Digital Audio Signal Processing Lecture-2 Microphone Array Processing

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Presentation transcript:

Digital Audio Signal Processing Lecture-2 Microphone Array Processing Marc Moonen Dept. E.E./ESAT-STADIUS, KU Leuven marc.moonen@esat.kuleuven.be homes.esat.kuleuven.be/~moonen/

Overview Introduction & beamforming basics Fixed beamforming Data model & definitions Fixed beamforming Filter-and-sum beamformer design Matched filtering White noise gain maximization Ex: Delay-and-sum beamforming Superdirective beamforming Directivity maximization Directional microphones (delay-and-subtract) Adaptive Beamforming LCMV beamformer Generalized sidelobe canceler

Introduction Directivity pattern of a microphone A microphone (*) is characterized by a `directivity pattern which specifies the gain & phase shift that the microphone gives to a signal coming from a certain direction (i.e. `angle-of-arrival’) In general the directivity pattern is a function of frequency (ω) In a 3D scenario `angle-of-arrival’ is azimuth + elevation angle Will consider only 2D scenarios for simplicity, with one angle-of arrival (θ), hence directivity pattern is H(ω,θ) Directivity pattern is fixed and defined by physical microphone design (*) We do digital signal prcessing, so this includes front-end filtering/A-to-D/.. |H(ω,θ)| for 1 frequency

Introduction + Virtual directivity pattern : By weighting or filtering (=freq. dependent weighting) and then summing signals from different microphones, a (software controlled) virtual directivity pattern (=weigthed sum of individual patterns) can be produced This assumes all microphones receive the same signals (so are all in the same position). However… + :

Introduction However, in a microphone array different microphones are in different positions/locations, hence also receive different signals Example : uniform linear array i.e. microphones placed on a line & uniform inter-micr. distances (d) & ideal micr. characteristics (p.9) For a far-field source signal (plane waveforms), each microphone receives the same signal, up to an angle-dependent delay… fs=sampling rate c=propagation speed + :

Introduction + Beamforming = `spatial filtering’ based on microphone characteristics (directivity patterns) AND microphone array configuration (`spatial sampling’) Classification: Fixed beamforming: data-independent, fixed filters Fm e.g. delay-and-sum, filter-and-sum Adaptive beamforming: data-dependent filters Fm e.g. LCMV-beamformer, generalized sidelobe canceler + :

Introduction Background/history: ideas borrowed from antenna array design and processing for radar & (later) wireless communications Microphone array processing considerably more difficult than antenna array processing: narrowband radio signals versus broadband audio signals far-field (plane wavefronts) versus near-field (spherical wavefronts) pure-delay environment versus multi-path environment Applications: voice controlled systems (e.g. Xbox Kinect), speech communication systems, hearing aids,…

Data model and definitions Data model: source signal in far-field (see p.13 for near-field) Microphone signals are filtered versions of source signal S() at angle  Stack all microphone signals (m=1..M) in a vector d is `steering vector’ Output signal after `filter-and-sum’ is H instead of T for convenience (**)

Data model and definitions Data model: source signal in far-field If all microphones have the same directivity pattern Ho(ω,θ), steering vector can be factored as… Will often consider arrays with ideal omni-directional microphones : Ho(ω,θ)=1 Example : uniform linear array, see p.5 microphone-1 is used as a reference (=arbitrary)

Data model and definitions In a linear array (p.5) :  =90o=broadside direction  = 0o =end-fire direction Array directivity pattern (compare to p.3) = `transfer function’ for source at angle  ( -π<< π ) Steering direction = angle  with maximum amplification (for 1 freq.) Beamwidth (BW) = region around max with (e.g.) amplification > -3dB (for 1 freq.)

Data model and definitions Data model: source signal + noise Microphone signals are corrupted by additive noise Define noise correlation matrix as Will assume noise field is homogeneous, i.e. all diagonal elements of noise correlation matrix are equal : Then noise coherence matrix is

Data model and definitions Array Gain = improvement in SNR for source at angle  ( -π<< π ) White Noise Gain =array gain for spatially uncorrelated noise (e.g. sensor noise) ps: often used as a measure for robustness Directivity =array gain for diffuse noise (=coming from all directions) DI and WNG evaluated at max is often used as a performance criterion |signal transfer function|^2 |noise transfer function|^2 skip this formula PS: ω is rad/sample ( -Π≤ω≤Π ) ω fs is rad/sec

PS: Near-field beamforming Far-field assumptions not valid for sources close to microphone array spherical wavefronts instead of planar waveforms include attenuation of signals 2 coordinates ,r (=position q) instead of 1 coordinate  (in 2D case) Different steering vector (e.g. with Hm(ω,θ)=1 m=1..M) : e e=1 (3D)…2 (2D) with q position of source pref position of reference microphone pm position of mth microphone

PS: Multipath propagation In a multipath scenario, acoustic waves are reflected against walls, objects, etc.. Every reflection may be treated as a separate source (near-field or far-field) A more realistic data model is then.. with q position of source and Hm(ω,q), complete transfer function from source position to m-the microphone (incl. micr. characteristic, position, and multipath propagation) `Beamforming’ aspect vanishes here, see also Lecture-3 (`multi-channel noise reduction’)

Overview Introduction & beamforming basics Fixed beamforming Data model & definitions Fixed beamforming Filter-and-sum beamformer design Matched filtering White noise gain maximization Ex: Delay-and-sum beamforming Superdirective beamforming Directivity maximization Directional microphones (delay-and-subtract) Adaptive Beamforming LCMV beamformer Generalized sidelobe canceler

Filter-and-sum beamformer design Basic: procedure based on page 10 Array directivity pattern to be matched to given (desired) pattern over frequency/angle range of interest Non-linear optimization for FIR filter design (=ignore phase response) Quadratic optimization for FIR filter design (=co-design phase response)

Filter-and-sum beamformer design Quadratic optimization for FIR filter design (continued) With optimal solution is Kronecker product

Filter-and-sum beamformer design Design example M=8 Logarithmic array N=50 fs=8 kHz

Matched filtering: WNG maximization Basic: procedure based on page 12 Maximize White Noise Gain (WNG) for given steering angle ψ A priori knowledge/assumptions: angle-of-arrival ψ of desired signal + corresponding steering vector noise scenario = white

Matched filtering: WNG maximization Maximization in is equivalent to minimization of noise output power (under white input noise), subject to unit response for steering angle (**) Optimal solution (`matched filter’) is [FIR approximation]

Matched filtering example: Delay-and-sum Basic: Microphone signals are delayed and then summed together Fractional delays implemented with truncated interpolation filters (=FIR) Consider array with ideal omni-directional micr’s Then array can be steered to angle  : Hence (for ideal omni-dir. micr.’s) this is matched filter solution

Matched filtering example: Delay-and-sum ideal omni-dir. micr.’s Array directivity pattern H(ω,θ): =destructive interference =constructive interference White noise gain : (independent of ω) For ideal omni-dir. micr. array, delay-and-sum beamformer provides WNG equal to M for all freqs. (in the direction of steering angle ψ).

Matched filtering example: Delay-and-sum ideal omni-dir. micr.’s Array directivity pattern H(,) for uniform linear array: H(,) has sinc-like shape and is frequency-dependent M=5 microphones d=3 cm inter-microphone distance =60 steering angle fs=16 kHz sampling frequency =endfire =60 wavelength=4cm

Matched filtering example: Delay-and-sum ideal omni-dir. micr.’s For an ambiguity, called spatial aliasing, occurs. This is analogous to time-domain aliasing where now the spatial sampling (=d) is too large. Aliasing does not occur (for any ) if M=5, =60, fs=16 kHz, d=8 cm

Matched filtering example: Delay-and-sum ideal omni-dir. micr.’s Beamwidth for a uniform linear array: hence large dependence on # microphones, distance (compare p.24 & 25) and frequency (e.g. BW infinitely large at DC) Array topologies: Uniformly spaced arrays Nested (logarithmic) arrays (small d for high , large d for small ) 2D- (planar) / 3D-arrays with e.g. =1/sqrt(2) (-3 dB)

Super-directive beamforming : DI maximization Basic: procedure based on page 12 Maximize Directivity (DI) for given steering angle ψ A priori knowledge/assumptions: angle-of-arrival ψ of desired signal + corresponding steering vector noise scenario = diffuse

Super-directive beamforming : DI maximization Maximization in is equivalent to minimization of noise output power (under diffuse input noise), subject to unit response for steering angle (**) Optimal solution is [FIR approximation]

Super-directive beamforming : DI maximization ideal omni-dir. micr.’s Directivity patterns for end-fire steering (ψ=0): Superdirective beamformer has highest DI, but very poor WNG (at low frequencies, where diffuse noise coherence matrix becomes ill-conditioned) hence problems with robustness (e.g. sensor noise) ! M=5 d=3 cm fs=16 kHz WNG=M= 5 PS: diffuse noise ≈ white noise for high frequencies (cfr. ωΠ and c/fs=λmin/2≈min(dj-di) in diffuse noise coherence matrix) DI=WNG=5 DI=M 2=25 Maximum directivity=M.M obtained for end-fire steering and for frequency->0 (no proof)

Differential microphones : Delay-and-subtract First-order differential microphone = directional microphone 2 closely spaced microphones, where one microphone is delayed (=hardware) and whose outputs are then subtracted from each other Array directivity pattern: First-order high-pass frequency dependence P() = freq.independent (!) directional response 0  1  1 : P() is scaled cosine, shifted up with 1 such that max = 0o (=end-fire) and P(max )=1 d/c <<,  <<

Differential microphones : Delay-and-subtract Types: dipole, cardioid, hypercardioid, supercardioid (HJ84) =endfire =broadside Dipole: 1= 0 (=0) zero at 90o DI=4.8dB Hypercardioid: 1= 0.25 zero at 109o highest DI=6.0dB Supercardioid: 1= zero at 125o, DI=5.7 dB highest front-to-back ratio Cardioid: 1= 0.5 zero at 180o DI=4.8dB

Overview Introduction & beamforming basics Fixed beamforming Data model & definitions Fixed beamforming Filter-and-sum beamformer design Matched filtering White noise gain maximization Ex: Delay-and-sum beamforming Superdirective beamforming Directivity maximization Directional microphones (delay-and-subtract) Adaptive Beamforming LCMV beamformer Generalized sidelobe canceler

LCMV beamformer + Adaptive filter-and-sum structure: : Aim is to minimize noise output power, while maintaining a chosen response in a given look direction (and/or other linear constraints, see below). This is similar to operation of a superdirective array (in diffuse noise), or delay-and-sum (in white noise), cfr. (**) p.20 & 27, but now noise field is unknown ! Implemented as adaptive FIR filter : + :

LCMV beamformer LCMV = Linearly Constrained Minimum Variance f designed to minimize output power (variance) (read on..) z[k] : To avoid desired signal cancellation, add (J) linear constraints Example: fix array response in look-direction ψ for sample freqs ωi For sufficiently large J constrained output power minimization approximately corresponds to constrained output noise power minimization(why?) Solution is (obtained using Lagrange-multipliers, etc..):

Generalized sidelobe canceler (GSC) GSC = Adaptive filter formulation of the LCMV-problem Constrained optimisation is reformulated as a constraint pre-processing, followed by an unconstrained optimisation, leading to a simple adaptation scheme LCMV-problem is Define `blocking matrix’ Ca, ,with columns spanning the null-space of C Define ‘quiescent response vector’ fq satisfying constraints Parametrize all f’s that satisfy constraints (verify!) I.e. filter f can be decomposed in a fixed part fq and a variable part Ca. fa

Generalized sidelobe canceler GSC = Adaptive filter formulation of the LCMV-problem Constrained optimisation is reformulated as a constraint pre-processing, followed by an unconstrained optimisation, leading to a simple adaptation scheme LCMV-problem is Unconstrained optimization of fa : (MN-J coefficients)

Generalized sidelobe canceler GSC (continued) Hence unconstrained optimization of fa can be implemented as an adaptive filter (adaptive linear combiner), with filter inputs (=‘left- hand sides’) equal to and desired filter output (=‘right-hand side’) equal to LMS algorithm :

Generalized sidelobe canceler GSC then consists of three parts: Fixed beamformer (cfr. fq ), satisfying constraints but not yet minimum variance), creating `speech reference’ Blocking matrix (cfr. Ca), placing spatial nulls in the direction of the speech source (at sampling frequencies) (cfr. C’.Ca=0), creating `noise references’ Multi-channel adaptive filter (linear combiner) your favourite one, e.g. LMS

Generalized sidelobe canceler A popular & significantly cheaper GSC realization is as follows Note that some reorganization has been done: the blocking matrix now generates (typically) M-1 (instead of MN-J) noise references, the multichannel adaptive filter performs FIR-filtering on each noise reference (instead of merely scaling in the linear combiner). Philosophy is the same, mathematics are different (details on next slide). Postproc y1 yM

Generalized sidelobe canceler Math details: (for Delta’s=0) select `sparse’ blocking matrix such that : =input to multi-channel adaptive filter =use this as blocking matrix now

Generalized sidelobe canceler Blocking matrix Ca (cfr. scheme page 24) Creating (M-1) independent noise references by placing spatial nulls in look-direction different possibilities (broadside steering) Griffiths-Jim Problems of GSC: impossible to reduce noise from look-direction reverberation effects cause signal leakage in noise references adaptive filter should only be updated when no speech is present to avoid signal cancellation! Walsh