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SFO marine stratus forecast system documentation

Summary

San Francisco International Airport (SFO) experiences frequent low ceiling conditions during the summer season due to marine stratus clouds. Stratus in the approach zone prevents dual approaches to the airport??s closely spaced parallel runways, effectively reducing arrival capacity by half. The stratus typically behaves on a daily cycle, with dissipation occurring during the hours following sunrise. Often the low ceiling conditions persist throughout the morning hours and interfere with the high rate of air traffic scheduled into SFO from mid-morning to early afternoon. Air traffic managers require accurate forecasts of clearing time to efficiently administer Ground Delay Programs (GDPs) to match the rate of arriving aircraft with expected capacity. The San Francisco Marine Stratus Forecast System was developed as a tool for anticipating the time of stratus clearing. The system relies on field-deployed sensors as well as routinely available regional surface observations and satellite data from the Geostationary Operational Environmental Satellite (GOES-West). Data are collected, processed, and input to a suite of forecast models to predict the time that the approach zone will be sufficiently clear to perform dual approaches. Data observations and model forecasts are delivered to users on an interactive display accessible via the Internet. The system prototype was developed under the sponsorship of the FAA Aviation Weather Research Program (AWRP). MIT Lincoln Laboratory served as technical lead for the project, in collaboration with San Jose State University, the University of Quebec at Montreal, and the Center Weather Service Unit (CWSU) at the Oakland Air Route Traffic Control Center (ARTCC). The National Weather Service (NWS), under the direction of the NWS Forecast Office in Monterey, assumed responsibility for operation and maintenance of the system following technical transfer in 2004. This document was compiled as a resource to support continuing system operation and maintenance.
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Summary

San Francisco International Airport (SFO) experiences frequent low ceiling conditions during the summer season due to marine stratus clouds. Stratus in the approach zone prevents dual approaches to the airport??s closely spaced parallel runways, effectively reducing arrival capacity by half. The stratus typically behaves on a daily cycle, with dissipation...

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Improving the resolution advisory reversal logic of the traffic alert and collision avoidance system

Published in:
25th IEEE/AIAA Digital Avionics Systems Conf., 15-18 October 2006, pp. 561-570.

Summary

The Traffic Alert and Collision Avoidance System (TCAS II) is the worldwide standard system for manned aircraft to avoid collisions with airborne transponder-equipped traffic. A safety vulnerability of the collision avoidance logic was reported by European analysts, who also proposed a change to correct it. The safety issue concerns limitations in the ability of TCAS to reverse the sense of a Resolution Advisory (RA) during an encounter. The issue was addressed by a team of experts1 in the Requirements Working Group (RWG) of RTCA Special Committee 147 [1]. This paper discusses the problem, the metrics and methods used in the analysis, and presents results that quantify the effectiveness of the proposed solution. Finally, recommendations are presented for implementing the change.
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Summary

The Traffic Alert and Collision Avoidance System (TCAS II) is the worldwide standard system for manned aircraft to avoid collisions with airborne transponder-equipped traffic. A safety vulnerability of the collision avoidance logic was reported by European analysts, who also proposed a change to correct it. The safety issue concerns limitations...

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A wafer-scale 3-D circuit integration technology

Published in:
IEEE Trans. Electron Devices, Vol. 53, No. 10, October 2006, pp. 2507-2516.

Summary

The rationale and development of a wafer-scale three-dimensional (3-D) integrated circuit technology are described. The essential elements of the 3-D technology are integrated circuit fabrication on silicon-on-insulator wafers, precision wafer-wafer alignment using an in-house-developed alignment system, low-temperature wafer-wafer bonding to transfer and stack active circuit layers, and interconnection of the circuit layers with dense-vertical connections with sub-[Omega] 3-D via resistances. The 3-D integration process is described as well as the properties of the four enabling technologies. The wafer-scale 3-D technology imposes constraints on the placement of the first lithographic level in a wafer-stepper process. Control of wafer distortion and wafer bow is required to achieve submicrometer vertical vias. Three-tier digital and analog 3-D circuits were designed and fabricated. The performance characteristics of a 3-D ring oscillator, a 1024 x 1024 visible imager with an 8-um pixel pitch, and a 64 x 64 Geiger-mode laser radar chip are described.
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Summary

The rationale and development of a wafer-scale three-dimensional (3-D) integrated circuit technology are described. The essential elements of the 3-D technology are integrated circuit fabrication on silicon-on-insulator wafers, precision wafer-wafer alignment using an in-house-developed alignment system, low-temperature wafer-wafer bonding to transfer and stack active circuit layers, and interconnection of the...

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Analysis of nonmodal phonation using minimum entropy deconvolution

Published in:
Proc. Int. Conf. on Spoken Language Processing, ICSLP INTERSPEECH, 17-21 September 2006, pp. 1702-1705.

Summary

Nonmodal phonation occurs when glottal pulses exhibit nonuniform pulse-to-pulse characteristics such as irregular spacings, amplitudes, and/or shapes. The analysis of regions of such nonmodality has application to automatic speech, speaker, language, and dialect recognition. In this paper, we examine the usefulness of a technique called minimum-entropy deconvolution, or MED, for the analysis of pulse events in nonmodal speech. Our study presents evidence for both natural and synthetic speech that MED decomposes nonmodal phonation into a series of sharp pulses and a set of mixedphase impulse responses. We show that the estimated impulse responses are quantitatively similar to those in our synthesis model. A hybrid method incorporating aspects of both MED and linear prediction is also introduced. We show preliminary evidence that the hybrid method has benefit over MED alone for composite impulse-response estimation by being more robust to short-time windowing effects as well as a speech aspiration noise component.
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Summary

Nonmodal phonation occurs when glottal pulses exhibit nonuniform pulse-to-pulse characteristics such as irregular spacings, amplitudes, and/or shapes. The analysis of regions of such nonmodality has application to automatic speech, speaker, language, and dialect recognition. In this paper, we examine the usefulness of a technique called minimum-entropy deconvolution, or MED, for...

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Lincoln Laboratory high-speed solid-state imager technology

Published in:
SPIE Vol. 6279, 27th Int. Congress on High-Speed Photography and Photonics, 17-22 September 2006, 62791K.

Summary

Massachusetts Institute of Technology, Lincoln Laboratory (MIT LL) has been developing both continuous and burst solid-state focal-plane-array technology for a variety of high-speed imaging applications. For continuous imaging, a 128 ¿ 128-pixel charge coupled device (CCD) has been fabricated with multiple output ports for operating rates greater than 10,000 frames per second with readout noise of less than 10 e- rms. An electronic shutter has been integrated into the pixels of the back-illuminated (BI) CCD imagers that give snapshot exposure times of less than 10 ns. For burst imaging, a 5 cm x 5 cm, 512 x 512-element, multi-frame CCD imager that collects four sequential image frames at megahertz rates has been developed for the Los Alamos National Laboratory Dual Axis Radiographic Hydrodynamic Test (DARHT) facility. To operate at fast frame rates with high sensitivity, the imager uses the same electronic shutter technology as the continuously framing 128 x 128 CCD imager. The design concept and test results are described for the burst-frame-rate imager. Also discussed is an evolving solid-state imager technology that has interesting characteristics for creating large-format x-ray detectors with ultra-short exposure times (100 to 300 ps). The detector will consist of CMOS readouts for high speed sampling (tens of picoseconds transistor switching times) that are bump bonded to deep-depletion silicon photodiodes. A 64 x 64-pixel CMOS test chip has been designed, fabricated and characterized to investigate the feasibility of making large-format detectors with short, simultaneous exposure times.
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Summary

Massachusetts Institute of Technology, Lincoln Laboratory (MIT LL) has been developing both continuous and burst solid-state focal-plane-array technology for a variety of high-speed imaging applications. For continuous imaging, a 128 ¿ 128-pixel charge coupled device (CCD) has been fabricated with multiple output ports for operating rates greater than 10,000 frames...

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Reducing speech coding distortion for speaker identification

Author:
Published in:
Int. Conf. on Spoken Language Processing, ICSLP, 17-21 September 2006.

Summary

In this paper, we investigate the degradation of speaker identification performance due to speech coding algorithms used in digital telephone networks, cellular telephony, and voice over IP. By analyzing the difference between front-end feature vectors derived from coded and uncoded speech in terms of spectral distortion, we are able to quantify this coding degradation. This leads to two novel methods for distortion compensation: codebook and LPC compensation. Both are shown to significantly reduce front-end mismatch, with the second approach providing the most encouraging results. Full experiments using a GMM-UBM speaker ID system confirm the usefulness of both the front-end distortion analysis and the LPC compensation technique.
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Summary

In this paper, we investigate the degradation of speaker identification performance due to speech coding algorithms used in digital telephone networks, cellular telephony, and voice over IP. By analyzing the difference between front-end feature vectors derived from coded and uncoded speech in terms of spectral distortion, we are able to...

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Pitch-scale modification using the modulated aspiration noise source

Published in:
INTERSPEECH, 17-21 September 2006.

Summary

Spectral harmonic/noise component analysis of spoken vowels shows evidence of noise modulations with peaks in the estimated noise source component synchronous with both the open phase of the periodic source and with time instants of glottal closure. Inspired by this observation of natural modulations and of fullband energy in the aspiration noise source, we develop an alternate approach to high-quality pitch-scale modification of continuous speech. Our strategy takes a dual processing approach, in which the harmonic and noise components of the speech signal are separately analyzed, modified, and re-synthesized. The periodic component is modified using standard modification techniques, and the noise component is handled by modifying characteristics of its source waveform. Since we have modeled an inherent coupling between the periodic and aspiration noise sources, the modification algorithm is designed to preserve the synchrony between temporal modulations of the two sources. The reconstructed modified signal is perceived in informal listening to be natural-sounding and typically reduces artifacts that occur in standard modification techniques.
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Summary

Spectral harmonic/noise component analysis of spoken vowels shows evidence of noise modulations with peaks in the estimated noise source component synchronous with both the open phase of the periodic source and with time instants of glottal closure. Inspired by this observation of natural modulations and of fullband energy in the...

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Missing feature theory with soft spectral subtraction for speaker verification

Published in:
Interspeech 2006, ICSLP, 17-21 September 2006.

Summary

This paper considers the problem of training/testing mismatch in the context of speaker verification and, in particular, explores the application of missing feature theory in the case of additive white Gaussian noise corruption in testing. Missing feature theory allows for corrupted features to be removed from scoring, the initial step of which is the detection of these features. One method of detection, employing spectral subtraction, is studied in a controlled manner and it is shown that with missing feature compensation the resulting verification performance is improved as long as a minimum number of features remain. Finally, a blending of "soft" spectral subtraction for noise mitigation and missing feature compensation is presented. The resulting performance improves on the constituent techniques alone, reducing the equal error rate by about 15% over an SNR range of 5 - 25 dB.
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Summary

This paper considers the problem of training/testing mismatch in the context of speaker verification and, in particular, explores the application of missing feature theory in the case of additive white Gaussian noise corruption in testing. Missing feature theory allows for corrupted features to be removed from scoring, the initial step...

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An overview of automatic speaker diarization systems

Published in:
IEEE Trans. Audio, Speech, and Language Processing, Vol. 14, No. 5, September 2006, pp. 1557-1565.

Summary

Audio diarization is the process of annotating an input audio channel with information that attributes (possibly overlapping) temporal regions of signal energy to their specific sources. These sources can include particular speakers, music, background noise sources, and other signal source/channel characteristics. Diarization can be used for helping speech recognition, facilitating the searching and indexing of audio archives, and increasing the richness of automatic transcriptions, making them more readable. In this paper, we provide an overview of the approaches currently used in a key area of audio diarization, namely speaker diarization, and discuss their relative merits and limitations. Performances using the different techniques are compared within the framework of the speaker diarization task in the DARPA EARS Rich Transcription evaluations. We also look at how the techniques are being introduced into real broadcast news systems and their portability to other domains and tasks such as meetings and speaker verification.
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Summary

Audio diarization is the process of annotating an input audio channel with information that attributes (possibly overlapping) temporal regions of signal energy to their specific sources. These sources can include particular speakers, music, background noise sources, and other signal source/channel characteristics. Diarization can be used for helping speech recognition, facilitating...

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Coherent beam combining of large number of PM fibres in 2-D fibre array

Published in:
Electron. Lett., Vol. 42, No. 18, 31 August 2006, pp. 17-18.

Summary

Coherent combining of a record 48 PM fibres in a phased array configuration is reported. The resulting Strehl ratio degrades by
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Summary

Coherent combining of a record 48 PM fibres in a phased array configuration is reported. The resulting Strehl ratio degrades by

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