ECE8423 8443––Adaptive Pattern Recognition ECE Signal Processing LECTURE 07: TIME-DELAY ESTIMATION AND ADPCM • Objectives: Time-Delay Estimation The LMS Time-Delay Estimator Adaptive Differential PCM • Resources: CNX: Time-Delay Estimation CNX: Beamforming Wiki: Pulse Code Modulation Kurssit: Differential Coding of Images • URL: .../publications/courses/ece_8423/lectures/current/lecture_07.ppt • MP3: .../publications/courses/ece_8423/lectures/current/lecture_07.mp3 Time-Delay Estimation • Our challenge is to estimate the time-delay between two signals. • Applications include radar, sonar, geophysics, biomedical, and audio. • Two measurements are made at sensors P1 and P2 separated by a distance d. • A simple model for the received signals is: x1 (t ) s(t ) v1 (t ) x 2 (t ) s(t D) v 2 (t ) s(t) is the signal which is derived from some distant source, and modeled as traveling at a constant speed. • v1(t) and v2(t) are additive noise terms measured at the receiver and D is the delay. v1(t) and v2(t) are assumed to be zero-mean, stationary, mutually uncorrelated. • We also assume that these measurement noises are uncorrelated with the signal, s(t). (We will relax these assumptions later.) ECE 8423: Lecture 07, Slide 1 Angle of Arrival • The delay, D, can be related to the angle of arrival of the signal: d D sin c where is the arrival of bearing angle to the source, and c is the propagation velocity of the signal through the medium. • The estimation of the bearing angle is reduced to the estimation of the delay, D, given the noisy measurements x1(t) and x2(t). • Note that this approach can be extended to multiple dimensions using multiple sensors. ECE 8423: Lecture 07, Slide 2 Correlation-Based Estimation • The time-delay can be estimated using correlation techniques: rx1x2 Ex1 (t ) x 2 (t ) We can set the time-delay as the peak in the correlation function. • A more rigorous approach is known as the Knapp and Carter generalized correlation method: ry1 y2 w rx1x2 • A maximum likelihood estimate can be derived that achieves the Cramer-Rao lower bound as the estimation time increases. • The ML weighting function can be expressed in terms of the magnitudesquared coherence function (in the frequency domain): msc 12 2 Rx1x2 2 Rx1x1 Rx2 x2 • The ML weighting function can be shown to have the form: Rss / Rvv2 WML 1 (2 Rss / Rvv ) assuming that the noises at each sensor have the same spectra, Rvv(). ECE 8423: Lecture 07, Slide 3 The LMS Time-Delay Estimator • We can formulate this using our standard LMS approach: f n1 f n e(n)x n x2 (n) s(n D) v2 (n) e(n) x2 (n) y(n) x2 (n) f nt x (n1 ) x1 (n) s(n) v1 (n) • The delay is taken as the maximum value of the filter’s impulse response. + f – y (n) e(n) • However, sometimes this peak does not occur exactly at a sample instant, and a more refined estimate of the delay requires interpolation. • However, for simplicity, let’s assume D=kT (integer multiple of samples). • We can write an expression for the error: e(n) x2 (n) y(n) [s(n D) v2 (n) f (n)] [s(n) v1 (n)] • Given that v1(n) and v2(n) are mutually uncorrelated, a good solution might be: f ( n) ( n D ) which produces: e(n) v2 (n) v1 (n D) and E e 2 n 22 12 • But there are drawbacks to this approach. Why? ECE 8423: Lecture 07, Slide 4 Two-Sided Wiener Filter • Recall our two-sided Wiener solution for the LMS filter: Fe j e R x1 x2 e j R x1 x1 j x1 (n) s(n) v1 (n) x 2 ( n) s ( n D ) v 2 ( n ) • In the noise-free case (v1 (n) v2 (n) 0 ): Fe j Rss e j e jD jD e Rss e j f ( n) ( n D ) • A more general solution for non-zero noise and v1 (n) v2 (n) is: Fe j Rss e j e jD j j Rss e Rvv e • To illustrate this process, consider four examples (fs = 1000 Hz, D = 10 samp.): (1) s (n) w(n) H e H e 1 1 (2) s (n) h1 (n) w(n) where H 1 e j 2 / T 1 150 F 350 Hz (3) s (n) h1 (n) w(n) where (4) s (n) h1 (n) w(n) where ECE 8423: Lecture 07, Slide 5 j 2 / T 1 1 j 2 / T 200 F 300 Hz 225 F 275 Hz Examples ECE 8423: Lecture 07, Slide 6 Adaptive Differential Pulse Code Modulation • We can use our LMS filter as a linear predictor to reduce the dynamic range of and compress an audio signal. s (n) – Q f • The error signal can be written as: to receiver sˆ( n) sˆ(n / n 1) e(n) s(n) sˆ(n / n 1) • The error signal is applied to a quantizer: eq (n) Qe(n) + e q (n ) sˆ( n) e q (n ) + from transmitter + • The prediction filter has the form: sˆ(n / n 1) f nt 1e q n f • At the receiver: sˆ(n) sˆ(n / n 1) eq (n) • The filter coefficients are updated using: f n f n-1 e q n 1eq (n) • The challenge in such systems is to allow the filters at the transmitter and receiver to stay in sync without transmitting side information about the filter. ECE 8423: Lecture 07, Slide 7 Echo Cancellation N 1 r (i ) h(k ) y (i k ) k 0 N 1 rˆ(i ) a k y (i k ) k 0 a k (i 1) a k (i ) 2 E[e(i ) y (i k ) a k (i ) 2 M ECE 8423: Lecture 07, Slide 8 M 1 e(i m) y(i m k ) m 0 (i ) 1 2 Py (i ) Py (i ) L y (i ) 2 L y (i 1) (1 ) L y (i ) y (i ) Summary • We demonstrated how to use an LMS filter to estimate the time delay (and source location) of a signal. • We discussed correlation and LMS approaches to this problem. • introduced adaptive differential pulse code modulation as an application of LMS filtering. • Briefly discussed an echo cancellation application. • Next: we will investigate new computational forms of the LMS filter (e.g., leaky gradient descent and lattice filters). ECE 8423: Lecture 07, Slide 9
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