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Search indexed NASA NTRS and DOE OSTI research on propulsion, heat transfer, battery materials and energy systems. Follow report and document links to the original sources.

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At least 109 records · Page 6

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.↗

Near-toll quality digital speech transmission in the mobile satellite service

This paper discusses system considerations for near-toll quality digital speech transmission in a 5 kHz mobile satellite system channel. Tradeoffs are shown for power performance versus delay for a 4800 bps speech compression system in conjunction with a 16 state rate 2/3 trellis coded 8PSK modulation system. The suggested system has an additional 150 ms of delay beyond the propagation delay and requires an E(b)/N(0) of about 7 dB for a Ricean channel assumption with line-of-sight to diffuse component ratio of 10 assuming ideal synchronization. An additional loss of 2 to 3 dB is expected for synchronization in fading environment.

Townes, S. A.↗

4800 B/S speech compression techniques for mobile satellite systems

This paper will discuss three 4800 bps digital speech compression techniques currently being investigated for application in the mobile satellite service. These three techniques, vector adaptive predictive coding, vector excitation coding, and the self excited vocoder, are the most promising among a number of techniques being developed to possibly provide near-toll-quality speech compression while still keeping the bit-rate low enough for a power and bandwidth limited satellite service.

Townes, S. A.↗

Pitch-Learning Algorithm For Speech Encoders

Adaptive algorithm detects and corrects errors in sequence of estimates of pitch period of speech. Algorithm operates in conjunction with techniques used to estimate pitch period. Used in such parametric and hybrid speech coders as linear predictive coders and adaptive predictive coders.

Bhaskar, B. R. Udaya↗

Neural network based speech synthesizer: A preliminary report

A neural net based speech synthesis project is discussed. The novelty is that the reproduced speech was extracted from actual voice recordings. In essence, the neural network learns the timing, pitch fluctuations, connectivity between individual sounds, and speaking habits unique to that individual person. The parallel distributed processing network used for this project is the generalized backward propagation network which has been modified to also learn sequences of actions or states given in a particular plan.

Villarreal, James A.↗

Smart command recognizer (SCR) - For development, test, and implementation of speech commands

The SCR, a rapid prototyping system for the development, testing, and implementation of speech commands in a flight simulator or test aircraft, is described. A single unit performs all functions needed during these three phases of system development, while the use of common software and speech command data structure files greatly reduces the preparation time for successive development phases. As a smart peripheral to a simulation or flight host computer, the SCR interprets the pilot's spoken input and passes command codes to the simulation or flight computer.

Simpson, Carol A.↗

More About Vector Adaptive/Predictive Coding Of Speech

Report presents additional information about digital speech-encoding and -decoding system described in "Vector Adaptive/Predictive Encoding of Speech" (NPO-17230). Summarizes development of vector adaptive/predictive coding (VAPC) system and describes basic functions of algorithm. Describes refinements introduced enabling receiver to cope with errors. VAPC algorithm implemented in integrated-circuit coding/decoding processors (codecs). VAPC and other codecs tested under variety of operating conditions. Tests designed to reveal effects of various background quiet and noisy environments and of poor telephone equipment. VAPC found competitive with and, in some respects, superior to other 4.8-kb/s codecs and other codecs of similar complexity.

Jedrey, Thomas C.↗

Headphone localization of speech stimuli

Recently, three dimensional acoustic display systems have been developed that synthesize virtual sound sources over headphones based on filtering by Head-Related Transfer Functions (HRTFs), the direction-dependent spectral changes caused primarily by the outer ears. Here, 11 inexperienced subjects judged the apparent spatial location of headphone-presented speech stimuli filtered with non-individualized HRTFs. About half of the subjects 'pulled' their judgements toward either the median or the lateral-vertical planes, and estimates were almost always elevated. Individual differences were pronounced for the distance judgements; 15 to 46 percent of stimuli were heard inside the head with the shortest estimates near the median plane. The results infer that most listeners can obtain useful azimuth information from speech stimuli filtered by nonindividualized RTFs. Measurements of localization error and reversal rates are comparable with a previous study that used broadband noise stimuli.

Begault, Durand R.↗

Headphone localization of speech

Three-dimensional acoustic display systems have recently been developed that synthesize virtual sound sources over headphones based on filtering by head-related transfer functions (HRTFs), the direction-dependent spectral changes caused primarily by the pinnae. In this study, 11 inexperienced subjects judged the apparent spatial location of headphone-presented speech stimuli filtered with nonindividualized HRTFs. About half of the subjects 'pulled' their judgments toward either the median or the lateral-vertical planes, and estimates were almost always elevated. Individual differences were pronounced for the distance judgments; 15 to 46 percent of stimuli were heard inside the head, with the shortest estimates near the median plane. The results suggest that most listeners can obtain useful azimuth information from speech stimuli filtered by nonindividualized HRTFs. Measurements of localization error and reversal rates are comparable with a previous study that used broadband noise stimuli.

Begault, Durand R.↗

Military applications of automatic speech recognition and future requirements

An updated summary of the state-of-the-art of automatic speech recognition and its relevance to military applications is provided. A number of potential systems for military applications are under development. These include: (1) digital narrowband communication systems; (2) automatic speech verification; (3) on-line cartographic processing unit; (4) word recognition for militarized tactical data system; and (5) voice recognition and synthesis for aircraft cockpit.

Beek, Bruno↗

Statistical assessment of speech system performance

Methods for the normalization of performance tests results of speech recognition systems are presented. Technological accomplishments in speech recognition systems, as well as planned research activities are described.

Moshier, Stephen L.↗

Performing speech recognition research with hypercard

The purpose of this paper is to describe a HyperCard-based system for performing speech recognition research and to instruct Human Factors professionals on how to use the system to obtain detailed data about the user interface of a prototype speech recognition application.

Shepherd, Chip↗

Multi-channel spatial auditory display for speech communications

A spatial auditory display for multiple speech communications was developed at NASA-Ames Research Center. Input is spatialized by use of simplified head-related transfer functions, adapted for FIR filtering on Motorola 56001 digital signal processors. Hardware and firmware design implementations are overviewed for the initial prototype developed for NASA-Kennedy Space Center. An adaptive staircase method was used to determine intelligibility levels of four letter call signs used by launch personnel at NASA, against diotic speech babble. Spatial positions at 30 deg azimuth increments were evaluated. The results from eight subjects showed a maximal intelligibility improvement of about 6 to 7 dB when the signal was spatialized to 60 deg or 90 deg azimuth positions.

Begault, Durand↗

Speech Aids

Designed to assist deaf and hearing impaired-persons in achieving better speech, Resnick Worldwide Inc.'s device provides a visual means of cuing the deaf as a speech-improvement measure. This is done by electronically processing the subjects' sounds and comparing them with optimum values which are displayed for comparison.

Source record↗

Method for detection and correction of errors in speech pitch period estimates

A method of detecting and correcting received values of a pitch period estimate of a speech signal for use in a speech coder or the like. An average is calculated of the nonzero values of received pitch period estimate since the previous reset. If a current pitch period estimate is within a range of 0.75 to 1.25 times the average, it is assumed correct, while if not, a correction process is carried out. If correction is required successively for more than a preset number of times, which will most likely occur when the speaker changes, the average is discarded and a new average calculated.

Bhaskar, Udaya↗

Integrated Spacesuit Audio System Enhances Speech Quality and Reduces Noise

A new approach has been proposed for increasing astronaut comfort and speech capture. Currently, the special design of a spacesuit forms an extreme acoustic environment making it difficult to capture clear speech without compromising comfort. The proposed Integrated Spacesuit Audio (ISA) system is to incorporate the microphones into the helmet and use software to extract voice signals from background noise.

Huang, Yiteng Arden↗

Tampa Bay International Business Summit Keynote Speech

A keynote speech outlining the importance of collaboration and diversity in the workplace. The 20-minute speech describes NASA's challenges and accomplishments over the years and what lies ahead. Topics include: diversity and inclusion principles, international cooperation, Kennedy Space Center planning and development, opportunities for cooperation, and NASA's vision for exploration.

Clary, Christina↗

Performance Evaluation of Speech Recognition Systems as a Next-Generation Pilot-Vehicle Interface Technology

During the flight trials known as Gulfstream-V Synthetic Vision Systems Integrated Technology Evaluation (GV-SITE), a Speech Recognition System (SRS) was used by the evaluation pilots. The SRS system was intended to be an intuitive interface for display control (rather than knobs, buttons, etc.). This paper describes the performance of the current "state of the art" Speech Recognition System (SRS). The commercially available technology was evaluated as an application for possible inclusion in commercial aircraft flight decks as a crew-to-vehicle interface. Specifically, the technology is to be used as an interface from aircrew to the onboard displays, controls, and flight management tasks. A flight test of a SRS as well as a laboratory test was conducted.

Arthur, Jarvis J., III↗