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LOUD: A 1020-Node Microphone Array and Acoustic Beamformer* Eugene Weinstein 1 , Kenneth Steele 2 , Anant Agarwal 2,3 , James Glass 3 1 Courant Institute of Mathematical Sciences 2 Tilera Corporation 3 MIT Computer Science and Artificial Intelligence Lab * Based on work done at MIT CSAIL

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Page 1: LOUD: A 1020-Node Microphone Array and Acoustic ...groups.csail.mit.edu/cag/mic-array/papers/loud-slides.pdfLarge Microphone Arrays • Large acOUstic Data (LOUD) array: 1020 microphones

LOUD: A 1020-Node Microphone Array and Acoustic Beamformer*Eugene Weinstein1, Kenneth Steele2, Anant Agarwal2,3,

James Glass3

1 Courant Institute of Mathematical Sciences2 Tilera Corporation

3 MIT Computer Science and Artificial Intelligence Lab* Based on work done at MIT CSAIL

Page 2: LOUD: A 1020-Node Microphone Array and Acoustic ...groups.csail.mit.edu/cag/mic-array/papers/loud-slides.pdfLarge Microphone Arrays • Large acOUstic Data (LOUD) array: 1020 microphones

Introduction

• Recording sound in high-noise settings is difficult

• e.g., noisy lab or conference room

• Can use close-talking microphones (e.g., lapel mic)

• However, an untethered solution is more natural

• Idea: use software-steerable microphone arrays

• Isolate and amplify sound using beamforming

• Target application: speech recognition

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Large Microphone Arrays

• Large acOUstic Data (LOUD) array: 1020 microphones

• Microphone array gain increases linearly with the number of microphones

• Past large-array speech recognition experiments scarce

• Processing large quantities of data in real-time is a compelling application for novel computing architectures

• LOUD generates 400 Mbits/sec

• We use Raw, a 16-tile parallel architecture

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Acoustic Beamforming

• Selectively amplify a sound source at a particular location

• Take advantage of sound propagation through space

• Use simple delay-and-sum beamformingSound Source

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Microphones

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Two-microphone PCB

• On-board A/D converter feeds into CPLD

• Data streamed to CPU using time-division multiplexing

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1020-Microphone Array

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Microphone Positions

• Automated procedure to calibrate microphone positions

• Play a test audio “chirp” through a speaker

• Record with reference mic at speaker position and at each array mic

• Peak of cross-correlation function between reference, array microphones gives propagation delay

• Solve for precise array geometry

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Experiments

• Setting: extremely noisy hardware lab

• Subject and “interferer” talking at the same time

• Goal: demonstrate that speech recognition accuracy improves with microphone array size

• Speaker-independent recognizer for digit strings

• Record 150 utterances with interferer, 110 without

• Baseline: high quality close-talking mic, 80 utterances

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Recognition Accuracy

• Word error rate (WER) decreases with array size

• WER drops by 87% (w/ interferer), 91% (no interferer) from one to 1020 mics

• Accuracy approaches close-talking microphone levels!

ICSV14 • 9–12 July 2007 • Cairns • Australia

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Array with interfererArray without interfererClose!talking microphone

Figure 3. Experimental word error rates.

5.1. Discussion

The results in Figures 2 and 3 demonstrate the benefit of using arrays of this size. In this work,

we focused on the system design, and did not implement sophisticated beamforming algorithms

or other signal processing software components. However, even with the simplest beamformer

possible, we were able to obtain increasing SNRs and gains in recognition accuracy from one

to 1020 microphones.

The most drastic jump in the recognition accuracy curve is seen when the number of

microphones goes from 32 to 60. This could be because this completes the full line of the array

(60 microphones), making the beam width almost twice as narrow as with 32 microphones.

After this point, adding more microphones does not make the array wider, just taller. Another

possible reason for the jump is that the landmark component of the recognizer [5] was not

optimized for low SNRs. We note that the WER even with 1020 microphones (12.4% and

8.0%) is clearly significantly short of the 1.2% baseline from the close-talking microphone; and

this is consistent with the SNRs noted in the recordings. However, with more complicated signal

processing and beamforming algorithms and a better match between the recognizer training and

test conditions (see Section 4), it is reasonable to expect that the recognition accuracy of audio

recorded with the array can approach that of a close-talking microphone.

Comparison with past work is difficult for several reasons. One reason is differences in

experimental conditions. Our data were collected in a very noisy environment; likely noisier

than most of the currently-published results. For instance, Moore et al. [12] cite an accuracy

rate of 42.4% with a single omnidirectional microphone and one interferer, compared to our

3.0%; Adcock [1] is at 58%; and Sullivan [20] is at 33%. While SNR is one intuitive way

of comparing noise levels, it is actually difficult to compare based on SNR, since the various

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LOUD Demo

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Summary/Future Work

• LOUD allows high-quality untethered recording in very noisy settings

• Speech recognition experiments demonstrate benefit of large arrays

• Future work:

• Implement more sophisticated beamforming techniques

• Automatic speaker tracking

• Conduct more experiments with different geometries, noise settings

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