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At least 19 records

Digital Multicasting of Multiple Audio Streams

The Mission Control Center Voice Over Internet Protocol (MCC VOIP) system (see figure) comprises hardware and software that effect simultaneous, nearly real-time transmission of as many as 14 different audio streams to authorized listeners via the MCC intranet and/or the Internet. The original version of the MCC VOIP system was conceived to enable flight-support personnel located in offices outside a spacecraft mission control center to monitor audio loops within the mission control center. Different versions of the MCC VOIP system could be used for a variety of public and commercial purposes - for example, to enable members of the general public to monitor one or more NASA audio streams through their home computers, to enable air-traffic supervisors to monitor communication between airline pilots and air-traffic controllers in training, and to monitor conferences among brokers in a stock exchange. At the transmitting end, the audio-distribution process begins with feeding the audio signals to analog-to-digital converters. The resulting digital streams are sent through the MCC intranet, using a user datagram protocol (UDP), to a server that converts them to encrypted data packets. The encrypted data packets are then routed to the personal computers of authorized users by use of multicasting techniques. The total data-processing load on the portion of the system upstream of and including the encryption server is the total load imposed by all of the audio streams being encoded, regardless of the number of the listeners or the number of streams being monitored concurrently by the listeners. The personal computer of a user authorized to listen is equipped with special- purpose MCC audio-player software. When the user launches the program, the user is prompted to provide identification and a password. In one of two access- control provisions, the program is hard-coded to validate the user s identity and password against a list maintained on a domain-controller computer at the MCC. In the other access-control provision, the program verifies that the user is authorized to have access to the audio streams. Once both access-control checks are completed, the audio software presents a graphical display that includes audiostream-selection buttons and volume-control sliders. The user can select all or any subset of the available audio streams and can adjust the volume of each stream independently of that of the other streams. The audio-player program spawns a "read" process for the selected stream(s). The spawned process sends, to the router(s), a "multicast-join" request for the selected streams. The router(s) responds to the request by sending the encrypted multicast packets to the spawned process. The spawned process receives the encrypted multicast packets and sends a decryption packet to audio-driver software. As the volume or muting features are changed by the user, interrupts are sent to the spawned process to change the corresponding attributes sent to the audio-driver software. The total latency of this system - that is, the total time from the origination of the audio signals to generation of sound at a listener s computer - lies between four and six seconds.

Macha, Mitchell↗

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↗

A Virtual Audio Guidance and Alert System for Commercial Aircraft Operations

Our work in virtual reality systems at NASA Ames Research Center includes the area of aurally-guided visual search, using specially-designed audio cues and spatial audio processing (also known as virtual or "3-D audio") techniques (Begault, 1994). Previous studies at Ames had revealed that use of 3-D audio for Traffic Collision Avoidance System (TCAS) advisories significantly reduced head-down time, compared to a head-down map display (0.5 sec advantage) or no display at all (2.2 sec advantage) (Begault, 1993, 1995; Begault & Pittman, 1994; see Wenzel, 1994, for an audio demo). Since the crew must keep their head up and looking out the window as much as possible when taxiing under low-visibility conditions, and the potential for "blunder" is increased under such conditions, it was sensible to evaluate the audio spatial cueing for a prototype audio ground collision avoidance warning (GCAW) system, and a 3-D audio guidance system. Results were favorable for GCAW, but not for the audio guidance system.

Begault, Durand R.↗

Three-Dimensional Audio Client Library

The Three-Dimensional Audio Client Library (3DAudio library) is a group of software routines written to facilitate development of both stand-alone (audio only) and immersive virtual-reality application programs that utilize three-dimensional audio displays. The library is intended to enable the development of three-dimensional audio client application programs by use of a code base common to multiple audio server computers. The 3DAudio library calls vendor-specific audio client libraries and currently supports the AuSIM Gold-Server and Lake Huron audio servers. 3DAudio library routines contain common functions for (1) initiation and termination of a client/audio server session, (2) configuration-file input, (3) positioning functions, (4) coordinate transformations, (5) audio transport functions, (6) rendering functions, (7) debugging functions, and (8) event-list-sequencing functions. The 3DAudio software is written in the C++ programming language and currently operates under the Linux, IRIX, and Windows operating systems.

Rizzi, Stephen A.↗

Space Shuttle Orbiter audio subsystem

The selection of the audio multiplex control configuration for the Space Shuttle Orbiter audio subsystem is discussed and special attention is given to the evaluation criteria of cost, weight and complexity. The specifications and design of the subsystem are described and detail is given to configurations of the audio terminal and audio central control unit (ATU, ACCU). The audio input from the ACCU, at a signal level of -12.2 to 14.8 dBV, nominal range, at 1 kHz, was found to have balanced source impedance and a balanced local impedance of 6000 + or - 600 ohms at 1 kHz, dc isolated. The Lyndon B. Johnson Space Center (JSC) electroacoustic test laboratory, an audio engineering facility consisting of a collection of acoustic test chambers, analyzed problems of speaker and headset performance, multiplexed control data coupled with audio channels, and the Orbiter cabin acoustic effects on the operational performance of voice communications. This system allows technical management and project engineering to address key constraining issues, such as identifying design deficiencies of the headset interface unit and the assessment of the Orbiter cabin performance of voice communications, which affect the subsystem development.

Stewart, C. H.↗

Detection of emetic activity in the cat by monitoring venous pressure and audio signals

To investigate the use of audio signals as a simple, noninvasive measure of emetic activity, the relationship between the somatic events and sounds associated with retching and vomiting was studied. Thoracic venous pressure obtained from an implanted external jugular catheter was shown to provide a precise measure of the somatic events associated with retching and vomiting. Changes in thoracic venous pressure monitored through an indwelling external jugular catheter with audio signals, obtained from a microphone located above the animal in a test chamber, were compared. In addition, two independent observers visually monitored emetic episodes. Retching and vomiting were induced by injection of xylazine (0.66mg/kg s.c.), or by motion. A unique audio signal at a frequency of approximately 250 Hz is produced at the time of the negative thoracic venous pressure change associated with retching. Sounds with higher frequencies (around 2500 Hz) occur in conjunction with the positive pressure changes associated with vomiting. These specific signals could be discriminated reliably by individuals reviewing the audio recordings of the sessions. Retching and those emetic episodes associated with positive venous pressure changes were detected accurately by audio monitoring, with 90 percent of retches and 100 percent of emetic episodes correctly identified. Retching was detected more accurately (p is less than .05) by audio monitoring than by direct visual observation. However, with visual observation a few incidents in which stomach contents were expelled in the absence of positive pressure changes or detectable sounds were identified. These data suggest that in emetic situations, the expulsion of stomach contents may be accomplished by more than one neuromuscular system and that audio signals can be used to detect emetic episodes associated with thoracic venous pressure changes.

A Nagahara↗

Laboratory and in-flight experiments to evaluate 3-D audio display technology

Laboratory and in-flight experiments were conducted to evaluate 3-D audio display technology for cockpit applications. A 3-D audio display generator was developed which digitally encodes naturally occurring direction information onto any audio signal and presents the binaural sound over headphones. The acoustic image is stabilized for head movement by use of an electromagnetic head-tracking device. In the laboratory, a 3-D audio display generator was used to spatially separate competing speech messages to improve the intelligibility of each message. Up to a 25 percent improvement in intelligibility was measured for spatially separated speech at high ambient noise levels (115 dB SPL). During the in-flight experiments, pilots reported that spatial separation of speech communications provided a noticeable improvement in intelligibility. The use of 3-D audio for target acquisition was also investigated. In the laboratory, 3-D audio enabled the acquisition of visual targets in about two seconds average response time at 17 degrees accuracy. During the in-flight experiments, pilots correctly identified ground targets 50, 75, and 100 percent of the time at separation angles of 12, 20, and 35 degrees, respectively. In general, pilot performance in the field with the 3-D audio display generator was as expected, based on data from laboratory experiments.

Ericson, Mark↗

Digital Audio Radio Broadcast Systems Laboratory Testing Nearly Complete

Radio history continues to be made at the NASA Lewis Research Center with the completion of phase one of the digital audio radio (DAR) testing conducted by the Consumer Electronics Group of the Electronic Industries Association. This satellite, satellite/terrestrial, and terrestrial digital technology will open up new audio broadcasting opportunities both domestically and worldwide. It will significantly improve the current quality of amplitude-modulated/frequency-modulated (AM/FM) radio with a new digitally modulated radio signal and will introduce true compact-disc-quality (CD-quality) sound for the first time. Lewis is hosting the laboratory testing of seven proposed digital audio radio systems and modes. Two of the proposed systems operate in two modes each, making a total of nine systems being tested. The nine systems are divided into the following types of transmission: in-band on-channel (IBOC), in-band adjacent-channel (IBAC), and new bands. The laboratory testing was conducted by the Consumer Electronics Group of the Electronic Industries Association. Subjective assessments of the audio recordings for each of the nine systems was conducted by the Communications Research Center in Ottawa, Canada, under contract to the Electronic Industries Association. The Communications Research Center has the only CCIR-qualified (Consultative Committee for International Radio) audio testing facility in North America. The main goals of the U.S. testing process are to (1) provide technical data to the Federal Communication Commission (FCC) so that it can establish a standard for digital audio receivers and transmitters and (2) provide the receiver and transmitter industries with the proper standards upon which to build their equipment. In addition, the data will be forwarded to the International Telecommunications Union to help in the establishment of international standards for digital audio receivers and transmitters, thus allowing U.S. manufacturers to compete in the world market.

Source record↗

High-Fidelity Piezoelectric Audio Device

ModalMax is a very innovative means of harnessing the vibration of a piezoelectric actuator to produce an energy efficient low-profile device with high-bandwidth high-fidelity audio response. The piezoelectric audio device outperforms many commercially available speakers made using speaker cones. The piezoelectric device weighs substantially less (4 g) than the speaker cones which use magnets (10 g). ModalMax devices have extreme fabrication simplicity. The entire audio device is fabricated by lamination. The simplicity of the design lends itself to lower cost. The piezoelectric audio device can be used without its acoustic chambers and thereby resulting in a very low thickness of 0.023 in. (0.58 mm). The piezoelectric audio device can be completely encapsulated, which makes it very attractive for use in wet environments. Encapsulation does not significantly alter the audio response. Its small size (see Figure 1) is applicable to many consumer electronic products, such as pagers, portable radios, headphones, laptop computers, computer monitors, toys, and electronic games. The audio device can also be used in automobile or aircraft sound systems.

Woodward, Stanley E.↗

Direct broadcast satellite-audio, portable and mobile reception tradeoffs

This paper reports on the findings of a systems tradeoffs study on direct broadcast satellite-radio (DBS-R). Based on emerging advanced subband and transform audio coding systems, four ranges of bit rates: 16-32 kbps, 48-64 kbps, 96-128 kbps and 196-256 kbps are identified for DBS-R. The corresponding grades of audio quality will be subjectively comparable to AM broadcasting, monophonic FM, stereophonic FM, and CD quality audio, respectively. The satellite EIRP's needed for mobile DBS-R reception in suburban areas are sufficient for portable reception in most single family houses when allowance is made for the higher G/T of portable table-top receivers. As an example, the variation of the space segment cost as a function of frequency, audio quality, coverage capacity, and beam size is explored for a typical DBS-R system.

Golshan, Nasser↗

Worldwide survey of direct-to-listener digital audio delivery systems development since WARC-1992

Each country was allocated frequency band(s) for direct-to-listener digital audio broadcasting at WARC-92. These allocations were near 1500, 2300, and 2600 MHz. In addition, some countries are encouraging the development of digital audio broadcasting services for terrestrial delivery only in the VHF bands (at frequencies from roughly 50 to 300 MHz) and in the medium-wave broadcasting band (AM band) (from roughly 0.5 to 1.7 MHz). The development activity increase was explosive. Current development, as of February 1993, as it is known to the author is summarized. The information given includes the following characteristics, as appropriate, for each planned system: coverage areas, audio quality, number of audio channels, delivery via satellite/terrestrial or both, carrier frequency bands, modulation methods, source coding, and channel coding. Most proponents claim that they will be operational in 3 or 4 years.

Messer, Dion D.↗

Full Mesh Audio Conferencing Using the Point-to-Multipoint On-Board Switching Capability of ACTS

The purpose of this paper is to describe an implementation of audio conferencing using the ACTS T1-VSAT network. In particular, this implementation evaluates the use of the on-board switching capability of the satellite as a viable alternative for providing the multipoint connectivity normally provided by terrestrial audio bridge equipment The system that was implemented provides full mesh, full-duplex audio conferencing, with end-to-end voice paths between all participants requiring only a single hop (i.e. 250 msec. delay). Moreover, it addresses the lack of spontaneity in current systems by allowing a user to easily start a conference from any standard telephone handset connected to an ACTS earth station, and quickly add new members to the conference at any time using the 'hook flash' capability. No prior scheduling of resources is required and there is no central point of control, thereby providing the user with the spontaneity desired in audio conference control.

Rivett, Mary L.↗

Improvements of ModalMax High-Fidelity Piezoelectric Audio Device

ModalMax audio speakers have been enhanced by innovative means of tailoring the vibration response of thin piezoelectric plates to produce a high-fidelity audio response. The ModalMax audio speakers are 1 mm in thickness. The device completely supplants the need to have a separate driver and speaker cone. ModalMax speakers can perform the same applications of cone speakers, but unlike cone speakers, ModalMax speakers can function in harsh environments such as high humidity or extreme wetness. New design features allow the speakers to be completely submersed in salt water, making them well suited for maritime applications. The sound produced from the ModalMax audio speakers has sound spatial resolution that is readily discernable for headset users.

Woodard, Stanley E.↗

PVT+AUDIO. Online Psychoacoustic Testing iOS App for Evaluating AAM/UAM Noise Resonse

PVT+AUDIO is an extension of the existing Psychomoter Vigilance Task application that NASA released over 2 years ago for evaluating pilot fatigue. PVT+AUDIO is designed to present audio stimuli using a method of limits to determine the relationship between objective acoustic parameters and psychacoustic response, for application to Urban Air Mobility (UAM) and Advanced Air Mobility (AAM). Responses include annoyance, acceptance, blend with ambient sound, and detection. The app runs within Apple's iOS ecosystem and utilizes binaural audio simulations of vehicles and ambient sound, and is designed to compensate for headphone response. A prototype experiment is described, along with issues related to calibration and subject engagement.

aircraft noise disturbance↗

Classification and regression models of audio and vibration signals for machine state monitoring in precision machining systems

Here we present a data-driven method for monitoring machine status in manufacturing processes. Audio and vibration data from precision machining are used for inference in two operating scenarios: (a) variable machine health states (anomaly detection); and (b) settings of machine operation (state estimation). Audio and vibration signals are first processed through Fast Fourier Transform and Principal Component Analysis to extract transformed and informative features. These features are then used in the training of classification and regression models for machine state monitoring. Specifically, three classifiers (K-nearest neighbors, convolutional neural networks and support vector machines) and two regressors (support vector regression and neural network regression) were explored, in terms of their accuracy in machine state prediction. It is shown that the audio and vibration signals are sufficiently rich in information about the machine that 100% state classification accuracy could be accomplished. Data fusion was also explored, showing overall superior accuracy of data-driven regression models.

42 ENGINEERING↗

Audio distribution and Monitoring Circuit

Versatile circuit accepts and distributes TV audio signals. Three-meter audio distribution and monitoring circuit provides flexibility in monitoring, mixing, and distributing audio inputs and outputs at various signal and impedance levels. Program material is simultaneously monitored on three channels, or single-channel version built to monitor transmitted or received signal levels, drive speakers, interface to building communications, and drive long-line circuits.

Kirkland, J. M.↗

Virtual environment display for a 3D audio room simulation

The development of a virtual environment simulation system integrating a 3D acoustic audio model with an immersive 3D visual scene is discussed. The system complements the acoustic model and is specified to: allow the listener to freely move about the space, a room of manipulable size, shape, and audio character, while interactively relocating the sound sources; reinforce the listener's feeling of telepresence in the acoustical environment with visual and proprioceptive sensations; enhance the audio with the graphic and interactive components, rather than overwhelm or reduce it; and serve as a research testbed and technology transfer demonstration. The hardware/software design of two demonstration systems, one installed and one portable, are discussed through the development of four iterative configurations.

Chapin, William L.↗