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At least 91 records · Page 5

Automatic Speech Acquisition and Recognition for Spacesuit Audio Systems

NASA has a widely recognized but unmet need for novel human-machine interface technologies that can facilitate communication during astronaut extravehicular activities (EVAs), when loud noises and strong reverberations inside spacesuits make communication challenging. WeVoice, Inc., has developed a multichannel signal-processing method for speech acquisition in noisy and reverberant environments that enables automatic speech recognition (ASR) technology inside spacesuits. The technology reduces noise by exploiting differences between the statistical nature of signals (i.e., speech) and noise that exists in the spatial and temporal domains. As a result, ASR accuracy can be improved to the level at which crewmembers will find the speech interface useful. System components and features include beam forming/multichannel noise reduction, single-channel noise reduction, speech feature extraction, feature transformation and normalization, feature compression, and ASR decoding. Arithmetic complexity models were developed and will help designers of real-time ASR systems select proper tasks when confronted with constraints in computational resources. In Phase I of the project, WeVoice validated the technology. The company further refined the technology in Phase II and developed a prototype for testing and use by suited astronauts.

Ye, Sherry↗

Ad Hoc Selection of Voice over Internet Streams

A method and apparatus for a communication system technique involving ad hoc selection of at least two audio streams is provided. Each of the at least two audio streams is a packetized version of an audio source. A data connection exists between a server and a client where a transport protocol actively propagates the at least two audio streams from the server to the client. Furthermore, software instructions executable on the client indicate a presence of the at least two audio streams, allow selection of at least one of the at least two audio streams, and direct the selected at least one of the at least two audio streams for audio playback.

Macha, Mitchell G.↗

Ad Hoc Selection of Voice over Internet Streams

A method and apparatus for a communication system technique involving ad hoc selection of at least two audio streams is provided. Each of the at least two audio streams is a packetized version of an audio source. A data connection exists between a server and a client where a transport protocol actively propagates the at least two audio streams from the server to the client. Furthermore, software instructions executable on the client indicate a presence of the at least two audio streams, allow selection of at least one of the at least two audio streams, and direct the selected at least one of the at least two audio streams for audio playback.

Macha, Mitchell G.↗

Entertainment and Pacification System For Car Seat

An entertainment and pacification system for use with a child car seat has speakers mounted in the child car seat with a plurality of audio sources and an anti-noise audio system coupled to the child car seat. A controllable switching system provides for, at any given time, the selective activation of i) one of the audio sources such that the audio signal generated thereby is coupled to one or more of the speakers, and ii) the anti-noise audio system such that an ambient-noise-canceling audio signal generated thereby is coupled to one or more of the speakers. The controllable switching system can receive commands generated at one of first controls located at the child car seat and second controls located remotely with respect to the child car seat with commands generated by the second controls overriding commands generated by the first controls.

Elrod, Susan Vinz↗

Acoustic Calibration of the Exterior Effects Room at the NASA Langley Research Center

The Exterior Effects Room (EER) at the NASA Langley Research Center is a 39-seat auditorium built for psychoacoustic studies of aircraft community noise. The original reproduction system employed monaural playback and hence lacked sound localization capability. In an effort to more closely recreate field test conditions, a significant upgrade was undertaken to allow simulation of a three-dimensional audio and visual environment. The 3D audio system consists of 27 mid and high frequency satellite speakers and 4 subwoofers, driven by a real-time audio server running an implementation of Vector Base Amplitude Panning. The audio server is part of a larger simulation system, which controls the audio and visual presentation of recorded and synthesized aircraft flyovers. The focus of this work is on the calibration of the 3D audio system, including gains used in the amplitude panning algorithm, speaker equalization, and absolute gain control. Because the speakers are installed in an irregularly shaped room, the speaker equalization includes time delay and gain compensation due to different mounting distances from the focal point, filtering for color compensation due to different installations (half space, corner, baffled/unbaffled), and cross-over filtering.

Faller, Kenneth J., II↗

Intern Abstract for Spring 2016

The Human Interface Branch - EV3 - is evaluating Organic lighting-emitting diodes (OLEDs) as an upgrade for current displays on future spacecraft. OLEDs have many advantages over current displays. Conventional displays require constant backlighting which draws a lot of power, but with OLEDs they generate light themselves. OLEDs are lighter, and weight is always a concern with space launches. OLEDs also grant greater viewing angles. OLEDs have been in the commercial market for almost ten years now. What is not known is how they will perform in a space-like environment; specifically deep space far away from the Earth's magnetosphere. In this environment, the OLEDs can be expected to experience vacuum and galactic radiation. The intern's responsibility has been to prepare the OLED for a battery of tests. Unfortunately, it will not be ready for testing at the end of the internship. That being said much progress has been made: a) Developed procedures to safely disassemble the tablet. b) Inventoried and identified critical electronic components. c) 3D printed a testing apparatus. d) Wrote software in Python that will test the OLED screen while being radiated. e) Built circuits to restart the tablet and the test pattern, and ensure it doesn't fall asleep during radiation testing. f) Built enclosure that will house all of the electronics Also, the intern has been working on a way to take messages from a simulated Caution and Warnings system, process said messages into packets, send audio packets to a multicast address that audio boxes are listening to, and output spoken audio. Currently, Cautions and Warnings use a tone to alert crew members of a situation, and then crew members have to read through their checklists to determine what the tone means. In urgent situations, EV3 wants to deliver concise and specific alerts to the crew to facilitate any mitigation efforts on their part. Significant progress was made on this project: a) Open channel with the simulated Caution and Warning system to acquire messages. b) Configure audio boxes. c) Grab pre-recorded audio files. d) Packetize the audio stream. A third project that was assigned to implement LED indicator modules for an Omnibus project. The Omnibus project is investigating better ways designing lighting for the interior of spacecraft-both spacecraft lighting and avionics box status lighting indication. The current scheme contains too much of the blue light spectrum that disrupts the sleep cycle. The LED indicator modules are to simulate the indicators running on a spacecraft. Lighting data will be gathered by human factors personal and use in a model underdevelopment to model spacecraft lighting. Significant progress was made on this project: Designed circuit layout a) Tested LEDs at LETF. b) Created GUI for the indicators. c) Created code for the Arduino to run that will illuminate the indicator modules.

Gibson, William↗

Transcribing Air Traffic Control System Command Center Planning Telecons Using Cloud-Based Automatic Speech Recognition

This paper addresses the challenge of using Automatic Speech Recognition (ASR) technology to transcribe regular teleconferences that happen between FAA Air Traffic Control System Command Center (ATCSCC) planners, stakeholders and air users. These planning teleconferences (aka telecons or planning webinars) are an integral part of managing air traffic in the U.S. National Airspace System (NAS). In particular, the meetings facilitate the creation and modification of various traffic management initiatives (TMIs), that are used to regulate the flow of air traffic. This is typically a human intensive process, requiring specialists to listen to the entire meeting audio (10-20 minutes duration) and inferring the state of the NAS (e.g., weather phenomenon) that was discussed. It would be advantageous to have digital transcripts of the audio and have useful information (e.g., related to TMIs) automatically extracted from the transcripts. In this regard, we are exploring the adoption of state-of-the-art speech to text and Natural Language Processing (NLP) tools that will achieve our objective of digitizing the webinar audio. Unfortunately, the highly technical phraseology present in the audio and limited data availability for model building make ASR difficult. To overcome this challenge, we have taken the critical first step in creating a human transcription dataset from ~20 hours of speech in the ATCSCC audio with the help of subject matter experts. A novelty of our work is the creation of a ground truth transcription dataset for ATCSCC teleconference webinars, which is particularly important for Aviation domain-specific NLP tasks. Using Microsoft Speech Studio, a cloud-based ASR platform, we have fine-tuned the English pre-trained ASR models (available in speech studio) and achieved an average word error rate (WER) of 6.81%. The baseline ASR also provides a digital version of each planning webinar, making it accessible and text-searchable for future references. Additionally, the transcriptions can serve as a bridge between raw audio data and a range of text-based NLP tasks, such as named entity recognition (NER) and intent classification, potentially enhancing the digital footprint of the webinars and other connected data sources. Our work has several potential applications. Firstly, the transcriptions can be analyzed to understand the complex decision process of creating, implementing and modifying TMIs and may also contribute to TMI prediction services. Secondly, our dataset and model can be used to develop more accurate ASR systems for aviation-specific language, which can bring about digital communication in the aviation industry (and aid current “voice only” communications, which are inherently error-prone). Lastly, the transcriptions themselves can be used as a valuable resource for training other NLP models.

Stephen S. B. Clarke↗

The NT digital micro tape recorder

The description of an audio recorder may at first glance seem out of place in a conference which has been dedicated to the discussion of the technology and requirements of mass data storage. However, there are several advanced features of the NT system which will be of interest to the mass storage technologist. Moreover, there are a sufficient number of data storage formats in current use which have evolved from their audio counterparts to recommend a close attention to major innovative introductions of audio storage formats. While the existing analog micro-cassette recorder has been (and will continue to be) adequate for various uses, there are significant benefits to be gained through the application of digital technology. The elimination of background tape hiss and the availability of two relatively wide band channels (for stereo recording), for example, would greatly enhance listenability and speech intelligibility. And with the use of advanced high-density recording and LSI circuit technologies, a digital micro recorder can realize unprecedented compactness with excellent energy efficiency. This is what was accomplished with the NT-1 Digital Micro Recorder. Its remarkably compact size contributes to its portability. The high-density NT format enables up to two hours of low-noise digital stereo recording on a cassette the size of a postage stamp. Its highly energy-efficient mechanical and electrical design results in low power consumption; the unit can be operated up to 7 hours (for continuous recording) on a single AA alkaline battery. Advanced user conveniences include a multifunction LCD readout. The unit's compactness and energy-efficiency, in particular, are attributes that cannot be matched by existing analog and digital audio formats. The size, performance, and features of the NT format are of benefit primarily to those who desire improved portability and audio quality in a personal memo product. The NT Recorder is the result of over ten years of intensive, multi-disciplinary research and development. What follows is a discussion of the technologies that have made the NT possible: (1) NT format mechanics, (2) NT media, (3) NT circuitry and board.

Sasaki, Toshikazu↗

The VTLA System of Course Delivery and Faculty Development in Materials Education

There is a national need for high-quality, upper division courses that address critical topics in materials synthesis, particularly those beyond the present expertise of the typical university department's faculty. A new project has been started to test a novel distance education and faculty development system, called Video Tape Live Audio (VTLA). This, if successful, would at once enlarge the national Materials Science and Engineering (MSE) student cohort studying material synthesis and develop faculty expertise at the receiving sites. The mechanics for the VTLA scheme are as follows: A course is designed in the field selected for emphasis and for which there is likely to be considerable demand, in this example 'Ceramic Materials Synthesis: Theory and Case Studies'. One of the very best researcher/teachers records lectures of TV studio quality with appropriate visuals. Universities and colleges which wish to offer the course agree to offer it at the same hour at least once a week. The videotaped lectures and accompanying text, readings and visuals are shipped to the professor in charge, who has an appropriate background. The professor arranges the classroom TV presentation equipment and supervises the course. Video lectures are played during regular course hours twice a week with time for discussion by the supervising professor. Typically the third weekly classroom period is scheduled by all sites at a common designated hour, during which the course author/presenter answers questions, provides greater depth, etc. on a live audio link to all course sites. Questions are submitted by fax and e-mail prior to the audio tutorial. coordinating professors at various sites have separate audio teleconferences at the beginning and end of the course, dealing with the philosophical and pedagogical approach to the course, content and mechanics. Following service once or twice as an 'apprentice' to the course, the coordinating professors may then offer it without the necessity of the live audio tutorial.

Berrettini, Robert↗

Apparatus for providing sensory substitution of force feedback

A feedback apparatus for an operator to control an effector that is remote from the operator to interact with a remote environment has a local input device to be manipulated by the operator. Sensors in the effector's environment are capable of sensing the amplitude of forces arising between the effector and its environment, the direction of application of such forces, or both amplitude and direction. A feedback signal corresponding to such a component of the force, is generated and transmitted to the environment of the operator. The signal is transduced into an auditory sensory substitution signal to which the operator is sensitive. Sound production apparatus present the auditory signal to the operator. The full range of the force amplitude may be represented by a single, audio speaker. Auditory display elements may be stereo headphones or free standing audio speakers, numbering from one to many more than two. The location of the application of the force may also be specified by the location of audio speakers that generate signals corresponding to specific forces. Alternatively, the location may be specified by the frequency of an audio signal, or by the apparent location of an audio signal, as simulated by a combination of signals originating at different locations.

Massimino, Michael J.↗

(abstract) Synthesis of Speaker Facial Movements to Match Selected Speech Sequences

We are developing a system for synthesizing image sequences the simulate the facial motion of a speaker. To perform this synthesis, we are pursuing two major areas of effort. We are developing the necessary computer graphics technology to synthesize a realistic image sequence of a person speaking selected speech sequences. Next, we are developing a model that expresses the relation between spoken phonemes and face/mouth shape. A subject is video taped speaking an arbitrary text that contains expression of the full list of desired database phonemes. The subject is video taped from the front speaking normally, recording both audio and video detail simultaneously. Using the audio track, we identify the specific video frames on the tape relating to each spoken phoneme. From this range we digitize the video frame which represents the extreme of mouth motion/shape. Thus, we construct a database of images of face/mouth shape related to spoken phonemes. A selected audio speech sequence is recorded which is the basis for synthesizing a matching video sequence; the speaker need not be the same as used for constructing the database. The audio sequence is analyzed to determine the spoken phoneme sequence and the relative timing of the enunciation of those phonemes. Synthesizing an image sequence corresponding to the spoken phoneme sequence is accomplished using a graphics technique known as morphing. Image sequence keyframes necessary for this processing are based on the spoken phoneme sequence and timing. We have been successful in synthesizing the facial motion of a native English speaker for a small set of arbitrary speech segments. Our future work will focus on advancement of the face shape/phoneme model and independent control of facial features.

graphics audio digitizing morphing enunciation pho↗

A digital audio/video interleaving system

A method of interleaving an audio signal with its associated video signal for simultaneous transmission or recording, and the subsequent separation of the two signals, is described. Comparisons are made between the new audio signal interleaving system and the Skylab Pam audio/video interleaving system, pointing out improvements gained by using the digital audio/video interleaving system. It was found that the digital technique is the simplest, most effective and most reliable method for interleaving audio and/or other types of data into the video signal for the Shuttle Orbiter application. Details of the design of a multiplexer capable of accommodating two basic data channels, each consisting of a single 31.5-kb/s digital bit stream are given. An adaptive slope delta modulation system is introduced to digitize audio signals, producing a high immunity of work intelligibility to channel errors, primarily due to the robust nature of the delta-modulation algorithm.

Richards, R. W.↗

Method and apparatus for operating on companded PCM voice data

The method and apparatus constructed in accordance with this invention permits a plurality of parties to speak to each other on a conference line with a minimum of interference. The apparatus digitizes audio signals. Each of the parties has an audio transmitter and receiver provided for transmitting and receiving audio signals. The audio signals are converted to a PCM companded eight-bit parallel signal followed by a conversion to a serial signal for transmitting to a remote location and then reconverting each of the companded signals to a first-eight-bit parallel signal. The eight-bit parallel signal is fed to one input of a pre-programmed ROM. This eight-bit signal provides one-half of a sixteen-bit address of a lookup ROM. The other half of the sixteen-bit ROM address is supplied by another suscriber over an identical circuit.

Byrne, F.↗

Effect of vergence on the gain of the linear vestibulo-ocular reflex

We measured the linear vestibulo-ocular reflex (LVOR) and vergence, using binocular search coils, in 3 humans. The subjects were accelerated sinusoidally at 0.5 Hz and 0.2 g peak acceleration, in complete darkness, while performing three different tasks: i) mental arithmetic; ii) tracking a remembered target at either 0.34 m or 0.14 m distance; and iii) maintaining vergence at either of these distances by means of audio biofeedback based on vergence. Subjects could control vergence using the audio feedback; there was greater convergence with the near audio target. However, there was no significant difference in vergence between the near and far remembered target conditions. With audio feedback, the amplitude of smooth tracking was not consistently different for the near and the far conditions. However, the amplitude of tracking (saccades and smooth component) in the remembered target conditions was greater for near than for far targets. These results suggest that linear VOR amplitude is not determined by vergence alone.

NASA Program Space Physiology and Countermeasures↗

Flying by Ear: Blind Flight with a Music-Based Artificial Horizon

Two experiments were conducted in actual flight operations to evaluate an audio artificial horizon display that imposed aircraft attitude information on pilot-selected music. The first experiment examined a pilot's ability to identify, with vision obscured, a change in aircraft roll or pitch, with and without the audio artificial horizon display. The results suggest that the audio horizon display improves the accuracy of attitude identification overall, but differentially affects response time across conditions. In the second experiment, subject pilots performed recoveries from displaced aircraft attitudes using either standard visual instruments, or, with vision obscured, the audio artificial horizon display. The results suggest that subjects were able to maneuver the aircraft to within its safety envelope. Overall, pilots were able to benefit from the display, suggesting that such a display could help to improve overall safety in general aviation.

Simpson, Brian D.↗

Method and apparatus for obtaining complete speech signals for speech recognition applications

The present invention relates to a method and apparatus for obtaining complete speech signals for speech recognition applications. In one embodiment, the method continuously records an audio stream comprising a sequence of frames to a circular buffer. When a user command to commence or terminate speech recognition is received, the method obtains a number of frames of the audio stream occurring before or after the user command in order to identify an augmented audio signal for speech recognition processing. In further embodiments, the method analyzes the augmented audio signal in order to locate starting and ending speech endpoints that bound at least a portion of speech to be processed for recognition. At least one of the speech endpoints is located using a Hidden Markov Model.

Abrash, Victor↗

Using Simulation to Analyze Acoustic Environments

One of the main projects that was worked on this semester was creating an acoustic model for the Advanced Space Suit in Comsol Multiphysics. The geometry tools built into the software were used to create an accurate model of the helmet and upper torso of the suit. After running the simulation, plots of the sound pressure level within the suit were produced, as seen below in Figure 1. These plots show significant nulls which should be avoided when placing microphones inside the suit. In the future, this model can be easily adapted to changes in the suit design to determine optimal microphone placements and other acoustic properties. Another major project was creating an acoustic diverter that will potentially be used to route audio into the Space Station's Node 1. The concept of the project was to create geometry to divert sound from a neighboring module, the US Lab, into Node 1. By doing this, no new audio equipment would need to be installed in Node 1. After creating an initial design for the diverter, analysis was performed in Comsol in order to determine how changes in geometry would affect acoustic performance, as shown in Figure 2. These results were used to produce a physical prototype diverter on a 3D printer. With the physical prototype, testing was conducted in an anechoic chamber to determine the true effectiveness of the design, as seen in Figure 3. The results from this testing have been compared to the Comsol simulation results to analyze how closely the Comsol results are to real-world performance. While the Comsol results do not seem to closely resemble the real world performance, this testing has provided valuable insight into how much trust can be placed in the results of Comsol simulations. A final project that was worked on during this tour was the Audio Interface Unit (AIU) design for the Orion program. The AIU is a small device that will be used for as an audio communication device both during launch and on-orbit. The unit will have functions including push-to-talk buttons and volume control. With this project, an existing design was modified based on prior feedback that had been received. With the modified design, I created a 3D printed prototype, shown in Figure 4, which was then used in suited evaluations performed by crew members. The feedback received from those evaluations will be utilized to help create the best possible Orion AIU. As a whole, a number of different interesting engineering projects were worked on over the course of this semester. For many of these projects, acoustic simulations provided valuable insight into how different environments would respond to sound. While work is still underway to verify the results of these simulations, the results are fascinating because of the interesting ways that sound waves interact with the environment. Going forward, it will be interesting to see how closely these results can be matched by real-world test data.

Wood, Eric J.↗