with two frequencies: 5 Hz and 20 Hz
freq1 <- 5
freq2 <- 20

signal <- sin(2 * pi * freq1 * t) + 0.5 * sin(2 * pi * freq2 * t)

plot(t, signal, type = "l",
     xlab = "Time (seconds)",
     ylab = "Amplitude",
     main = "Signal with 5 Hz and 20 Hz components")

Looking at this signal, it’s not immediately obvious that it’s composed of two sine waves. Let’s use the Fourier Transform to reveal the hidden frequencies.

The Fast Fourier Transform (FFT) in R

R provides the fft() function which implements the Fast Fourier Transform, an efficient algorithm for computing the Discrete Fourier Transform (DFT).

# Compute the FFT
fft_result <- fft(signal)

# The FFT returns complex numbers
head(fft_result)
[1] -8.437695e-15+ 0.0000000i  7.025444e-03- 0.2257903i
[3]  3.167017e-02- 0.5084300i  9.109146e-02- 0.9733405i
[5]  2.734064e-01- 2.1861049i  7.768581e+00-49.5475020i

The FFT returns complex numbers. The magnitude (absolute value) tells us the amplitude of each frequency component. The phase (argument) tells us the phase shift.

# Calculate magnitude
magnitude <- Mod(fft_result)

# Create frequency axis
n <- length(signal)
freq <- (0:(n-1)) * sample_rate / n

# Plot only the first half (positive frequencies)
# The FFT output is symmetric for real signals
half_n <- floor(n/2)

plot(freq[1:half_n], magnitude[1:half_n], type = "h",
     xlab = "Frequency (Hz)",
     ylab = "Magnitude",
     main = "Frequency Spectrum")

Notice the peaks at 5 Hz and 20 Hz - exactly the frequencies we used to create our signal! The peak at 5 Hz is twice as high as the peak at 20 Hz, reflecting the amplitude ratio (1.0 vs 0.5) in our original signal.

Normalising the FFT output

To get meaningful amplitude values, we need to normalise the FFT output.

# Normalise by the number of samples
# Multiply by 2 for single-sided spectrum (except DC component)
magnitude_normalised <- (2 * Mod(fft_result) / n)[1:half_n]
magnitude_normalised[1] <- magnitude_normalised[1] / 2  # DC component

plot(freq[1:half_n], magnitude_normalised, type = "h",
     xlab = "Frequency (Hz)",
     ylab = "Amplitude",
     main = "Normalised Frequency Spectrum")

# Add points to highlight peaks
abline(h = 0, col = "gray")

Now the amplitudes approximately match our original signal (1.0 for 5 Hz and 0.5 for 20 Hz).

A practical example: Removing noise

One common application is filtering noise from a signal. Let’s add noise to our signal and then filter it out.

# Add random noise
set.seed(42)
noisy_signal <- signal + rnorm(length(signal), sd = 0.5)

par(mfrow = c(2, 1), mar = c(4, 4, 2, 1))
plot(t, signal, type = "l",
     xlab = "Time (seconds)", ylab = "Amplitude",
     main = "Original Signal")
plot(t, noisy_signal, type = "l",
     xlab = "Time (seconds)", ylab = "Amplitude",
     main = "Noisy Signal")

par(mfrow = c(1, 1))
# FFT of noisy signal
fft_noisy <- fft(noisy_signal)

# Create a low-pass filter (keep frequencies below 25 Hz)
cutoff <- 25
filter_mask <- rep(0, n)
freq_indices <- which(freq <= cutoff | freq >= (sample_rate - cutoff))
filter_mask[freq_indices] <- 1

# Apply filter
fft_filtered <- fft_noisy * filter_mask

# Inverse FFT to get back to time domain
filtered_signal <- Re(fft(fft_filtered, inverse = TRUE) / n)

par(mfrow = c(2, 1), mar = c(4, 4, 2, 1))
plot(t, noisy_signal, type = "l", col = "gray",
     xlab = "Time (seconds)", ylab = "Amplitude",
     main = "Noisy vs Filtered Signal")
lines(t, filtered_signal, col = "blue", lwd = 2)
legend("topright", legend = c("Noisy", "Filtered"),
       col = c("gray", "blue"), lty = 1, lwd = c(1, 2))

plot(t, signal, type = "l", col = "black",
     xlab = "Time (seconds)", ylab = "Amplitude",
     main = "Original vs Filtered Signal")
lines(t, filtered_signal, col = "blue", lwd = 2, lty = 2)
legend("topright", legend = c("Original", "Filtered"),
       col = c("black", "blue"), lty = c(1, 2), lwd = c(1, 2))

par(mfrow = c(1, 1))

Understanding the mathematics

The Discrete Fourier Transform is defined as:

\[X_k = \sum_{n=0}^{N-1} x_n \cdot e^{-i 2\pi k n / N}\]

Where:

  • \(x_n\) is the input signal at time point \(n\)
  • \(X_k\) is the complex amplitude at frequency \(k\)
  • \(N\) is the total number of samples
  • \(e^{-i 2\pi k n / N}\) represents complex sinusoids (from Euler’s formula: \(e^{i\theta} = \cos\theta + i\sin\theta\))

The inverse transform reconstructs the original signal:

\[x_n = \frac{1}{N} \sum_{k=0}^{N-1} X_k \cdot e^{i 2\pi k n / N}\]

Computing DFT manually

Let’s verify our understanding by computing the DFT manually for a small signal.

# Simple signal
x <- c(1, 2, 3, 4)
N <- length(x)

# Manual DFT
manual_dft <- function(x) {
  N <- length(x)
  X <- complex(N)
  for (k in 0:(N-1)) {
    for (n in 0:(N-1)) {
      X[k+1] <- X[k+1] + x[n+1] * exp(-1i * 2 * pi * k * n / N)
    }
  }
  return(X)
}

# Compare manual vs fft()
manual_result <- manual_dft(x)
fft_result <- fft(x)

data.frame(
  k = 0:(N-1),
  manual_real = round(Re(manual_result), 6),
  manual_imag = round(Im(manual_result), 6),
  fft_real = round(Re(fft_result), 6),
  fft_imag = round(Im(fft_result), 6)
)
  k manual_real manual_imag fft_real fft_imag
1 0          10           0       10        0
2 1          -2           2       -2        2
3 2          -2           0       -2        0
4 3          -2          -2       -2       -2

The results match! The FFT is simply a faster algorithm (\(O(N \log N)\) vs \(O(N^2)\)) for computing the same thing.

Power spectrum

The power spectrum shows the distribution of signal power across frequencies. It’s computed as the squared magnitude of the FFT.

# Create a more complex signal
t_long <- seq(0, 2, by = 1/sample_rate)
complex_signal <- sin(2 * pi * 3 * t_long) +
                  0.7 * sin(2 * pi * 7 * t_long) +
                  0.3 * sin(2 * pi * 15 * t_long)

# Compute power spectrum
fft_complex <- fft(complex_signal)
n_long <- length(complex_signal)
power <- (Mod(fft_complex)^2) / n_long
freq_long <- (0:(n_long-1)) * sample_rate / n_long

# Plot
half_n_long <- floor(n_long/2)
plot(freq_long[1:half_n_long], power[1:half_n_long], type = "h",
     xlab = "Frequency (Hz)",
     ylab = "Power",
     main = "Power Spectrum")

Summary

Key takeaways:

  1. The Fourier Transform converts signals from time domain to frequency domain
  2. Use fft() in R for the Fast Fourier Transform
  3. The FFT returns complex numbers; use Mod() for magnitude, Arg() for phase
  4. For real signals, the FFT output is symmetric - only the first half is needed
  5. Remember to normalise by dividing by the number of samples

sessionInfo()
R version 4.5.0 (2025-04-11)
Platform: x86_64-pc-linux-gnu
Running under: Ubuntu 24.04.3 LTS

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