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System is well suited to PBX telephone exchange. Economic electronic changing offers extensive conference capability in telephone communication. System may be used for party to party calls as well.
Simple circuit provides interface between normally incompatible voicecommunication lines. Circuit maintains 40 dB of isolation between input and output halves of four-wire line permitting two-wire line to be connected. Balancing potentiometer, Rg, adjusts gain of IC2 to null feed through from input to output. Adjustment is done on workbench just after assembly.
Satellite sound broadcasting is, as the name implies, the use of satellite techniques and technology to broadcast directly from space to low-cost, consumer-quality receivers the types of sound programs commonly received in the AM and FM broadcast bands. It would be a ubiquitous service available to the general public in the home, in the car, and out in the open.
Beginning with the fourth flight of the Space Transport System (STS-4), objective measurements of inner ear function were conducted in near-zero G conditions in earth orbit. The problem of space motion sickness (SMS) was approached much like any disequilibrium problem encountered clinically. However, objective testing techniques had built-in limitations superimposed by the strict parameters inherent in each mission. An attempt was made to objectively characterize SMS, and to first ascertain whether the objective measurements indicated that this disorder was of peripheral or central origin. Electronystagmography and auditory brain stem response recordings were the primary investigative tools. One of the authors (W.E.T.) was a mission specialist on board the orbiter Challenger on the eighth shuttle mission (STS-8) and had the opportunity to make direct and personal observations regarding SMS, an opportunity which has added immeasurably to our understanding of this disorder. Except for two abnormal ENG records, which remain to be explained, the remaining ENG records and all the ABR records made in the weightless environment of space were normal.
Numerous applications benefit from the presentation of computer-generated auditory stimuli at points discontiguous with the computer itself. Modification of an IBM-compatible computer for use of an external speaker is relatively easy but not intuitive. This modification is briefly described.
Weather is one of the major causes of general aviation accidents. Researchers are addressing this problem from various perspectives including improving meteorological forecasting techniques, collecting additional weather data automatically via on-board sensors and "flight" modems, and improving weather data dissemination and presentation. We approach the problem from the improved presentation perspective and propose weather visualization and interaction methods tailored for general aviation pilots. Our system, Aviation Weather Data Visualization Environment (AWE), utilizes information visualization techniques, a direct manipulation graphical interface, and a speech-based interface to improve a pilot's situational awareness of relevant weather data. The system design is based on a user study and feedback from pilots.
The challenges facing successful multimedia presentation depend largely on the expectations of the designer and end user for a given application. Perceptual limitations in distance, elevation and azimuth sound source simulation differ significantly between headphone and cross-talk cancellation loudspeaker listening and therefore must be considered. Simulation of an environmental context is desirable but the quality depends on processing resources and lack of interaction with the host acoustical environment. While techniques such as data reduction of head-related transfer functions have been used widely to improve simulation fidelity, another approach involves determining thresholds for environmental acoustic events. Psychoacoustic studies relevant to this approach are reviewed in consideration of multimedia applications
A real-time vector adaptive predictive coder which approximates each vector of K speech samples by using each of M fixed vectors in a first codebook to excite a time-varying synthesis filter and picking the vector that minimizes distortion. Predictive analysis for each frame determines parameters used for computing from vectors in the first codebook zero-state response vectors that are stored at the same address (index) in a second codebook. Encoding of input speech vectors s.sub.n is then carried out using the second codebook. When the vector that minimizes distortion is found, its index is transmitted to a decoder which has a codebook identical to the first codebook of the decoder. There the index is used to read out a vector that is used to synthesize an output speech vector s.sub.n. The parameters used in the encoder are quantized, for example by using a table, and the indices are transmitted to the decoder where they are decoded to specify transfer characteristics of filters used in producing the vector s.sub.n from the receiver codebook vector selected by the vector index transmitted.
A vector excitation coder compresses vectors by using an optimum codebook designed off line, using an initial arbitrary codebook and a set of speech training vectors exploiting codevector sparsity (i.e., by making zero all but a selected number of samples of lowest amplitude in each of N codebook vectors). A fast-search method selects a number N.sub.c of good excitation vectors from the codebook, where N.sub.c is much smaller tha ORIGIN OF INVENTION The invention described herein was made in the performance of work under a NASA contract, and is subject to the provisions of Public Law 96-517 (35 USC 202) under which the inventors were granted a request to retain title.
A voice-command human-machine interface system has been developed for spacesuit extravehicular activity (EVA) missions. A multichannel acoustic signal processing method has been created for distant speech acquisition in noisy and reverberant environments. This technology reduces noise by exploiting differences in the statistical nature of signal (i.e., speech) and noise that exists in the spatial and temporal domains. As a result, the automatic speech recognition (ASR) accuracy can be improved to the level at which crewmembers would find the speech interface useful. The developed speech human/machine interface will enable both crewmember usability and operational efficiency. It can enjoy a fast rate of data/text entry, small overall size, and can be lightweight. In addition, this design will free the hands and eyes of a suited crewmember. The system components and steps include beam forming/multi-channel noise reduction, single-channel noise reduction, speech feature extraction, feature transformation and normalization, feature compression, model adaption, ASR HMM (Hidden Markov Model) training, and ASR decoding. A state-of-the-art phoneme recognizer can obtain an accuracy rate of 65 percent when the training and testing data are free of noise. When it is used in spacesuits, the rate drops to about 33 percent. With the developed microphone array speech-processing technologies, the performance is improved and the phoneme recognition accuracy rate rises to 44 percent. The recognizer can be further improved by combining the microphone array and HMM model adaptation techniques and using speech samples collected from inside spacesuits. In addition, arithmetic complexity models for the major HMMbased ASR components were developed. They can help real-time ASR system designers select proper tasks when in the face of constraints in computational resources.
Test process, milestones and inputs are unknowns to first-time users of the ADL. The User Test Planning Guide aids in establishing expectations for both NASA and non-NASA facility customers. The potential audience for this guide includes both internal and commercial spaceflight hardware/software developers. It is intended to assist their test engineering personnel in test planning and execution. Material covered includes a roadmap of the test process, roles and responsibilities of facility and user, major milestones, facility capabilities, and inputs required by the facility. Samples of deliverables, test article interfaces, and inputs necessary to define test scope, cost, and schedule are included as an appendix to the guide.
Neuroscientist Seth Horowitz was part of a NASA-funded team at State University of New York Stony Brook demonstrating that low-amplitude vestibular stimulation could induce sleep. After recognizing the same stimulation could be applied through sound, Horowitz founded Sleep Genius, located in Park City, Utah, and released a mobile app of the same name that helps people to get a more restful sleep.
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.
What is anomaly detection? The process of identifying rare, unexpected, or abnormal patterns in data.
Systems and methods for detecting a tampering and identifying a location of a digital recording are provided. A frequency sequence and a phase angle sequence may be extracted from the digital recording. A portion of the frequency sequence may be matched to one of a plurality of reference frequency sequences, and a portion of the phase angle sequence may be matched to one of a plurality of reference phase angle sequences. Tampering of the digital recording may be detected when the frequency and phase sequences differ from the matched reference sequences. Moreover, a noise sequence may be extracted from the extracted frequency sequence. A location of the digital recording may be identified by matching the noise sequence to one of a plurality of noise sequences of the plurality of reference frequency sequences.
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.
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.
Delivering audible content to a targeted listener without disturbing others is paramount in audio engineering. However, achieving this goal has long been challenging due to the diffraction of low-frequency (long-wavelength) audio waves in linear acoustics. Here, we introduce an approach for creating remote audio spots, dubbed audible enclaves, by harnessing the local nonlinear interaction of two self-bending ultrasonic beams with distinct spectra. The self-bending ultrasonic beams created by acoustic metasurfaces, though inaudible, can bypass obstacles such as human heads. At their intersection behind obstacles, highly localized audible enclaves are formed due to the local nonlinear interactions. Additionally, we demonstrate the ultrabroadband capabilities of our metasurface-based implementation both numerically and experimentally, spanning from 125 Hz to 4 kHz (6 octave bands), covering the majority of the audible frequency range. The practicality of our proposed technique is underscored by its compact implementation size (0.16 m, equivalent to 0.06 wavelengths at 125 Hz), as well as its robust performance under wideband transient audio signal excitation and in a common room with reverberations. Our proposed audible enclaves hold significant potential for various applications in advanced audio engineering, including private speech communications, immersive spatial audio reproduction, and high-resolution sound/quiet zone control.