Optical Geolocation for Small Unmanned Aerial Systems

This paper presents an airborne optical geolocation system using four optical targets to provide position and attitude estimation for a sUAS supporting the NASA Acoustic Research Mission (ARM), where the goal is to reduce nuisance airframe noise during approach and landing. A large precision positioned microphone array captures the airframe noise for multiple passes of a Gulfstream III aircraft. For health monitoring of the microphone array, the Acoustic Calibration Vehicle (ACV) sUAS completes daily flights with an onboard speaker emitting tones at frequencies optimized for determining microphone functionality. An accurate position estimate of the ACV relative to the array is needed for microphone health monitoring. To this end, an optical geolocation system using a downward facing camera mounted to the ACV was developed. The 3D positioning of the ACV is computed using the pinhole camera model. A novel optical geolocation algorithm first detects the targets, then a recursive algorithm tightens the localization of the targets. Finally, the position of the sUAS is computed using the image coordinates of the targets, the 3D world coordinates of the targets, and the camera matrix. A Real-Time Kinematic GPS system is used to compare the optical geolocation system

The 2005 AMI system for the transcription of speech in meetings

In this paper we describe the 2005 AMI system for the transcription\ud of speech in meetings used for participation in the 2005 NIST\ud RT evaluations. The system was designed for participation in the speech\ud to text part of the evaluations, in particular for transcription of speech\ud recorded with multiple distant microphones and independent headset\ud microphones. System performance was tested on both conference room\ud and lecture style meetings. Although input sources are processed using\ud different front-ends, the recognition process is based on a unified system\ud architecture. The system operates in multiple passes and makes use\ud of state of the art technologies such as discriminative training, vocal\ud tract length normalisation, heteroscedastic linear discriminant analysis,\ud speaker adaptation with maximum likelihood linear regression and minimum\ud word error rate decoding. In this paper we describe the system performance\ud on the official development and test sets for the NIST RT05s\ud evaluations. The system was jointly developed in less than 10 months\ud by a multi-site team and was shown to achieve very competitive performance

Acoustic Vector-Corrected Impedance Meter

We describe the development of a novel instrument intended for the measurement of the acoustical reflection coefficient of materials. The instrument effectively implements a one-port vector-corrected network analyzer in the acoustic, rather than the electromagnetic, domain. Employing the well-documented methods of error correction familiar to microwave engineers, this instrument permits automated measurement of an acoustic impedance presented to a waveguide port. A dual-directional coupler allows a working frequency range of well over an octave. In principle, a set of six couplers would allow measurement from 100 to 50000 Hz

Raking the Cocktail Party

We present the concept of an acoustic rake receiver---a microphone beamformer that uses echoes to improve the noise and interference suppression. The rake idea is well-known in wireless communications; it involves constructively combining different multipath components that arrive at the receiver antennas. Unlike spread-spectrum signals used in wireless communications, speech signals are not orthogonal to their shifts. Therefore, we focus on the spatial structure, rather than temporal. Instead of explicitly estimating the channel, we create correspondences between early echoes in time and image sources in space. These multiple sources of the desired and the interfering signal offer additional spatial diversity that we can exploit in the beamformer design. We present several "intuitive" and optimal formulations of acoustic rake receivers, and show theoretically and numerically that the rake formulation of the maximum signal-to-interference-and-noise beamformer offers significant performance boosts in terms of noise and interference suppression. Beyond signal-to-noise ratio, we observe gains in terms of the \emph{perceptual evaluation of speech quality} (PESQ) metric for the speech quality. We accompany the paper by the complete simulation and processing chain written in Python. The code and the sound samples are available online at \url{http://lcav.github.io/AcousticRakeReceiver/}.Comment: 12 pages, 11 figures, Accepted for publication in IEEE Journal on Selected Topics in Signal Processing (Special Issue on Spatial Audio

On the potential of channel selection for recognition of reverberated speech with multiple microphones

The performance of ASR systems in a room environment with distant microphones is strongly affected by reverberation. As the degree of signal distortion varies among acoustic channels (i.e. microphones), the recognition accuracy can benefit from a proper channel selection. In this paper, we experimentally show that there exists a large margin for WER reduction by channel selection, and discuss several possible methods which do not require any a-priori classification. Moreover, by using a LVCSR task, a significant WER reduction is shown with a simple technique which uses a measure computed from the sub-band time envelope of the various microphone signals.Peer ReviewedPreprin

Relative Sound Localization for Sources in a Haphazard Speaker Array

A rapidly deployable, easy to use method of automatically configuring multi-channel audio systems is described. Compensating for non-ideal speaker positioning is a problem seen in immersive audio-visual art installations, home theater surround sound setups, and live concerts. Manual configuration requires expertise and time, while automatic methods promise to reduce these costs, enabling quick and easy setup and operation. Ideally the system should outperform a human in aural sound source localization. A naïve method is proposed and paired software is evaluated aiming to cut down on setup time, use readily available hardware, and enable satisfactory multi-channel spatialization and sound-source localization