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At least 55 records · Page 3

Acoustical Testing Laboratory Developed to Support the Low-Noise Design of Microgravity Space Flight Hardware

The NASA John H. Glenn Research Center at Lewis Field has designed and constructed an Acoustical Testing Laboratory to support the low-noise design of microgravity space flight hardware. This new laboratory will provide acoustic emissions testing and noise control services for a variety of customers, particularly for microgravity space flight hardware that must meet International Space Station limits on noise emissions. These limits have been imposed by the space station to support hearing conservation, speech communication, and safety goals as well as to prevent noise-induced vibrations that could impact microgravity research data. The Acoustical Testing Laboratory consists of a 23 by 27 by 20 ft (height) convertible hemi/anechoic chamber and separate sound-attenuating test support enclosure. Absorptive 34-in. fiberglass wedges in the test chamber provide an anechoic environment down to 100 Hz. A spring-isolated floor system affords vibration isolation above 3 Hz. These criteria, along with very low design background levels, will enable the acquisition of accurate and repeatable acoustical measurements on test articles, up to a full space station rack in size, that produce very little noise. Removable floor wedges will allow the test chamber to operate in either a hemi/anechoic or anechoic configuration, depending on the size of the test article and the specific test being conducted. The test support enclosure functions as a control room during normal operations but, alternatively, may be used as a noise-control enclosure for test articles that require the operation of noise-generating test support equipment.

Cooper, Beth A.↗

Fluids and Combustion Facility Acoustic Emissions Controlled by Aggressive Low-Noise Design Process

The Fluids and Combustion Facility (FCF) is a dual-rack microgravity research facility that is being developed by Northrop Grumman Information Technology (NGIT) for the International Space Station (ISS) at the NASA Glenn Research Center. As an on-orbit test bed, FCF will host a succession of experiments in fluid and combustion physics. The Fluids Integrated Rack (FIR) and the Combustion Integrated Rack (CIR) must meet ISS acoustic emission requirements (ref. 1), which support speech communication and hearing-loss-prevention goals for ISS crew. To meet these requirements, the NGIT acoustics team implemented an aggressive low-noise design effort that incorporated frequent acoustic emission testing for all internal noise sources, larger-scale systems, and fully integrated racks (ref. 2). Glenn's Acoustical Testing Laboratory (ref. 3) provided acoustical testing services (see the following photograph) as well as specialized acoustical engineering support as part of the low-noise design process (ref. 4).

Cooper, Beth A.↗

Effect of Whole-Body Vibration on Speech: Effect on Intelligibility - Part 2

The effect on speech intelligibility was measured for speech where talkers reading Diagnostic Rhyme Test material were exposed to 0.7 g whole body vibration to simulate space vehicle launch. Across all talkers, the effect of vibration was to degrade the percentage of correctly transcribed words from 83% to 74%. The magnitude of the effect of vibration on speech communication varies between individuals, for both talkers and listeners. A worst case scenario for intelligibility would be the most sensitive listener hearing the most sensitive talker; one participant s intelligibility was reduced by 26% (97% to 71%) for one of the talkers.

Begault, Durand R.↗

A low-delay 8 Kb/s backward-adaptive CELP coder

Code excited linear prediction coding is an efficient technique for compressing speech sequences. Communications quality of speech can be obtained at bit rates below 8 Kb/s. However, relatively large coding delays are necessary to buffer the input speech in order to perform the LPC analysis. A low delay 8 Kb/s CELP coder is introduced in which the short term predictor is based on past synthesized speech. A new distortion measure that improves the tracking of the formant filter is discussed. Formal listening tests showed that the performance of the backward adaptive coder is almost as good as the conventional CELP coder.

Neumeyer, L. G.↗

Applications for Subvocal Speech

A research and development effort now underway is directed toward the use of subvocal speech for communication in settings in which (1) acoustic noise could interfere excessively with ordinary vocal communication and/or (2) acoustic silence or secrecy of communication is required. By "subvocal speech" is meant sub-audible electromyographic (EMG) signals, associated with speech, that are acquired from the surface of the larynx and lingual areas of the throat. Topics addressed in this effort include recognition of the sub-vocal EMG signals that represent specific original words or phrases; transformation (including encoding and/or enciphering) of the signals into forms that are less vulnerable to distortion, degradation, and/or interception; and reconstruction of the original words or phrases at the receiving end of a communication link. Potential applications include ordinary verbal communications among hazardous- material-cleanup workers in protective suits, workers in noisy environments, divers, and firefighters, and secret communications among law-enforcement officers and military personnel in combat and other confrontational situations.

Jorgensen, Charles↗

Automatic speech recognition technology development at ITT Defense Communications Division

An assessment of the applications of automatic speech recognition to defense communication systems is presented. Future research efforts include investigations into the following areas: (1) dynamic programming; (2) recognition of speech degraded by noise; (3) speaker independent recognition; (4) large vocabulary recognition; (5) word spotting and continuous speech recognition; and (6) isolated word recognition.

White, George M.↗

Speech Intelligibility Advantages using an Acoustic Beamformer Display

A speech intelligibility test conforming to the Modified Rhyme Test of ANSI S3.2 "Method for Measuring the Intelligibility of Speech Over Communication Systems" was conducted using a prototype 12-channel acoustic beamformer system. The target speech material (signal) was identified against speech babble (noise), with calculated signal-noise ratios of 0, 5 and 10 dB. The signal was delivered at a fixed beam orientation of 135 deg (re 90 deg as the frontal direction of the array) and the noise at 135 deg (co-located) and 0 deg (separated). A significant improvement in intelligibility from 57% to 73% was found for spatial separation for the same signal-noise ratio (0 dB). Significant effects for improved intelligibility due to spatial separation were also found for higher signal-noise ratios (5 and 10 dB).

acoustic beamforming↗

Automatic Speech Recognition for Launch Control Center Communication Using Recurrent Neural Networks with Data Augmentation and Custom Language Model

Transcribing voice communications in NASA’s launch control center is important for information utilization. However, automatic speech recognition in this environment is particularly challenging due to the lack of training data, unfamiliar words in acronyms, multiple different speakers and accents, and conversational characteristics of speaking. We used bidirectional deep recurrent neural networks to train and test speech recognition performance. We showed that data augmentation and custom language models can improve speech recognition accuracy. Transcribing communications from the launch control center will help the machine analyze information and accelerate knowledge generation.

Chow, Edward↗

Space-Station-Interior Noise-Analysis Program

Intelligibility of speech evaluated for specified acoustical environments. Program makes systematic prediction of noise and vibration environment of craft defined by user and evaluates relative acceptability of predicted environment for effective communication by speech. Written in MicroSoft FORTRAN Version 3.3.

Stusnick, Eric↗

A user's guide for the signal processing software for image and speech compression developed in the Communications and Signal Processing Laboratory (CSPL), version 1

A complete documentation of the software developed in the Communication and Signal Processing Laboratory (CSPL) during the period of July 1985 to March 1986 is provided. Utility programs and subroutines that were developed for a user-friendly image and speech processing environment are described. Additional programs for data compression of image and speech type signals are included. Also, programs for the zero-memory and block transform quantization in the presence of channel noise are described. Finally, several routines for simulating the perfromance of image compression algorithms are included.

Kumar, P.↗

Research and development of a versatile portable speech prosthesis

The Versatile Portable Speech Prosthesis (VPSP), a synthetic speech output communication aid for non-speaking people is described. It was intended initially for severely physically limited people with cerebral palsy who are in electric wheelchairs. Hence, it was designed to be placed on a wheelchair and powered from a wheelchair battery. It can easily be separated from the wheelchair. The VPSP is versatile because it is designed to accept any means of single switch, multiple switch, or keyboard control which physically limited people have the ability to use. It is portable because it is mounted on and can go with the electric wheelchair. It is a speech prosthesis, obviously, because it speaks with a synthetic voice for people unable to speak with their own voices. Both hardware and software are described.

Source record↗

xEMU Suit Integrated Audio Communications System: Ambient and EVA Pressure Testing System Performance

Testing across several airlock and EVA thermal and pressure scenarios has demonstrated that the Integrated Audio System of NASA’s Exploration Extravehicular Mobility Unit (xEMU) spacesuits transmits and receives intelligible audio communications without the use of a commcap or similar worn device. The xEMU audio system consists of internal loudspeakers and digital microphones (Integrated Communications System –ICS) combined with an adaptive Acoustic Echo Canceller (AEC), outbound voice operated transmission (VOX), and automatic gain control (AGC). Transducers are mounted in an “exploded commcap” configuration with helmet-attached speakers near the ears and three microphones positioned at the collar. The AGC removes inbound audio signals (e.g.,suit, Mission Control, Lander, C&W tones) from the outbound comms stream, reducing echo and feedback (squeal) in low-noise suit environments. The reduction of worn communication equipment increases crewmember comfort, range of movement, and situational awareness. However, test results also highlight the need for proper fan, duct, pump, and gas flow integration with suit acoustics and audio. Ductwork may serve as waveguides for various component and structure-borne noise. Sharply angled ducts can generate turbulent-flow noise. Gas flow from inlets above the crewmember’s head can generate noise when cascading over the faceplate and collar (or commcap) microphones. Sufficient acoustic noise levels (1) require increased gain to boost inbound audio, and (2) may distort signals resulting in AEC disruption or artifacts. Suit-noise levels decline with reduced pressure (density), but then elevated speech and audio effort/power become necessary. Whether the Integrated Audio System, commcap, or other device is used, suit acoustic noise can mask speech in outbound comms. This reduces intelligibility and requires other AECs/devices to suppress comms noise. Yet, adjusting a few components may yield significant improvement. We discuss xEMU audio functionality, demonstrate how acoustical treatment combined with inbound signal conditioning improved clarity during tests, and discuss future modifications.

xEMU↗

xEMU Suit Integrated Audio Communications System: Ambient and EVA Pressure Testing System Performance

Testing across several airlock and EVA thermal and pressure scenarios has demonstrated that the Integrated Audio System of NASA’s Exploration Extravehicular Mobility Unit (xEMU) spacesuits transmits and receives intelligible audio communications without the use of a commcap or similar worn device. The xEMU audio system consists of internal loudspeakers and digital microphones (Integrated Communications System –ICS) combined with an adaptive Acoustic Echo Canceller (AEC), outbound voice operated transmission (VOX), and automatic gain control (AGC). Transducers are mounted in an “exploded commcap” configuration with helmet-attached speakers near the ears and three microphones positioned at the collar. The AGC removes inbound audio signals (e.g.,suit, Mission Control, Lander, C&W tones) from the outbound comms stream, reducing echo and feedback (squeal) in low-noise suit environments. The reduction of worn communication equipment increases crewmember comfort, range of movement, and situational awareness. However, test results also highlight the need for proper fan, duct, pump, and gas flow integration with suit acoustics and audio. Ductwork may serve as waveguides for various component and structure-borne noise. Sharply angled ducts can generate turbulent-flow noise. Gas flow from inlets above the crewmember’s head can generate noise when cascading over the faceplate and collar (or commcap) microphones. Sufficient acoustic noise levels (1) require increased gain to boost inbound audio, and (2) may distort signals resulting in AEC disruption or artifacts. Suit-noise levels decline with reduced pressure (density), but then elevated speech and audio effort/power become necessary. Whether the Integrated Audio System, commcap, or other device is used, suit acoustic noise can mask speech in outbound comms. This reduces intelligibility and requires other AECs/devices to suppress comms noise. Yet, adjusting a few components may yield significant improvement. We discuss xEMU audio functionality, demonstrate how acoustical treatment combined with inbound signal conditioning improved clarity during tests, and discuss future modifications.

xEMU↗

Linguistics

Linguistic research on transformational grammar, Lithuanian morphophonemics, English directionals, and concept of performance

SPEECH↗

Vector excitation coding with dynamic bit allocation

Vector excitation coding (VXC) has shown promise for digital transmission of fairly high communications quality speech at low bit rates, but current version of the algorithm still suffer from audible degradations, particularly at 4.8 kb/s. The authors examine the technique of dynamic bit allocation (DBA) to improve the performance of VXC for a given bit rate. The approach is based on the observation that the minimum bit rate needed to code adequately both the long-term and short-term predictors in VXC varies dynamically with time. Therefore, in frames where fewer bits suffice for the predictors, the unneeded bits can be reallocated to other coder parameter sets such as excitation vectors. By dynamically distributing available bits among the different coder parameter sets while keeping the total number of bits in each frame fixed, it is possible to improve overall coder performance without increasing bit rate.

Yong, Mei↗

Agent oriented programming

The goal of our research is a methodology for creating robust software in distributed and dynamic environments. The approach taken is to endow software objects with explicit information about one another, to have them interact through a commitment mechanism, and to equip them with a speech-acty communication language. System-level applications include software interoperation and compositionality. A government application of specific interest is an infrastructure for coordination among multiple planners. Daily activity applications include personal software assistants, such as programmable email, scheduling, and new group agents. Research topics include definition of mental state of agents, design of agent languages as well as interpreters for those languages, and mechanisms for coordination within agent societies such as artificial social laws and conventions.

Shoham, Yoav↗

Kennedy Space Center ITC-1 Internship Overview

As an intern for Priscilla Elfrey in the ITC-1 department, I was involved in many activities that have helped me to develop many new skills. I supported four different projects during my internship, which included the Center for Life Cycle Design (CfLCD), SISO Space Interoperability Smackdown, RTI Teacher Mentor Program, and the Discrete Event Simulation Integrated Visualization Environment Team (DIVE). I provided the CfLCD with web based research on cyber security initiatives involving simulation, education for young children, cloud computing, Otronicon, and Science, Technology, Engineering, and Mathematics (STEM) education initiatives. I also attended STEM meetings regarding simulation courses, and educational course enhancements. To further improve the SISO Simulation event, I provided observation feedback to the technical advisory board. I also helped to set up a chat federation for HLA. The third project involved the RTI Teacher Mentor program, which I helped to organize. Last, but not least, I worked with the DIVE team to develop new software to help visualize discrete event simulations. All of these projects have provided experience on an interdisciplinary level ranging from speech and communication to solving complex problems using math and science.

Ni, Marcus↗

Performance of a low data rate speech codec for land-mobile satellite communications

In an effort to foster the development of new technologies for the emerging land mobile satellite communications services, JPL funded two development contracts in 1984: one to the Univ. of Calif., Santa Barbara and the other to the Georgia Inst. of Technology, to develop algorithms and real time hardware for near toll quality speech compression at 4800 bits per second. Both universities have developed and delivered speech codecs to JPL, and the UCSB codec was extensively tested by JPL in a variety of experimental setups. The basic UCSB speech codec algorithms and the test results of the various experiments performed with this codec are presented.

Gersho, Allen↗