Array signal processing uses multiple sensors (microphones, hydrophones, antennas, seismic geophones) to spatially locate and enhance signals. A linear or planar array of sensors receives signals from different directions with phase differences determined by geometry and signal wavelength. Beamforming steers the array's sensitivity to emphasize signals from a target direction while suppressing interference. Conventional beamforming uses fixed weights (delay-and-sum); adaptive beamforming (MVDR/Capon, LCMV) minimizes output power subject to maintaining a unit gain in the target direction. Direction-of-arrival (DOA) estimation (MUSIC, ESPRIT) identifies source locations without steering, using spectral factorization or subspace methods. Applications include radar, sonar, radio astronomy, and acoustic source localization.
Simulate a line array receiving signals from multiple sources (targets + interference) at different angles of arrival. Implement delay-and-sum beamforming and observe how the array's beampattern (gain vs. angle) varies with frequency and number of sensors. Implement adaptive Capon beamforming and observe gain in the target direction while nulling interference. Estimate DOA using MUSIC algorithm (compute spatial correlation matrix, perform eigendecomposition, search for angles that maximize noise subspace projection). Validate on synthetic and real data (ship engine noise in sonar, radio astronomy).
Imagine a row of microphones recording sound from a distant speaker. The sound wavefront arrives at each microphone with a slight time delay — it hits the microphone closest to the speaker first, then the next one, and so on. This phase difference between microphones contains information about the source direction. Array signal processing exploits this: by carefully combining (weighting and summing) signals from all microphones, you can focus the array's "hearing" in a chosen direction, suppressing sound from other directions.
Delay-and-sum beamforming is the foundation. Compute the expected phase shift φ₀ = (2πd/λ)sin(θ₀) that a signal from angle θ₀ would induce between adjacent sensors (d = sensor spacing, λ = wavelength). Apply phase shifts −φ₀ to each sensor (called "steering"), sum them. Signals arriving from θ₀ add constructively (their phases align); signals from other angles add incoherently (phases cancel partially). The array gain is the sum of all weights (M microphones), so SNR improves by a factor of roughly M. The beampattern (gain vs. angle) is the array's spatial filter: it has a main lobe (high gain) pointing at θ₀ and side lobes (partial suppression of other angles). More sensors give narrower main lobes and better sidelobe suppression.
Adaptive beamforming goes further: instead of using a fixed beampattern, adapt the weights to minimize interference and noise while maintaining the target signal. The MVDR (Minimum Variance Distortionless Response) beamformer solves: minimize w^H R_xx w (total output power) subject to w^H a(θ₀) = 1 (unit gain in target direction), where R_xx is the estimated input covariance and a(θ₀) is the steering vector. The solution is w* = R_xx^{-1} a(θ₀) / (a(θ₀)^H R_xx^{-1} a(θ₀)). The constraint ensures the target signal is undistorted; minimizing power simultaneously suppresses interference and noise. In high-interference environments (e.g., radar with jamming), MVDR can null multiple interferers by producing deep nulls in the beampattern while maintaining the main lobe. The cost: computational (matrix inversion, O(M³)) and sensitivity to model error (if the covariance or steering vector is misestimated, the optimizer can fail catastrophically, even self-nulling the target).
Direction-of-arrival (DOA) estimation identifies source locations without steering the beam. The MUSIC algorithm uses eigendecomposition: decompose R_xx into signal and noise subspaces (K largest eigenvectors are signal subspace for K sources). The steering vector a(θ) for any angle lies in the signal subspace at true source directions, hence is orthogonal to the noise subspace. Search over angles to maximize the "music spectrum" P(θ) = 1 / ||a(θ)^H P_noise a(θ)||, where P_noise projects onto the noise subspace. At true source angles, the spectrum has sharp peaks. MUSIC is computationally expensive (eigendecomposition, angle search) but does not require steering vector matching — it identifies sources blindly.
Applications span:
The limits of array signal processing are fundamental: wavelength limits spatial resolution (smaller wavelengths → finer angles), and array aperture limits gain (larger aperture → more sensors → more gain). Cross-coupling between these and limited data (finite samples for covariance estimation) mean DOA resolution and sidelobe suppression are limited. Modern extensions use sparse arrays (nonuniform spacing, fewer elements for equivalent aperture), learned beamformers (neural networks training on data), and tensor methods (multi-dimensional signal processing for multi-frequency or time-varying scenarios).
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