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1

Reading Machine: From Text to Speech.  

National Technical Information Service (NTIS)

A machine with unrestricted vocabulary, that is capable of converting printed text into connected speech in real time, would be extremely useful to blind people. The problems in implementing such a machine are mainly (1) character recognition, (2) convers...

F. F. Lee

1969-01-01

2

Speak It! Text to Speech app.  

PubMed

You can copy emails, documents, web pages and PDF files, paste the text into Speak it! and have the text spoken back out loud. The app creates audio files of the text that can be stored and played repeatedly or emailed as attachments. PMID:25074111

Evans, Nicola

2014-07-30

3

Improving health literacy: a Web application for evaluating text-to-speech engines.  

PubMed

The Internet is increasingly used as a medium for gathering and exchanging health information exchange. Healthcare professionals and organizations need to consider barriers that may exist within their patient-oriented Web applications. One approach to making the Web more accessible for those with lower health literacy may be to supplement textual content with audio annotation using text-to-speech engines, allowing for the creation of a virtual surrogate reader. One challenge is that with numerous text-to-speech engines on the market, objective measures of quality are difficult to obtain. To facilitate comparisons of text-to-speech engines, we developed an open-source Web application that measures user reaction times, subjective quality ratings, and accuracy in completing tasks across different audio files created by text-to-speech engines. Our research endeavor was successful in building and piloting this Web application; significant differences were found for subjective ratings of quality across three text-to-speech engines priced at different levels. However, no significant differences were found with reaction times or accuracy between these text-to-speech engines. Future avenues of research include exploring more complex tasks, usability issues related to implementing text-to-speech features, and applied health promotion and education opportunities among vulnerable populations. PMID:20571370

Wolpin, Seth; Berry, Donna L; Kurth, Ann; Lober, William B

2010-01-01

4

Prosodic phrasing in a Polish text-to-speech system  

Microsoft Academic Search

This contribution presents the linguistic research underlying the implementation of prosodic phrasing in a Polish text-to-speech system 1. While in the past few years concatenative text-to - speech technology dramatically improved the acoustic quality of the synthesized voices, yet the naturalness and expressivity of present text-to-spe ech systems are still unsatisfactory. In particular , these systems usually read with a

Morena Danieli; Beata Dobrzy; Alberto Pacchiotti; Elena Cabrio

5

Choosing and Using Text-to-Speech Software  

ERIC Educational Resources Information Center

This article describes a computer-based technology for generating speech called text-to-speech (TTS). This software is ready for widespread use by libraries, other organizations, and individual users. It offers the affordable ability to turn just about any electronic text that is not image-based into an artificially spoken communication. The…

Peters, Tom; Bell, Lori

2007-01-01

6

Recent Improvements on Microsoft's Trainable Text - to - Speech System: Whistler  

Microsoft Academic Search

The Whistler text-to-speech engine was designed so that we can automatically construct the model parameters from training data. This paper focuses on the improvements on prosody and acoustic modeling, which are all derived through the use of probabilistic learning methods. Whistler can produce synthetic speech that sounds very natural and resembles the acoustic and prosodic characteristics of the original speaker.

Xuedong Huang; Alex Acero; Hsiao-wuen Hon; J Liu; S Meredith; M Plumpe

1997-01-01

7

SMaTalk: Standard Malay Text to Speech Talk System  

Microsoft Academic Search

This paper presents a rule-based text- to- speech (TTS) Synthesis System for Standard Malay, namely SMaTTS. The proposed system using sinusoidal method and some pre- recorded wave files in generating speech for the system. The use of phone database significantly decreases the amount of computer memory space used, thus making the system very light and embeddable. The overall system was

Othman O. Khalifa; Zakiah Hanim Ahmad; Aisha-Hassan A. Hashim; Teddy Suya Gunawan

8

SMaTTS: Standard Malay Text to Speech System  

Microsoft Academic Search

This paper presents a rule-based text- to- speech (TTS) Synthesis System for Standard Malay, namely SMaTTS. The proposed system using sinusoidal method and some pre- recorded wave files in generating speech for the system. The use of phone database significantly decreases the amount of computer memory space used, thus making the system very light and embeddable. The overall system was

Othman O. Khalifa; Zakiah Hanim Ahmad; Teddy Surya Gunawan

9

Novel Text-To-Speech Reading Modes for Educational Applications  

Microsoft Academic Search

In this paper we describe the development of two new text-to-speech reading modes for use in a com- puterised reading tutor for dyslexic children: a phoneme spelling mode which spells words phoneme by phoneme and a syllable mode, which allows speech to be synthesised as ei- ther isolated or lengthened syllables. These modes are used for modelling (i.e. to give

Lukas Latacz; Yuk On Kong; Wesley Mattheyses; Werner Verhelst

10

Review of text-to-speech conversion for English.  

PubMed

The automatic conversion of English text to synthetic speech is presently being performed, remarkably well, by a number of laboratory systems and commercial devices. Progress in this area has been made possible by advances in linguistic theory, acoustic-phonetic characterization of English sound patterns, perceptual psychology, mathematical modeling of speech production, structured programming, and computer hardware design. This review traces the early work on the development of speech synthesizers, discovery of minimal acoustic cues for phonetic contrasts, evolution of phonemic rule programs, incorporation of prosodic rules, and formulation of techniques for text analysis. Examples of rules are used liberally to illustrate the state of the art. Many of the examples are taken from Klattalk, a text-to-speech system developed by the author. A number of scientific problems are identified that prevent current systems from achieving the goal of completely human-sounding speech. While the emphasis is on rule programs that drive a format synthesizer, alternatives such as articulatory synthesis and waveform concatenation are also reviewed. An extensive bibliography has been assembled to show both the breadth of synthesis activity and the wealth of phenomena covered by rules in the best of these programs. A recording of selected examples of the historical development of synthetic speech, enclosed as a 33 1/3-rpm record, is described in the Appendix. PMID:2958525

Klatt, D H

1987-09-01

11

"Look What I Did!": Student Conferences with Text-to-Speech Software  

ERIC Educational Resources Information Center

The authors describe a strategy that empowers students to edit and revise their own writing. Students input their writing in to text-to-speech software that rereads the text aloud. While listening, students make necessary revisions and edits.

Young, Chase; Stover, Katie

2014-01-01

12

A High Quality Text-To-Speech System Composed of Multiple Neural Networks  

Microsoft Academic Search

While neural networks have been employed to handle several different text-to-speech tasks, ours is the first s ystem to use neural networks throughout, for both linguistic and acoustic processing. We divide the text-to-speech task into three subtasks, a linguistic module mapping from text to a linguistic representation, an acoustic module mapping from the linguistic representation to speech, and a video

Orhan Karaali; Gerald Corrigan; Noel Massey; Corey Miller; Otto Schnurr; Andrew Mackie

1998-01-01

13

An enhanced pitch modeling supporting a Greek Text to Speech system  

Microsoft Academic Search

This paper tries to describe as accurately as possible an enhanced procedure for predicting the appropriate prosodic structure of speech and apply it to a Greek Text to Speech system. The main focus will be a particular linear stylization of the fundamental frequency function F0 contour (pitch), by using the most important syntactic, grammatical and lexical features of the Greek

ILIAS SPAIS; GEORGE BAFAS; XENOFON PAPADOPOULOS

14

Embedded unit selection text-to-speech synthesis for mobile devices  

Microsoft Academic Search

Nowadays, unit selection based text-to-speech technology is the mainstream approach for near natural speech synthesis systems. However, this is achieved at the expense of raised requirements in terms of computational resources. This work describes design and implementation approaches for the efficient integration of this technology in computational environments with limited resources, such as mobile devices, with no considerable speech quality

Sotiris Karabetsos; Pirros Tsiakoulis; Aimilios Chalamandaris; Spyros Raptis

2009-01-01

15

Text-to-Speech, Text, and Hypertext: Reading and Spelling with the Computer.  

ERIC Educational Resources Information Center

Introduces this special issue. Discusses the analysis-by-synthesis principle of text-to-speech conversion; some classroom and research issues in designing "usable" computer texts; and the criteria of hypertext. Emphasizes the importance of the contextual aspects of text and hypertext and situated learning. (RS)

Leong, Che Kan

1992-01-01

16

Using Text-to-Speech Reading Support for an Adult with Mild Aphasia and Cognitive Impairment  

ERIC Educational Resources Information Center

This single case study served to examine text-to-speech (TTS) effects on reading rate and comprehension in an individual with mild aphasia and cognitive impairment. Findings showed faster reading, given TTS presented at a normal speaking rate, but no significant comprehension changes. TTS may support reading in people with aphasia when time…

Harvey, Judy; Hux, Karen; Snell, Jeffry

2013-01-01

17

Training intonational phrasing rules automatically for English and Spanish text-to-speech  

Microsoft Academic Search

We describe a procedure for acquiring intonational phrasing rules for text-to-speech synthesis automatically, from annotated text, and some evaluation of this procedure for English and Spanish. The procedure employs decision trees generated automatically, using Classification and Regression Tree techniques, from text corpora which have been hand-labeled by native speakers with likely locations of intonational boundaries, in conjunction with information available

Julia Hirschberg; Pilar Prieto

1996-01-01

18

Faking it: Synthetic text-to-speech synthesis for u nder-resourced languages - Experimental design  

Microsoft Academic Search

Speech synthesis or text-to-speech (TTS) systems are currently available for a number of the world's major languages, but for thousands of the world's 'minor' languages no such technology is available. While awaiting the development of such technology, we would like to try the stop-gap solution of using an existing TTS system for a major language (the base language) to 'fake'

Harold Somers

19

A unit selection text-to-speech synthesis system optimized for use with screen readers  

Microsoft Academic Search

Currently, unit-selection text-to-speech technology is the common approach for near-natural speech synthesis systems. Such systems provide an important aid for blind or partially-sighted people, when combined with screen reading software. However, although the overall quality of the synthetic speech achieved by such systems can be quite high, this fact alone does not guarantee a high level of user satisfaction. Many

Aimilios Chalamandaris; Sotiris Karabetsos; Pirros Tsiakoulis; Spyros Raptis

2010-01-01

20

Comparative evaluation of the speech quality of speech coders and text-to-speech synthesizers  

Microsoft Academic Search

In a joint project called SPIN, which is sponsored by the European Information Technology Program ESPRIT, a speech interface for office automation will be developed and tested. Two specific aspects of such an interface, which are discussed in this paper, have to do with speech store-and-forward and text-to-speech synthesis-by-rule. We will diagnostically evaluate the speech quality of the medium-band coders

L. C. W. Pols; G. W. Boxelaar

1986-01-01

21

Perception of synthetic speech produced automatically by rule: Intelligibility of eight text-to-speech systems.  

PubMed

We present the results of studies designed to measure the segmental intelligibility of eight text-to-speech systems and a natural speech control, using the Modified Rhyme Test (MRT). Results indicated that the voices tested could be grouped into four categories: natural speech, high-quality synthetic speech, moderate-quality synthetic speech, and low-quality synthetic speech. The overall performance of the best synthesis system, DECtalk-Paul, was equivalent to natural speech only in terms of performance on initial consonants. The findings are discussed in terms of recent work investigating the perception of synthetic speech under more severe conditions. Suggestions for future research on improving the quality of synthetic speech are also considered. PMID:23225916

Greene, Beth G; Logan, John S; Pisoni, David B

1986-03-01

22

Adaptive and Longitudinal Pharmaceutical Care Instruction Using an Interactive Voice Response/Text-to-Speech System  

PubMed Central

Objectives To develop a course structure that would more closely simulate the actual provision of pharmaceutical care. Design An interactive voice response/text-to-speech system (hardware and software) for obtaining patient data was designed and used in a pharmaceutical care laboratory. Students called the system to collect data, listen to progress notes, make recommendations, and update the pharmaceutical care plan for virtual patients. Laboratory time was utilized to evaluate patient progress and respond to recommendations as well as to identify and solve drug-related problems. Assessment Students' recorded communications with the system and completed care plans were evaluated and a competency-based final examination was administered. Peer evaluations and course evaluations were administered. Conclusion This innovative approach challenged students and promoted interactive learning. Student evaluations indicated we achieved our objective of creating a course that more closely simulated the actual provision of pharmaceutical care.

Hussein, Gamal; Kawahara, Nancy

2006-01-01

23

Advancements in text-to-speech technology and implications for AAC applications  

NASA Astrophysics Data System (ADS)

Intelligibility was the initial focus in text-to-speech (TTS) research, since it is clearly a necessary condition for the application of the technology. Sufficiently high intelligibility (approximating human speech) has been achieved in the last decade by the better formant-based and concatenative TTS systems. This led to commercially available TTS systems for highly motivated users, particularly the blind and vocally impaired. Some unnatural qualities of TTS were exploited by these users, such as very fast speaking rates and altered pitch ranges for flagging relevant information. Recently, the focus in TTS research has turned to improving naturalness, so that synthetic speech sounds more human and less robotic. Unit selection approaches to concatenative synthesis have dramatically improved TTS quality, although at the cost of larger and more complex systems. This advancement in naturalness has made TTS technology more acceptable to the general public. The vocally impaired appreciate a more natural voice with which to represent themselves when communicating with others. Unit selection TTS does not achieve such high speaking rates as the earlier TTS systems, however, which is a disadvantage to some AAC device users. An important new research emphasis is to improve and increase the range of emotional expressiveness of TTS.

Syrdal, Ann K.

2003-10-01

24

A Variable Break Prediction Method Using CART in a Japanese Text-to-Speech System  

NASA Astrophysics Data System (ADS)

Break prediction is an important step in text-to-speech systems as break indices (BIs) have a great influence on how to correctly represent prosodic phrase boundaries. However, an accurate prediction is difficult since BIs are often chosen according to the meaning of a sentence or the reading style of the speaker. In Japanese, the prediction of an accentual phrase boundary (APB) and major phrase boundary (MPB) is particularly difficult. Thus, this paper presents a method to complement the prediction errors of an APB and MPB. First, we define a subtle BI in which it is difficult to decide between an APB and MPB clearly as a variable break (VB), and an explicit BI as a fixed break (FB). The VB is chosen using the classification and regression tree, and multiple prosodic targets in relation to the pith and duration are then generated. Finally, unit-selection is conducted using multiple prosodic targets. The experimental results show that the proposed method improves the naturalness of synthesized speech.

Na, Deok-Su; Bae, Myung-Jin

25

Segmental intelligibility of four currently used text-to-speech synthesis methods  

NASA Astrophysics Data System (ADS)

The study investigated the segmental intelligibility of four currently available text-to-speech (TTS) products under 0-dB and 5-dB signal-to-noise ratios. The products were IBM ViaVoice™ version 5.1, which uses formant coding, Festival version 1.4.2, a diphone-based LPC TTS product, AT&T Next-Gen™, a half-phone-based TTS product that uses harmonic-plus-noise method for synthesis, and FlexVoice™2, a hybrid TTS product that combines concatenative and formant coding techniques. Overall, concatenative techniques were more intelligible than formant or hybrid techniques, with formant coding slightly better at modeling vowels and concatenative techniques marginally better at synthesizing consonants. No TTS product was better at resisting noise interference than others, although all were more intelligible at 5 dB than at 0-dB SNR. The better TTS products in this study were, on the average, 22% less intelligible and had about 3 times more phoneme errors than human voice under comparable listening conditions. The hybrid TTS technology of FlexVoice had the lowest intelligibility and highest error rates. There were discernible patterns of errors for stops, fricatives, and nasals. Unrestricted TTS output-e-mail messages, news reports, and so on-under high noise conditions prevalent in automobiles, airports, etc. will likely challenge the listeners.

Venkatagiri, Horabail S.

2003-04-01

26

Text-to-speech from concatenation of articulatory units derived from natural speech  

NASA Astrophysics Data System (ADS)

It has been conjectured that articulatory synthesis possesses the greatest potential for generating high quality synthetic speech. However, for text-to-speech (TTS), waveform concatenation techniques have proven more practical due in part to the challenge of generating appropriate trajectories of articulatory parameters. A waveform generation method for TTS that combines the practical success of concatenative methods with the quality potential of articulatory synthesis is under development. The system concatenates articulatory units derived from natural speech using an articulatory voice mimic. The mimic estimates articulatory parameters by minimizing a cost function that includes a spectral distance between natural and synthetic speech and a geometric distance that penalizes rapid or discontinuous changes in articulator positions. A database of articulatory trajectories representing phonetic units is constructed from the estimated parameters. For TTS, phonetic units generated by text analysis are used to select the corresponding articulatory units from the database. Duration modification, concatenation, and smoothing across units are performed in the articulatory domain resulting in a single articulatory trajectory for the complete utterance. Speech is synthesized from the trajectory using a two mass model for voicing, achieving a high degree of acoustic continuity across unit boundaries while also allowing for source-tract interaction.

Sinder, Daniel J.; Sondhi, M. Mohan

2003-04-01

27

Segmental intelligibility of four currently used text-to-speech synthesis methods.  

PubMed

The study investigated the segmental intelligibility of four currently available text-to-speech (TTS) products under 0-dB and 5-dB signal-to-noise ratios. The products were IBM ViaVoice version 5.1, which uses formant coding, Festival version 1.4.2, a diphone-based LPC TTS product, AT&T Next-Gen, a half-phone-based TTS product that uses harmonic-plus-noise method for synthesis, and FlexVoice2, a hybrid TTS product that combines concatenative and formant coding techniques. Overall, concatenative techniques were more intelligible than formant or hybrid techniques, with formant coding slightly better at modeling vowels and concatenative techniques marginally better at synthesizing consonants. No TTS product was better at resisting noise interference than others, although all were more intelligible at 5 dB than at 0-dB SNR. The better TTS products in this study were, on the average, 22% less intelligible and had about 3 times more phoneme errors than human voice under comparable listening conditions. The hybrid TTS technology of FlexVoice had the lowest intelligibility and highest error rates. There were discernible patterns of errors for stops, fricatives, and nasals. Unrestricted TTS output--e-mail messages, news reports, and so on--under high noise conditions prevalent in automobiles, airports, etc. will likely challenge the listeners. PMID:12703720

Venkatagiri, Horabail S

2003-04-01

28

Audio  

Microsoft Academic Search

\\u000a If you’re one of those who treat audio in Flash as an afterthought, think again. In many respects, audio is a major medium\\u000a for communicating your message. In this chapter, we dig into audio in Flash: where it comes from, what formats are supported,\\u000a and how to use it in Flash. Regardless whether you’re new to Flash or an old

Tom Green; David Stiller

29

A joint intelligibility evaluation of French text-to-speech synthesis systems: the EvaSy SUS\\/ACR campaign  

Microsoft Academic Search

The EVALDA\\/EvaSy project is dedicated to the evaluation of text-to-speech synthesis systems for the French language. It is subdivided into four components: evaluation of the grapheme-to-phoneme conversion module (Boula de Mareil et al., 2005), evaluation of prosody (Garcia et al., 2006), evaluation of intelligibility, and global evaluation of the quality of the synthesised speech. This paper reports on the key

Philippe Boula de Mareüil; Christophe d'Alessandro; Alexander Raake; Gérard Bailly; Marie-Neige Garcia; Michel Morel

30

Use of the magnitude estimation technique for assessing the performance of text-to-speech synthesis systems.  

PubMed

As text-to-speech systems develop, it becomes necessary to compare various solutions and to evaluate whether a change in the synthesis procedure has an effect on the listener's attitude to the system. The possibility of directly scaling intelligibility, naturalness, and user's satisfaction (i.e., acceptability) with the magnitude estimation technique is investigated. A magnitude estimation protocol suitable for this purpose is described. In general, within the limits of the methodological constraints discussed in this paper, the procedure appears to be reliable and valid for quantifying the perceived attributes of synthesized speech. PMID:2137144

Pavlovic, C V; Rossi, M; Espesser, R

1990-01-01

31

Analysis on Effects of Text-to-Speech and Avatar Agent in Evoking Users’ Spontaneous Listener’s Reactions  

Microsoft Academic Search

\\u000a This paper reports an analysis on effect of text-to-speech (TTS) and avatar agent in evoking user’s user’s spontaneous backchannels.\\u000a We construct an HMMbased dialogue-style TTS system that generates human-like cues that evoke users’ backchannels. We also\\u000a constructed an avatar agent that can make several listener’s reactions. A spoken dialogue system for information navigation\\u000a was implemented and was evaluated in terms

Teruhisa Misu; Etsuo Mizukami; Yoshinori Shiga; Shinichi Kawamoto; Hisashi Kawai; Satoshi Nakamura

32

Effects of Text-to-Speech Software on the Reading Rate and Comprehension Skills of High School Students with Specific Learning Disabilities  

ERIC Educational Resources Information Center

The purpose of this study was to examine the effects of a text-to-speech software program known as "Read Please" on the reading rate and reading comprehension accuracy of two high school students with specific learning disabilities (SLD) in reading. A single-subject A-B-A-B "withdrawal" research design (Alberto & Troutman, 2009) was used to…

Moorman, Amanda; Boon, Richard T.; Keller-Bell, Yolanda; Stagliano, Christina; Jeffs, Tara

2010-01-01

33

Listening to Revise: What a Study about Text-to-Speech Software Taught Us about Students' Expectations for Technology Use in the Writing Center  

ERIC Educational Resources Information Center

This is a story of a failed study. In 2007, the authors set out to demonstrate that Kurzweil 3000, an adaptive text-to-speech software program, would help any student revise with its read-aloud function and numerous writing tools. During the course of the study, the authors confronted their misconceptions about students' technology use and…

Conard-Salvo, Tammy; Spartz, John M.

2012-01-01

34

CONTEXT-AWARE MOBILE LEARNING WITH PEDIAPHON - A TEXT-TO-SPEECH INTERFACE TO THE FREE WIKIPEDIA ENCYCLOPEDIA FOR CELL PHONES AND MP3-PLAYERS  

Microsoft Academic Search

This paper presents an approach to generate audio based learning material dynamically from web pages for m-Learning and ubiquitous access. It introduces Pediaphon (an audio interface to the free Wikipedia online encyclopedia) as an example application for microlearning. The effective generation and the deployment of the audio data to the user via podcast or progressive download (pseudo streaming) are covered.

Andreas Bischoff

35

Using TTS Voices to Develop Audio Materials for Listening Comprehension: A Digital Approach  

ERIC Educational Resources Information Center

This paper reports a series of experiments with text-to-speech (TTS) voices. These experiments have been conducted to develop audio materials for listening comprehension as an alternative technology to traditionally used audio equipment like the compact cassette. The new generation of TTS voices based on unit selection synthesis provides…

Sha, Guoquan

2010-01-01

36

Audio 2008: Audio Fixation  

ERIC Educational Resources Information Center

Take a look around the bus or subway and see just how many people are bumping along to an iPod or an MP3 player. What they are listening to is their secret, but the many signature earbuds in sight should give one a real sense of just how pervasive digital audio has become. This article describes how that popularity is mirrored in library audio

Kaye, Alan L.

2008-01-01

37

Audio Mining  

NSDL National Science Digital Library

Occasionally referred to as audio indexing, audio mining is a computerized task involving the processing of an audio file, extracting the dialog and creating a textual transcript, and searching the transcript for certain words or phrases. Considering the amount of audio content on the Internet and other sources, it is clear that audio mining is a growing technology.To get an idea of what audio mining is and how it can be used, people can read this article from the Cutter Consortium (1). It lists six broad areas that can benefit from using the technology and briefly discusses each one. A more detailed introduction is offered on the Leavitt Communications Web site (2). This article delves into how audio mining works by giving a basic technical understanding of the process. A new method of searching an audio file, dubbed the "phonetic search engine," is compared to traditional methods in this white paper (3). A publication from the Compaq Cambridge Research Laboratory (4) discusses ways of collecting and analyzing information from an audio file. It also mentions SpeechBot, a Web-based tool for multimedia retrieval. Several papers can be downloaded from the home page of a research project studying the National Gallery of the Spoken Word (5). The repository is comprised of massive historical audio content, and the team at the University of Colorado is investigating phrase recognition to index the data. Have you ever had a tune stuck in your head, but not known the name of the artist or song title? The Musical Audio-Mining project (6) is working on ways to search for information about a song simply by humming part of it. Audio mining can also be used in the War on Terrorism, as is described in this article of Federal Computer Week (7). Massive amounts of recorded phone conversations are intercepted by the government each day, and audio mining would be an efficient way to sort through irrelevant material and catch suspicious activity. The World Wide Web Consortium released this draft of the Voice Extensible Markup Language (8), which could have applications for the audio mining community.

Leske, Cavin.

2002-01-01

38

Using Audio  

Microsoft Academic Search

\\u000a This chapter covers the following topics:\\u000a \\u000a \\u000a \\u000a \\u000a  \\u000a \\u000a How to load sound files\\u000a \\u000a \\u000a \\u000a  \\u000a \\u000a How to control audio behavior\\u000a \\u000a \\u000a \\u000a  \\u000a \\u000a How to read and display audio ID3 information\\u000a \\u000a \\u000a \\u000a  \\u000a \\u000a How to display the sound spectrum\\u000a \\u000a \\u000a \\u000a  \\u000a \\u000a How to control sound volume and panning\\u000a \\u000a \\u000a \\u000a The value of sound is subtle and undervalued. Often, it makes the difference between a good site and

Sean McSharry

39

Detecting double compression of audio signal  

NASA Astrophysics Data System (ADS)

MP3 is the most popular audio format nowadays in our daily life, for example music downloaded from the Internet and file saved in the digital recorder are often in MP3 format. However, low bitrate MP3s are often transcoded to high bitrate since high bitrate ones are of high commercial value. Also audio recording in digital recorder can be doctored easily by pervasive audio editing software. This paper presents two methods for the detection of double MP3 compression. The methods are essential for finding out fake-quality MP3 and audio forensics. The proposed methods use support vector machine classifiers with feature vectors formed by the distributions of the first digits of the quantized MDCT (modified discrete cosine transform) coefficients. Extensive experiments demonstrate the effectiveness of the proposed methods. To the best of our knowledge, this piece of work is the first one to detect double compression of audio signal.

Yang, Rui; Shi, Yun Q.; Huang, Jiwu

2010-02-01

40

RECENT ADVANCES IN MULTILINGUAL TEXT-TO-SPEECH SYNTHESIS  

Microsoft Academic Search

this paper we will discuss recent advances in multilingualtext-to-speech (TTS) synthesis research atAT&T Bell Laboratories. The TTS system developedat AT&T Bell Laboratories generates syntheticspeech by concatenating segments of natural speech.The architecture of the system is designed as a modularpipeline where each module handles one particularstep in the process of converting text into speech. Besidesconceptual and computational advantages, themodular structure has

Bernd M; Juergen Schroeter; Jan van Santen; Richard Sproat; Joseph Olive

1996-01-01

41

Implementation of Three Text to Speech Systems for Kurdish Language  

NASA Astrophysics Data System (ADS)

Nowadays, concatenative method is used in most modern TTS systems to produce artificial speech. The most important challenge in this method is choosing appropriate unit for creating database. This unit must warranty smoothness and high quality speech, and also, creating database for it must reasonable and inexpensive. For example, syllable, phoneme, allophone, and, diphone are appropriate units for all-purpose systems. In this paper, we implemented three synthesis systems for Kurdish language based on syllable, allophone, and diphone and compare their quality using subjective testing.

Bahrampour, Anvar; Barkhoda, Wafa; Azami, Bahram Zahir

42

Steganalysis for Audio Data.  

National Technical Information Service (NTIS)

The Audio WET system is a web-based evaluation system which provides the user the functionality of different watermarking algorithms (embedding and detecting) with a large database of audio signals (test material). The system also provides both single and...

J. Dittmann

2006-01-01

43

MPEG digital audio coding  

Microsoft Academic Search

The Moving Pictures Expert Group (MPEG) within the International Organization of Standardization (ISO) has developed a series of audio-visual standards known as MFEG-1 and MPEG-2. These audio-coding standards are the first international standards in the field of high-quality digital audio compression. MPEG-1 covers coding of stereophonic audio signals at high sampling rates aiming at transparent quality, whereas MPEG-2 also offers

P. Noll

1997-01-01

44

Streaming Audio Recorder  

NSDL National Science Digital Library

The Streaming Audio Recorder application allows users to record any type of streaming audio via their computers' speakers or microphone. It's a simple way to record audio from sites such as Grooveshark, YouTube, BBC, and others. The program is compatible with computers running Windows 2000 and newer.

2012-11-02

45

Automatic Audio Content Analysis  

Microsoft Academic Search

This paper describes the theoretic framework and applications of automatic audio content analysis. After explaining the basic properties of audio analysis, we present a toolbox being the basis for the development of audio analysis algorithms. We also describe new applications which can be developed using the toolset, among them music indexing and retrieval as well as violence detection in the

Silvia Pfeiffer; Stephan Fischer; Wolfgang Effelsberg

1996-01-01

46

Audio-visual affective expression recognition  

NASA Astrophysics Data System (ADS)

Automatic affective expression recognition has attracted more and more attention of researchers from different disciplines, which will significantly contribute to a new paradigm for human computer interaction (affect-sensitive interfaces, socially intelligent environments) and advance the research in the affect-related fields including psychology, psychiatry, and education. Multimodal information integration is a process that enables human to assess affective states robustly and flexibly. In order to understand the richness and subtleness of human emotion behavior, the computer should be able to integrate information from multiple sensors. We introduce in this paper our efforts toward machine understanding of audio-visual affective behavior, based on both deliberate and spontaneous displays. Some promising methods are presented to integrate information from both audio and visual modalities. Our experiments show the advantage of audio-visual fusion in affective expression recognition over audio-only or visual-only approaches.

Huang, Thomas S.; Zeng, Zhihong

2007-11-01

47

Comparison of audio and video data sources for quantitative analysis of echocardiographic Doppler velocity profiles  

Microsoft Academic Search

Computer based, off-line quantitative Doppler velocity profile image analysis can utilize either the audio or video formats of the echocardiography machine. The audio and video data are each corrupted, in some way, relative to the Doppler information from which they originate. To compare the suitability of each source for quantitative model-based image processing analysis, audio and video transmitral Doppler data

Andrew F. Hall; S. J. Kovacs

1994-01-01

48

Robust AVS Audio Watermarking  

NASA Astrophysics Data System (ADS)

Part III of AVS(China Audio and Video Coding Standard) is the first standard for Hi-Fi audio proposed in China and is becoming more popular in some IT industries. For MP3 audio, some efforts have been made to solve the problems such as copyright pirating and malicious modifications by the way of watermarking. But till now little efforts have been made to solve the same problems for AVS audio. In this paper, we present a novel robust watermarking algorithm which can protect the AVS audio from the above problems. The watermark is embedded into the AVS compressed bit stream. At the extracting end, the watermark bits can be extracted from the compressed bit stream directly without any computation. This algorithm achieves robustness to decoding/recoding attacks, and low complexity of both embedding and extracting while preserves the quality of the audio signals.

Wang, Yong; Huang, Jiwu

49

Topic in Depth - Audio Mining  

NSDL National Science Digital Library

Occasionally referred to as audio indexing, audio mining is a computerized task involving the processing of an audio file, extracting the dialog and creating a textual transcript, and searching the transcript for certain words or phrases. Considering the amount of audio content on the Internet and other sources, it is clear that audio mining is a growing technology of growing importance.

2010-09-15

50

Processing audio data  

US Patent & Trademark Office Database

An exemplary embodiment is a method of processing audio data comprising: characterising an audio data representative of a recorded sound scene into a set of sound sources occupying positions within a time and space reference frame; analysing the sound sources; and generating a modified audio data representing sound captured from at least one virtual microphone configured for moving about the recorded sound scene, wherein the virtual microphone is controlled in accordance with a result of the analysis of said audio data, to conduct a virtual tour of the recorded sound scene.

2011-01-25

51

Perceptually Based Audio Coding  

NASA Astrophysics Data System (ADS)

High-quality audio is a concept that is not exactly defined and not always properly understood. To some, it refers directly to the physical similarity between a real sound field and its electroacoustical reproduction. In this viewpoint, acoustical knowledge and electronic technology are the only limiting factors preventing audio quality from being perfect. To others, however, audio quality refers to the audible similarity between a real life sound event and an electronic reproduction. Given this viewpoint, the human auditory system with all its limitations becomes an essential factor determining audio quality.

Houtsma, Adrianus J. M.

52

Robust audio hashing for audio authentication watermarking  

NASA Astrophysics Data System (ADS)

Current systems and protocols based on cryptographic methods for integrity and authenticity verification of media data do not distinguish between legitimate signal transformation and malicious tampering that manipulates the content. Furthermore, they usually provide no localization or assessment of the relevance of such manipulations with respect to human perception or semantics. We present an algorithm for a robust message authentication code in the context of content fragile authentication watermarking to verify the integrity of audio recodings by means of robust audio fingerprinting. Experimental results show that the proposed algorithm provides both a high level of distinction between perceptually different audio data and a high robustness against signal transformations that do not change the perceived information. Furthermore, it is well suited for the integration in a content-based authentication watermarking system.

Zmudzinski, Sascha; Steinebach, Martin

2008-03-01

53

The Audio Interactive Tutor.  

ERIC Educational Resources Information Center

Describes The Audio Interactive Tutor (TAIT), an interactive audio/oral computer-assisted study device. TAIT's output consists of explanations and examples along with commands and questions requiring responses from the user. It uses speech recognition to determine the responses made by the user and constructs an evolving model of what the user…

Waters, Richard C.

1995-01-01

54

Audio detection algorithms  

NASA Astrophysics Data System (ADS)

Audio information concerning targets generally includes direction, frequencies, and energy levels. One use of audio cueing is to use direction information to help determine where more sensitive visual direction and acquisition sensors should be directed. Generally, use of audio cueing will shorten times required for visual detection, although there could be circumstances where the audio information is misleading and degrades visual performance. Audio signatures can also be useful for helping classify the emanating platform, as well as to provide estimates of its velocity. The Janus combat simulation is the premier high resolution model used by the Army and other agencies to conduct research. This model has a visual detection model which essentially incorporates algorithms as described by Hartman(1985). The model in its current form does not have any sound cueing capability. This report is part of a research effort to investigate the utility of developing such a capability.

Neta, B.; Mansager, B.

1992-08-01

55

RealAudio  

NSDL National Science Digital Library

The RealAudio free software player allows users with audio capability the ability to listen to audio files "on demand." The files are played-back instantaneously while they are downloading, rather than waiting for the entire file to be downloaded before the user can begin listening. As audio files are usually large, there is a user-friendly advantage in this technology. RealAudio allows for emulation of "radio" broadcasts over the Internet. Download the free player for Windows or Macintosh and connect to sites such as ABC and NPR to hear news broadcasts and interviews. The quality of the sound can be weak, but as the technology improves so will the sound.

1998-01-01

56

3D Audio System  

NASA Technical Reports Server (NTRS)

Ames Research Center research into virtual reality led to the development of the Convolvotron, a high speed digital audio processing system that delivers three-dimensional sound over headphones. It consists of a two-card set designed for use with a personal computer. The Convolvotron's primary application is presentation of 3D audio signals over headphones. Four independent sound sources are filtered with large time-varying filters that compensate for motion. The perceived location of the sound remains constant. Possible applications are in air traffic control towers or airplane cockpits, hearing and perception research and virtual reality development.

1992-01-01

57

Signal Processing for Audio HCI  

Microsoft Academic Search

\\u000a This chapter reviews recent advances in computer audio processing from the viewpoint of improving the human-computer interface.\\u000a Microphone arrays are described as basic tools for untethered audio acquisition, and principles for the synthesis of realistic\\u000a virtual audio are outlined. The influence of room acoustics on audio acquisition and production is also considered. The chapter\\u000a finishes with a review of several

Dmitry N. Zotkin; Ramani Duraiswami

58

AUDIO-CASI  

PubMed Central

This article reviews a multimedia application in the area of survey measurement research: adding audio capabilities to a computer-assisted interviewing system. Hardware and software issues are discussed, and potential hardware devices that operate from DOS platforms are reviewed. Three types of hardware devices are considered: PCMCIA devices, parallel port attachments, and laptops with built-in sound.

Cooley, Philip C.; Turner, Charles F.; O'Reilly, James M.; Allen, Danny R.; Hamill, David N.; Paddock, Richard E.

2011-01-01

59

AHA: Audio HTML Access  

Microsoft Academic Search

This report discusses the “AHA” system for presenting HTML in audio for blind users and others who wish to access the WWW non-visually. AHA is a framework and set of suggestions for HTML presentation based on an initial experiment. Further experimentation and further revisions will be performed with the system.

Frankie James

1997-01-01

60

Efficient audio signal processing for embedded systems  

NASA Astrophysics Data System (ADS)

As mobile platforms continue to pack on more computational power, electronics manufacturers start to differentiate their products by enhancing the audio features. However, consumers also demand smaller devices that could operate for longer time, hence imposing design constraints. In this research, we investigate two design strategies that would allow us to efficiently process audio signals on embedded systems such as mobile phones and portable electronics. In the first strategy, we exploit properties of the human auditory system to process audio signals. We designed a sound enhancement algorithm to make piezoelectric loudspeakers sound ”richer" and "fuller." Piezoelectric speakers have a small form factor but exhibit poor response in the low-frequency region. In the algorithm, we combine psychoacoustic bass extension and dynamic range compression to improve the perceived bass coming out from the tiny speakers. We also developed an audio energy reduction algorithm for loudspeaker power management. The perceptually transparent algorithm extends the battery life of mobile devices and prevents thermal damage in speakers. This method is similar to audio compression algorithms, which encode audio signals in such a ways that the compression artifacts are not easily perceivable. Instead of reducing the storage space, however, we suppress the audio contents that are below the hearing threshold, therefore reducing the signal energy. In the second strategy, we use low-power analog circuits to process the signal before digitizing it. We designed an analog front-end for sound detection and implemented it on a field programmable analog array (FPAA). The system is an example of an analog-to-information converter. The sound classifier front-end can be used in a wide range of applications because programmable floating-gate transistors are employed to store classifier weights. Moreover, we incorporated a feature selection algorithm to simplify the analog front-end. A machine learning algorithm AdaBoost is used to select the most relevant features for a particular sound detection application. In this classifier architecture, we combine simple "base" analog classifiers to form a strong one. We also designed the circuits to implement the AdaBoost-based analog classifier.

Chiu, Leung Kin

61

Audio?visual aids  

Microsoft Academic Search

AUDIO?VISUAL MATERIALS IN TEACHING, 16 mm. 13 1\\/2 minutes. Sound. Color, $150.00, Black and White, $75.00; and NEW DIMENSIONS THROUGH TEACHING FILMS, 16 mm. 27 minutes. Sound. Color, $180.00. Available through Coronet Instructional Films, Coronet Building, Chicago, Illinois 60601.ART: WHAT IS IT: WHY IS IT?, No. 47561. 16 mm. go minutes. Sound. Color. Cost $360. Available through Encyclopaedia Britannica Films,

Robertson Strawn; Ruth Hankowsky; Ronad J. Koperski; Mary Louise Gehring; Dora Heintz; Ruby Cloys Krider; Jimmie D. Trent

1967-01-01

62

Audio Coding and Its Applications  

NASA Astrophysics Data System (ADS)

This paper presents an overview of the MPEG audio standardization process as footsteps of the development of audio coding technologies in the last 25 years, with some technologies that the author has devoted a significant part of his career to as a research engineer. First, the development of audio coding technologies is overviewed by following the history of MPEG/Audio standardization. The MPEG-2/-4 AAC encoding process is then explained, which is now recognized as the standard framework for audio-data compression,followed by some technologies that the author has been deeply involved in as a research engineer. Finally, MPEG/Audio Standards are summarized from the viewpoint of redundancy reduction, and an outlook for the future is presented.

Sugiyama, Akihiko

63

Content analysis for audio classification and segmentation  

Microsoft Academic Search

In this paper, we present our study of audio content analysis for classification and segmentation, in which an audio stream is segmented according to audio type or speaker identity. We propose a robust approach that is capable of classifying and segmenting an audio stream into speech, music, environment sound, and silence. Audio classification is processed in two steps, which makes

Lie Lu; Hong-jiang Zhang; Hao Jiang

2002-01-01

64

Robust and efficient digital audio watermarking using audio content analysis  

NASA Astrophysics Data System (ADS)

Digital audio watermarking embeds inaudible information into digital audio data for the purposes of copyright protection, ownership verification, covert communication, and/or auxiliary data carrying. In this paper, we first describe the desirable characteristics of digital audio watermarks. Previous work on audio watermarking, which has primarily focused on the inaudibility of the embedded watermark and its robustness against attacks such as compression and noise, is then reviewed. In this research, special attention is paid to the synchronization attack caused by casual audio editing or malicious random cropping, which is a low-cost yet effective attack to watermarking algorithms developed before. A digital audio watermarking scheme of low complexity is proposed in this research as an effective way to deter users from misusing or illegally distributing audio data. The proposed scheme is based on audio content analysis using the wavelet filterbank while the watermark is embedded in the Fourier transform domain. A blind watermark detection technique is developed to identify the embedded watermark under various types of attacks.

Wu, Chung-Ping; Su, Po-Chyi; Kuo, C.-C. Jay

2000-05-01

65

Phonetic Searching Of Digital Audio  

NSDL National Science Digital Library

A new method of searching an audio file, dubbed the "phonetic search engine," is compared to traditional methods in this white paper. The 10-page pdf document has images and graphs to illustrate the process and results. Topics covered include audio searching techniques, implementation of new search methods, as well as current and future applications of the technology.

Cardillo, Peter S.; Clements, Mark; Miller, Michael

2007-12-10

66

MP3 robust Audio Watermarking  

Microsoft Academic Search

This paper presents methods for audio watermarking in the frequency domain. It will consider desired properties and possible applications of watermark ing algorithms. Special attention is given to statistical methods relying on the so-called patch work techniques. It will present a solution of robust watermarking of audio data and reflect the security properties of the tech- nique. Experimental results show

Michael Arnold; Sebastian Kanka

1999-01-01

67

Haptics in audio described movies  

Microsoft Academic Search

Similar to closed captioning for people who cannot hear, Audio Description (AD) in movies correspond to a secondary audio track describing events that are otherwise visually dominant. To aid people who are visually impaired or blind in understanding the happenings of a movie, the narrator describes the emotions, expressions, and actions presented in a movie scene, among many other details.

Lakshmie Narayan Viswanathan; Troy McDaniel; Sreekar Krishna; Sethuraman Panchanathan

2010-01-01

68

Metrological digital audio reconstruction  

DOEpatents

Audio information stored in the undulations of grooves in a medium such as a phonograph record may be reconstructed, with little or no contact, by measuring the groove shape using precision metrology methods coupled with digital image processing and numerical analysis. The effects of damage, wear, and contamination may be compensated, in many cases, through image processing and analysis methods. The speed and data handling capacity of available computing hardware make this approach practical. Two examples used a general purpose optical metrology system to study a 50 year old 78 r.p.m. phonograph record and a commercial confocal scanning probe to study a 1920's celluloid Edison cylinder. Comparisons are presented with stylus playback of the samples and with a digitally re-mastered version of an original magnetic recording. There is also a more extensive implementation of this approach, with dedicated hardware and software.

Fadeyev; Vitaliy (Berkeley, CA), Haber; Carl (Berkeley, CA)

2004-02-19

69

UNICEF Video/Audio  

NSDL National Science Digital Library

UNICEF is known throughout the world for their focus on the health, education, equality and protection of children. They produce a number of helpful research reports and policy briefs, and as visitors to this site will find out, a good deal of audio and visual material in the form of podcasts, video news reports, and radio programs. Visitors to the UNICEF Radio area will find a wide range of radio reports on topics such as Nigeria's efforts to contain outbreaks of avian influenza and the effects of floods in Mozambique on children. Visitors interested in podcasts will be impressed with the offerings here, as they include over one hundred total archived programs, and visitors can also sign up to receive each new addition to this collection.

70

Machine Vision  

NSDL National Science Digital Library

An overview of a generic image-based machine vision system is provided on this Web site (1). The tutorial describes the main components of such a system, how its accuracy is measured, and what scientific and industrial applications benefit from machine vision. A more technical perspective of machine vision technology is given in an online publication of the Automated Imaging Association (2). Monthly feature articles discuss breaking issues related to machine vision, and several technical papers can be downloaded, which are sorted into categories such as three dimensional imaging and nanotechnology. Researchers from the MIT's Artificial Intelligence Laboratory (3) are investigating how to enable a computer to interpret visual and audio signals from its human user. By using machine perception systems to track the user's gaze, for example, the computer could ascertain the focus of the user's attention, thereby facilitating interaction between the human and the computer. The project's homepage includes numerous research papers, as well as video demonstrations of some of its systems. Machine vision is also finding its way into vehicles. A March 2003 news article (4) highlights a field test in Michigan of a collision avoidance system that uses, among other things, machine vision to warn drivers that they are approaching a slower or stopped object too quickly. The 3D Computer Vision Group at Carnegie Mellon University (5) is involved in several projects, including three dimensional object recognition and humanoid robot vision. Many of the group's recent publications are available for download. NASA's Mars Exploration Rovers, the second of which was launched in July 2003, have vision systems that will let them safely navigate rough terrain. These systems are described in this conference paper (6), including specifics of the stereo vision algorithm and insights into future missions. A new implementation of machine vision comes from a former researcher from Cambridge University. His shape recognition system, which is detailed in this news article (7), is reportedly much more related to human visual processes than existing techniques. For additional developments related to this evolving technology, Machine Vision News (8) has information about worldwide research and new applications of machine vision systems.

Leske, Cavin.

71

Wavelet-Based Audio Embedding & Audio/Video Compression.  

National Technical Information Service (NTIS)

With the decline in military spending, the United States relies heavily on state side support. Communications has never been more important. High-quality audio and video capabilities are a must. Watermarking, traditionally used for copyright protection, i...

M. J. Mendenhall

2001-01-01

72

A Tutorial on MPEG\\/Audio Compression  

Microsoft Academic Search

ABSTRACT This tutorial covers the theory behind MPEG\\/audio compression This algorithm was developed by the Motion Picture Experts Group (MPEG), as an International Organization for Standardization (ISO) standard for the high fidelity compression of digital audio The MPEG\\/audio compression standard is one part of a multiple part standard that addresses the compression of video - 2), the compression of audio

Davis Pan

1995-01-01

73

Audio stream analysis for environmental sound classification  

Microsoft Academic Search

We present in this paper a framework for audio concept identification based on audio stream analysis and binary classifiers encapsulation. The system consists of three stages. The first stage is called the pre-processing level audio, where audio stream is segmented and silence segments are detected. In the second stage, speech, music and environmental sounds are automatically divided and further classified.

Issam Feki; Anis Ben Ammar; Adel M. Alimi

2011-01-01

74

A centralized audio presentation manager  

SciTech Connect

The centralized audio presentation manager addresses the problems which occur when multiple programs running simultaneously attempt to use the audio output of a computer system. Time dependence of sound means that certain auditory messages must be scheduled simultaneously, which can lead to perceptual problems due to psychoacoustic phenomena. Furthermore, the combination of speech and nonspeech audio is examined; each presents its own problems of perceptibility in an acoustic environment composed of multiple auditory streams. The centralized audio presentation manager receives abstract parameterized message requests from the currently running programs, and attempts to create and present a sonic representation in the most perceptible manner through the use of a theoretically and empirically designed rule set.

Papp, A.L. III; Blattner, M.M.

1994-05-16

75

jReality: a java library for real-time interactive 3D graphics and audio  

Microsoft Academic Search

We introduce jReality, a Java library for creating real-time interactive audiovisual applications with three-dimensional computer graphics and spatialized audio. Applications written for jReality will run unchanged on software and hardware platforms ranging from desktop machines with a single screen and stereo speakers to immersive virtual environments with motion tracking, multiple screens with 3D stereo projection, and multi-channel audio.

Steffen Weißmann; Charles Gunn; Peter Brinkmann; Tim Hoffmann; Ulrich Pinkall

2009-01-01

76

Preparation of sound base for a text-to-speech synthesis system  

NASA Astrophysics Data System (ADS)

We are giving several recommendations for the choice of parameters of the sound fragments in this report. The sound fragments are components of the sound base, used in Russian speech synthesis system by a text. It isn't the secret that quality of concatenation synthesis in many respects is defined at the stage of a speaker choice and preparation of base of speaker's voice samples. Formulated recommendations are received on the basis of the statistic analysis of big amount of various types of texts and concern both separate sound fragments and their groups. Parameters of sounds were taken with the help of the automatic linguistic processor including phonetic and prosodic transcriptors. The duration, intensity and main pitch frequency of sounds in various contexts and intonational contours were analyzed. The sound base produced according to the worked out recommendations, allows to make better intelligibility and naturalness of synthetic speech due to minimization of changes of speaker's voice samples.

Degtyarev, Vladimir M.; Gusev, Mikhail N.

2005-04-01

77

Blind audio watermark synchronization by passive audio fingerprinting  

NASA Astrophysics Data System (ADS)

Synchronization is still one of the most important issues in digital watermarking. Many attacks do not remove the watermark from the cover, but only disable the synchronization between the watermark and the detector. Most watermarking algorithms feature some synchronization strategy, but especially in audio watermarking this may not be sufficient to fight de-synchronization attacks: As the watermark is embedded over a given length of audio data, a good synchronization at the starting point of the retrieval process may be lost during retrieval. An example for this is time stretching, where the effects of playback speed modification sums up during retrieval. We introduce a novel synchronization approach applying passive audio fingerprinting to synchronize each watermarking bit individually. Storage of the fingerprint values is not necessary in our approach, improving the usability compared to existing solutions in this area.

Steinebach, Martin; Zmudzinski, Sascha

2007-02-01

78

FLAC: Free Lossless Audio Codec  

NSDL National Science Digital Library

The Free Lossless Audio Codec (FLAC) is a method of compressing audio data. Whereas the widely known MP3 format sacrifices sound quality for a smaller compressed size, FLAC does not lose information in the encoding process. FLAC is an open source software project, and its homepage contains downloadable encoding utilities for most common operating systems. An excellent overview of the FLAC format is given, which explains the underlying architecture and features of the codec. This section also provides links to background information and research papers upon which the development of FLAC was based. Thorough documentation is included online.

79

Kenneth S. Goldstein Audio Recordings  

NSDL National Science Digital Library

This remarkable collection consists of over 850 audio reels recorded primarily by Dr. Kenneth S. Goldstein. He was a folklorist, record producer, and teacher who happened to also find time to serve as chairman of the department of folklore and folklife at the University of Pennsylvania. These audio tapes include interviews with musicians and storytellers, recitations of folktales from Newfoundland and Labrador, Pennsylvania, and Scotland. First-time visitors might do well to look over the English Language Folktale reels and then move on to perform their own detailed search across the entire archive. Visitors can also elect to receive updates on the collection via their RSS feed.

80

Advances in audio source seperation and multisource audio content retrieval  

NASA Astrophysics Data System (ADS)

Audio source separation aims to extract the signals of individual sound sources from a given recording. In this paper, we review three recent advances which improve the robustness of source separation in real-world challenging scenarios and enable its use for multisource content retrieval tasks, such as automatic speech recognition (ASR) or acoustic event detection (AED) in noisy environments. We present a Flexible Audio Source Separation Toolkit (FASST) and discuss its advantages compared to earlier approaches such as independent component analysis (ICA) and sparse component analysis (SCA). We explain how cues as diverse as harmonicity, spectral envelope, temporal fine structure or spatial location can be jointly exploited by this toolkit. We subsequently present the uncertainty decoding (UD) framework for the integration of audio source separation and audio content retrieval. We show how the uncertainty about the separated source signals can be accurately estimated and propagated to the features. Finally, we explain how this uncertainty can be efficiently exploited by a classifier, both at the training and the decoding stage. We illustrate the resulting performance improvements in terms of speech separation quality and speaker recognition accuracy.

Vincent, Emmanuel

2012-05-01

81

Emphasizing the Audio in the Audio-Lingual Approach.  

National Technical Information Service (NTIS)

The neglect of listening comprehension in the audio-lingual approach to the teaching of foreign languages is pointed out. The need to develop listening comprehension, not only as a foundation for speaking, but as a skill in its own right, is stressed. The...

G. Newmark E. Diller

1964-01-01

82

50 CFR 27.72 - Audio equipment.  

Code of Federal Regulations, 2013 CFR

...OF THE INTERIOR (CONTINUED) THE NATIONAL WILDLIFE REFUGE SYSTEM PROHIBITED ACTS Disturbing Violations: Filming, Photography, and Light and Sound Equipment § 27.72 Audio equipment. The operation or use of audio devices including...

2013-10-01

83

Versatile Audio Mixing and Distribution Ammplifiers.  

National Technical Information Service (NTIS)

Past research projects in the Behavioral Sciences Department leaned heavily upon the use of a complex, audio equipment array interposed between human subjects and researchers. Requirements to distribute audio information from a single source to several de...

J. A. Diachenko B. Valjoen

1970-01-01

84

On the comparison of audio fingerprints for extracting quality parameters of compressed audio  

Microsoft Academic Search

Audio fingerprints can be seen as hashes of the perceptual content of an audio excerpt. Applications include linking metadata to unlabeled audio, watermark support, and broadcast monitoring. Existing systems identify a song by comparing its fingerprint to pre-computed fingerprints in a database. Small changes of the audio induce small differences in the fingerprint. The song is identified if these fingerprint

P. J. O. Doets; M. Menor Gisbert; R. L. Lagendijk

85

The Audio-Visual Man.  

ERIC Educational Resources Information Center

A series of twelve essays discuss the use of audiovisuals in religious education. The essays are divided into three sections: one which draws on the ideas of Marshall McLuhan and other educators to explore the newest ideas about audiovisual language and faith, one that describes how to learn and use the new language of audio and visual images, and…

Babin, Pierre, Ed.

86

Lossless Adaptive Digital Audio Steganography  

Microsoft Academic Search

This paper presents a lossless adaptive digital audio steganographic technique based on reversible two and higher dimensional integer transform. The adaptive technique is used to choose the best blocks for embedding perceptually inaudible stego information, and to select the best block sizes to maximize the number of blocks\\/capacity. The stego information is embedded in the integer domain by bit manipulation.

Sos S. Agaian; David Akopian; Okan Caglayan; S. A. D'Souza

2005-01-01

87

A Simple Audio Conductivity Device.  

ERIC Educational Resources Information Center

Describes a simple audio conductivity device built to address the problem of the lack of sensitivity needed to measure small differences in conductivity in crude conductivity devices. Uses a 9-V battery as a power supply and allows the relative resistance differences between substances to be detected by the frequency of its audible tones. Presents…

Berenato, Gregory; Maynard, David F.

1997-01-01

88

Performance of an Audio Perceptron.  

National Technical Information Service (NTIS)

Perceptrons are a class of simple adaptive pattern-recognition devices built of crude model neurons. In the work a perceptron is used to recognize patterns generated by an audio preprocessor. The preprocessor is modeled on the cochlea and cochlear ganglio...

M. G. Scattergood

1971-01-01

89

Audio watermarking for live performance  

Microsoft Academic Search

Audio watermarking has been used mainly for digitally stored content. Using real-time watermark embedding, its coverage can be extended to live broadcasts and live performances. In general, a conventional embedding algorithm receives a host signal (HS) and outputs the summation of the HS and a watermark signal (WS). However, when applied to real-time embedding, there are two problems: (1) delay

Ryuki Tachibana

2003-01-01

90

Radioactive Decay: Audio Data Collection  

ERIC Educational Resources Information Center

Many phenomena generate interesting audible time series. This data can be collected and processed using audio software. The free software package "Audacity" is used to demonstrate the process by recording, processing, and extracting click times from an inexpensive radiation detector. The high quality of the data is demonstrated with a simple…

Struthers, Allan

2009-01-01

91

Digital audio. Recursive digital filtering for high quality audio signals  

NASA Astrophysics Data System (ADS)

A high speed, low cost digital signal processor was designed and built suitable for real-time operations on high quality sound signals. The purpose of the work described here is to examine its application to audio filters of a kind currently found in equipment such as mixing desks. The methods by which digital audio filters may be designed are reviewed and a detailed study is presented of the many ways in which the design may be realized and implemented. The Report describes two structures in detail which were identified as providing high performance with ease of implementation. The effects of coefficient quantization and round-off noise are quantified by computer simulation of the hardware structure; and by these means, a specification for the required world-length is determined. An example is given of a four filter cascade for audio equalization with a suitable implementation in programmable hardware. The filters are controlled by flexible commands issued on an internationally standarized communications bus.

McNally, G. W.; Eng, C.

1981-12-01

92

Quantitative characterisation of audio data by ordinal symbolic dynamics  

NASA Astrophysics Data System (ADS)

Ordinal symbolic dynamics has developed into a valuable method to describe complex systems. Recently, using the concept of transcripts, the coupling behaviour of systems was assessed, combining the properties of the symmetric group with information theoretic ideas. In this contribution, methods from the field of ordinal symbolic dynamics are applied to the characterisation of audio data. Coupling complexity between frequency bands of solo violin music, as a fingerprint of the instrument, is used for classification purposes within a support vector machine scheme. Our results suggest that coupling complexity is able to capture essential characteristics, sufficient to distinguish among different violins.

Aschenbrenner, T.; Monetti, R.; Amigó, J. M.; Bunk, W.

2013-06-01

93

Working with HTML5 Audio and Video  

Microsoft Academic Search

\\u000a In this chapter, we’ll explore what you can do with two important HTML5 elements—audio and video— and we’ll show you how they can be used to create compelling applications. The audio and video elements add new media options\\u000a to HTML5 applications that allow you to use audio and video without plugins while providing a common, integrated, and scriptable\\u000a API.

Peter Lubbers; Brian Albers; Frank Salim

94

Classification of audio signals using AANN and GMM  

Microsoft Academic Search

Today, digital audio applications are part of our everyday lives. Audio classification can provide powerful tools for content management. If an audio clip automatically can be classified it can be stored in an organised database, which can improve the management of audio dramatically. In this paper, we propose effective algorithms to automatically classify audio clips into one of six classes:

P. Dhanalakshmi; S. Palanivel; Vennila Ramalingam

2011-01-01

95

Jet Audio 5.14 Basic  

NSDL National Science Digital Library

For persons looking for a handy and powerful way to play numerous types of audio or video files, Jet Audio 5.14 Basic will be a welcome addition to their computer. With this latest version, users can broadcast over the internet, utilize the built-in equalizer, and control the speed of recordings, along with sixteen other features. Additionally, users can manipulate the appearance (or "skin") of Jet Audio, or create their own skin as well. Jet Audio 5.14 Basic is compatible with all systems running Windows 98 and higher.

96

Three-Dimensional Audio Client Library  

NASA Technical Reports Server (NTRS)

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.

2005-01-01

97

Authenticity examination of compressed audio recordings using detection of multiple compression and encoders' identification.  

PubMed

Since the appearance of digital audio recordings, audio authentication has been becoming increasingly difficult. The currently available technologies and free editing software allow a forger to cut or paste any single word without audible artifacts. Nowadays, the only method referring to digital audio files commonly approved by forensic experts is the ENF criterion. It consists in fluctuation analysis of the mains frequency induced in electronic circuits of recording devices. Therefore, its effectiveness is strictly dependent on the presence of mains signal in the recording, which is a rare occurrence. Recently, much attention has been paid to authenticity analysis of compressed multimedia files and several solutions were proposed for detection of double compression in both digital video and digital audio. This paper addresses the problem of tampering detection in compressed audio files and discusses new methods that can be used for authenticity analysis of digital recordings. Presented approaches consist in evaluation of statistical features extracted from the MDCT coefficients as well as other parameters that may be obtained from compressed audio files. Calculated feature vectors are used for training selected machine learning algorithms. The detection of multiple compression covers up tampering activities as well as identification of traces of montage in digital audio recordings. To enhance the methods' robustness an encoder identification algorithm was developed and applied based on analysis of inherent parameters of compression. The effectiveness of tampering detection algorithms is tested on a predefined large music database consisting of nearly one million of compressed audio files. The influence of compression algorithms' parameters on the classification performance is discussed, based on the results of the current study. PMID:24637036

Korycki, Rafal

2014-05-01

98

Speaker identification based text to audio alignment for an audio retrieval system  

Microsoft Academic Search

We report on an audio retrieval system which lets Internet users efficiently access a large audio database containing recordings of the proceedings of the United States House of Representatives. The audio has been temporally aligned to text transcripts of the proceedings (which are manually generated by the US Government) using a novel method based on speaker identification. Speaker sequence and

Deb Roy; Carl Malamud

1997-01-01

99

A new implementation of the Silicon Audio Player based on an MPEG\\/audio decoder LSI  

Microsoft Academic Search

A new implementation of the Silicon Audio player is presented. It decodes data which has been encoded by the MPEG\\/Audio Layer II algorithm standardized by the ISO (International Standardization Organization). The encoded data is stored in a semiconductor memory card. Decoding is carried out by an MPEG\\/audio decoder chip. Thanks to this dedicated LSI chip, the power consumption of the

Akihiko Sugiyama; Masahiro Iwadare; Takashi Manabe; Nobuhiro Ohdate; Hideto Takano; Osamu Kitabatake; Eiji Hirao

1997-01-01

100

The Silicon Audio an audio-data compression and storage system with a semiconductor memory card  

Microsoft Academic Search

A new audio-data compression and storage system, the Silicon Audio, is presented. It employs the MPEG\\/Audio Layer II algorithm for data compression, which has been standardized by ISO (International Standardization Organization). A semiconductor memory card is equipped with to store the compressed signal. Decoding is carried out by a general purpose digital signal processor and a specially designed gate array

A. Sugiyama; M. Iwadare; N. Ohdate; T. Manabe; H. Takano; O. Kitabatake; E. Hirao

1995-01-01

101

Digital Audio Radio Field Tests  

NASA Technical Reports Server (NTRS)

Radio history continues to be made at the NASA Lewis Research Center with the beginning of phase two of Digital Audio Radio testing conducted by the Consumer Electronic Manufacturers Association (a sector of the Electronic Industries Association and the National Radio Systems Committee) and cosponsored by the Electronic Industries Association and the National Association of Broadcasters. The bulk of the field testing of the four systems should be complete by the end of October 1996, with results available soon thereafter. Lewis hosted phase one of the testing process, which included laboratory testing of seven proposed digital audio radio systems and modes (see the following table). Two of the proposed systems operate in two modes, thus making a total of nine systems for testing. These nine systems are divided into the following types of transmission: in-band on channel (IBOC), in-band adjacent channel (IBAC), and new bands - the L-band (1452 to 1492 MHz) and the S-band (2310 to 2360 MHz).

Hollansworth, James E.

1997-01-01

102

Issues of audio quality for video conferencing  

NASA Astrophysics Data System (ADS)

When choosing a video conferencing system ,it is natural for the potential buyer to look very closely at the quality of the audio. In order to assure a high-quality video conferencing, it is important to have a good picture, but it is vital to have high quality audio. The reason for this is that most of the information transferred in a video conference is actually in the audio channel. Not only is good audio mandatory to the effective exchange of information exchange of information, it has also been found that it can effect the perceived video quality. In this ape several key issues on audio quality are discussed. Since audio delay is among the most vexing problem, each video conferencing system has to make efforts to shorten it. What causes audio delay, how to measure actual delay and how to shorten it is provided in detail. Finally several strategies about system design are presented in order to improve audio quality. All the above are based on the video conferencing systems developed according to H.324 and suitable to video conferencing system implemented all using software.

Han, Qi; Zhou, Jingli; Yu, Shengsheng

1999-01-01

103

Enhancing Manual Scan Registration Using Audio Cues  

NASA Astrophysics Data System (ADS)

Indoor mapping and modelling requires that acquired data be processed by editing, fusing, formatting the data, amongst other operations. Currently the manual interaction the user has with the point cloud (data) while processing it is visual. Visual interaction does have limitations, however. One way of dealing with these limitations is to augment audio in point cloud processing. Audio augmentation entails associating points of interest in the point cloud with audio objects. In coarse scan registration, reverberation, intensity and frequency audio cues were exploited to help the user estimate depth and occupancy of space of points of interest. Depth estimations were made reliably well when intensity and frequency were both used as depth cues. Coarse changes of depth could be estimated in this manner. The depth between surfaces can therefore be estimated with the aid of the audio objects. Sound reflections of an audio object provided reliable information of the object surroundings in some instances. For a point/area of interest in the point cloud, these reflections can be used to determine the unseen events around that point/area of interest. Other processing techniques could benefit from this while other information is estimated using other audio cues like binaural cues and Head Related Transfer Functions. These other cues could be used in position estimations of audio objects to aid in problems such as indoor navigation problems.

Ntsoko, T.; Sithole, G.

2014-04-01

104

Let's Hear It for Audio Mining  

NSDL National Science Digital Library

A detailed introduction is offered on the Leavitt Communications Web site. This article delves into how audio mining works by giving a basic technical understanding of the process. Approaches to audio mining are discussed, as well as how the technology works, performance, languages, and the challenges faced by designers.

Leavitt, Neal

2007-12-11

105

Audio Bombing: Magnetic Cassette Tape Graffiti  

Microsoft Academic Search

Audio Bombing is an alternative form of graffiti that uses magnetic audiotape as its medium. Drawing from hip hop and graffiti culture Audio Bombing starts with a basic cassette tape. Using a tape recorder you can record any information you want on to a cassette (music, poems, philosophy, subversive literature, etc.). After recording you remove the tape and cut out

Mike Fleming; Kang Chang; Kyle Millns

2007-01-01

106

Features for audio and music classification  

Microsoft Academic Search

Four audio feature sets are evaluated in their ability to classify five general audio classes and seven pop- ular music genres. The feature sets include low-level signal properties, mel-frequency spectral coefficients, and two new sets based on perceptual models of hear- ing. The temporal behavior of the features is ana- lyzed and parameterized and these parameters are in- cluded as

Martin F. Mckinney; Jeroen Breebaart

2003-01-01

107

Digital Audio Sampling for Film and Video.  

ERIC Educational Resources Information Center

Digital audio sampling is explained, and some of its implications in digital sound applications are discussed. Digital sound equipment is rapidly replacing analog recording devices as the state-of-the-art in audio technology. The philosophy of digital recording involves doing away with the continuously variable analog waveforms and turning the…

Stanton, Michael J.

108

Application of psychoacoustics to audio signal processing  

Microsoft Academic Search

The application of psychoacoustics to audio signal processing has been successfully applied to bit-rate reduction coding schemes such as MPEG but has not seen significant implementations into other audio applications. Because of the nonlinear aspect of auditory perception, which includes peripheral mechanics and central cognitive abilities, algorithms derived to enhance one aspect of hearing may have a completely unexpected consequence

Brent Edwards

2001-01-01

109

Sound analysis using MPEG compressed audio  

Microsoft Academic Search

There is a huge amount of audio data available that is compressed using the MPEG audio compression standard. Sound analysis is based on the computation of short time feature vectors that describe the instantaneous spectral content of the sound. An interesting possibility is the calculation of features directly from compressed data. Since the bulk of the feature calculation is performed

George Tzanetakis; F. Cook

2000-01-01

110

Realistic audio in immersive video conferencing  

Microsoft Academic Search

With increasing computation power, network bandwidth, and improvements in display and capture technologies, fully immersive conferencing and tele-immersion is becoming ever closer to reality. Outside of video, one of the key components needed is high quality spatialized audio. This paper presents an implementation of a relatively low complexity, simple solution which allows realistic audio spatialization of arbitrary positions in a

Sanjeev Mehrotra; Wei-ge Chen; Zhengyou Zhang; Philip A. Chou

2011-01-01

111

The Subconscious Effect During Audio Monitoring.  

National Technical Information Service (NTIS)

The purpose of the study was to determine if a subconscious detection effect exists in audio monitoring. The vigilance task consisted of the detection of an audio signal masked by thermal noise. Thirty signals over a sixty minute watch were presented to e...

J. C. Bergner

1972-01-01

112

Robust audio watermarking for copyright protection  

NASA Astrophysics Data System (ADS)

A digital audio watermarking scheme of low complexity is proposed in this research as an effective way to deter users from misusing or illegally distributing audio data. Previous work on audio watermarking has primarily focused on the inaudibility of the embedded watermark and its robustness against attacks such as compression and noise. In this research, special attention is paid to the synchronization attack caused by casual audio editing or malicious random cropping, which is a low-cost yet effective attack to watermarking algorithms developed before. The proposed scheme is based on audio content analysis and watermark embedding in the Fourier transform domain. A blind watermark detection technique is developed to identify the embedded watermark under various types of attacks.

Wu, Chung-Ping; Su, Po-Chyi; Kuo, C.-C. J.

1999-11-01

113

Collusion-resistant audio fingerprinting system in the modulated complex lapped transform domain.  

PubMed

Collusion-resistant fingerprinting paradigm seems to be a practical solution to the piracy problem as it allows media owners to detect any unauthorized copy and trace it back to the dishonest users. Despite the billionaire losses in the music industry, most of the collusion-resistant fingerprinting systems are devoted to digital images and very few to audio signals. In this paper, state-of-the-art collusion-resistant fingerprinting ideas are extended to audio signals and the corresponding parameters and operation conditions are proposed. Moreover, in order to carry out fingerprint detection using just a fraction of the pirate audio clip, block-based embedding and its corresponding detector is proposed. Extensive simulations show the robustness of the proposed system against average collusion attack. Moreover, by using an efficient Fast Fourier Transform core and standard computer machines it is shown that the proposed system is suitable for real-world scenarios. PMID:23762455

Garcia-Hernandez, Jose Juan; Feregrino-Uribe, Claudia; Cumplido, Rene

2013-01-01

114

Mean Machines.  

ERIC Educational Resources Information Center

Suggests scales as alternative representations of numerical concepts and operations that can be used as arithmetic-mean machines, adding machines, multiplication machines, and geometric-mean machines. (ASK)

Flores, Alfinio

1998-01-01

115

Cluster: Metals. Course: Machine Shop. Research Project.  

ERIC Educational Resources Information Center

The set of 13 units is designed for use with an instructor in actual machine shop practice and is also keyed to audio visual and textual materials. Each unit contains a series of task packages which: specify prerequisites within the series (minimum is Unit 1); provide a narrative rationale for learning; list both general and specific objectives in…

Sanford - Lee County Schools, NC.

116

Simple Machines  

NSDL National Science Digital Library

Online Simple Machines Assignment OBJECTIVES: Student\\'s will be able to name and describe all seven simple machines. Students will be able to identify simple machines that they use everyday. Example: Clock = Gear INSTRUCTIONS: 1. Click on the Simple Machines Glossary page and familiarize yourself with the seven simple machines. Simple Machines Glossary Page 2. Students are to click on ...

Oldroyd, Mr.

2007-09-26

117

Digital audio for satellite network radio  

NASA Astrophysics Data System (ADS)

The paper presents the time-division-multiplexed (TDM) digital system supplied by Scientific-Atlanta for ABC, CBS, and NBC. The system has a transmission rate of 8.78 Mbps, and can demodulate, decode and demultiplex the data into the desired audio and data channels, supporting data rates equivalent to twenty 15 kHz audio channels at 384 kbps each. Digital transmission is used for data and channel use flexibility, and efficient usage of the satellite transponder. This TDM digital earth terminal configuration provides very high quality audio reception, built-in expansion capability for future services, and relative immunity to terrestrial interference.

McBride, A. L.

1982-04-01

118

Working with audio: integrating personal tape recorders and desktop computers  

Microsoft Academic Search

Audio data is rarely used on desktop computers today, although audio is otherwise widely used for communication tasks. This paper describes early work aimed at creating computer tools that support the ways users may want to work with audio data. User needs for the system were determined by intervieweing people already working with audio data, using existing devices such as

Leo Degen; Richard Mander; Gitta Salomon

1992-01-01

119

Pattern classification models for classifying and indexing audio signals  

Microsoft Academic Search

In the age of digital information, audio data has become an important part in many modern computer applications. Audio classification and indexing has been becoming a focus in the research of audio processing and pattern recognition. In this paper, we propose effective algorithms to automatically classify audio clips into one of six classes: music, news, sports, advertisement, cartoon and movie.

P. Dhanalakshmi; S. Palanivel; Vennila Ramalingam

2011-01-01

120

High Performance MPEG-Audio Decoder IC.  

National Technical Information Service (NTIS)

The emerging digital audio and video compression technology brings both an opportunity and a new challenge to IC design. The pervasive application of compression technology to consumer electronics will require high volume, low cost IC's and fast time to m...

M. Thorn G. Benbassat K. Cyr S. Li M. Gill

1993-01-01

121

An Improved Audio-Frequency Generator  

Microsoft Academic Search

This paper describes in detail the construction of an audio-frequency generator, for use in making radio-frequency measurments. The variable audio-frequency output is the beat note between two sources of radio frequency; the one a piezo oscillator, and the other a variable oscillator. The output is continuously variable from 50 to 1500 cycles per second. The entire unit is assembled very

E. G. Lapham

1932-01-01

122

An audio-driven dancing avatar  

Microsoft Academic Search

We present a framework for training and synthesis of an audio-driven dancing avatar. The avatar is trained for a given musical\\u000a genre using the multicamera video recordings of a dance performance. The video is analyzed to capture the time-varying posture\\u000a of the dancer’s body whereas the musical audio signal is processed to extract the beat information. We consider two different

Ferda Ofli; Yasemin Demir; Yücel Yemez; Engin Erzin; A. Murat Tekalp; Koray Balc?; ?dil K?zo?lu; Lale Akarun; Cristian Canton-Ferrer; Joëlle Tilmanne; Elif Bozkurt; A. Tanju Erdem

2008-01-01

123

Audio watermarking with error-correcting code  

Microsoft Academic Search

A non-blind two-channel time-frequency digital bits audio watermarking scheme with error-correcting code is described in this paper. The proposed method operates by encoding the watermark bits with cyclic code before embedding them into the audio signal. Time-frequency compression-expansion technique is used to embed the watermark bits. The coefficients to be deleted or added for the time-frequency compression-expansion technique are determined

Htay Htay Yee; Foo Say Wei

2009-01-01

124

Sparsity and Synchrony in Blind Audio-Visual Source Separation  

Microsoft Academic Search

Abstract When dealing with an audio-visual scene, humans tend to integrate both modalities by assessing the temporal synchrony,between,relevant audio and video events. In this way we are able to detect and separate audio-visual sources. Inspired by these observations, we propose a novel method to decompose sequences made of a video signal and a one-microphone audio track into audio-visual sources. First,

Anna Llagostera Casanovas; Gianluca Monaci; Pierre Vandergheynst; Remi Gribonval

125

Methods and systems for image or audio recognition processing  

US Patent & Trademark Office Database

Many of the detailed technologies are useful in enabling a smart phone to respond to a user's environment, e.g., so it can serve as an intuitive hearing and seeing device. A few of the detailed arrangements involve optimizing division of shared processing tasks between the phone and remote devices; using a phone GPU for exhaustive speculative execution and machine vision purposes (including facial recognition); novel device architectures involving abstraction layers that facilitate substitution of different local and remote services; interactions with private networks as they relate to audio/image processing; adapting the orders in which operations are executed, and the types of data that are exchanged with remote servers, in accordance with current context; reconfiguring networks based on sensed social affiliations among users and in accordance with predictive models of user behavior; etc. A great variety of other features and arrangements are also detailed.

2014-07-01

126

Introspective Machine.  

National Technical Information Service (NTIS)

An introspective machine, capable of passing judgment on its own deductive performances, is modelled and analyzed. First, the class of ideal machines which is provided with unlimited resources is studied. Since ideal introspective machines are usually unf...

G. A. W. Vreeswijk

1989-01-01

127

Digital Multicasting of Multiple Audio Streams  

NASA Technical Reports Server (NTRS)

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; Bullock, John

2007-01-01

128

Automatic chord recognition from audio using a supervised HMM trained with audio-from-symbolic data  

Microsoft Academic Search

A novel approach for obtaining labeled training data is presented to directly estimate the model parameters in a supervised learning algorithm for automatic chord recognition from the raw audio. To this end, harmonic analysis is first performed on symbolic data to generate label files. In paral-lel, we synthesize audio data from the same symbolic data, which are then provided to

Kyogu Lee; Malcolm Slaney

2006-01-01

129

Could Audio-Described Films Benefit from Audio Introductions? An Audience Response Study  

ERIC Educational Resources Information Center

Introduction: Time constraints limit the quantity and type of information conveyed in audio description (AD) for films, in particular the cinematic aspects. Inspired by introductory notes for theatre AD, this study developed audio introductions (AIs) for "Slumdog Millionaire" and "Man on Wire." Each AI comprised 10 minutes of…

Romero-Fresco, Pablo; Fryer, Louise

2013-01-01

130

Multimodal audio guide for museums and exhibitions  

NASA Astrophysics Data System (ADS)

In our paper we introduce a new Audio Guide concept for exploring buildings, realms and exhibitions. Actual proposed solutions work in most cases with pre-defined devices, which users have to buy or borrow. These systems often go along with complex technical installations and require a great degree of user training for device handling. Furthermore, the activation of audio commentary related to the exhibition objects is typically based on additional components like infrared, radio frequency or GPS technology. Beside the necessity of installation of specific devices for user location, these approaches often only support automatic activation with no or limited user interaction. Therefore, elaboration of alternative concepts appears worthwhile. Motivated by these aspects, we introduce a new concept based on usage of the visitor's own mobile smart phone. The advantages in our approach are twofold: firstly the Audio Guide can be used in various places without any purchase and extensive installation of additional components in or around the exhibition object. Secondly, the visitors can experience the exhibition on individual tours only by uploading the Audio Guide at a single point of entry, the Audio Guide Service Counter, and keeping it on her or his personal device. Furthermore, since the user usually is quite familiar with the interface of her or his phone and can thus interact with the application device easily. Our technical concept makes use of two general ideas for location detection and activation. Firstly, we suggest an enhanced interactive number based activation by exploiting the visual capabilities of modern smart phones and secondly we outline an active digital audio watermarking approach, where information about objects are transmitted via an analog audio channel.

Gebbensleben, Sandra; Dittmann, Jana; Vielhauer, Claus

2006-02-01

131

Machine Shop Grinding Machines.  

ERIC Educational Resources Information Center

This curriculum manual is one in a series of machine shop curriculum manuals intended for use in full-time secondary and postsecondary classes, as well as part-time adult classes. The curriculum can also be adapted to open-entry, open-exit programs. Its purpose is to equip students with basic knowledge and skills that will enable them to enter the…

Dunn, James

132

Audio stream classification for multimedia database search  

NASA Astrophysics Data System (ADS)

Search and retrieval of huge archives of Multimedia data is a challenging task. A classification step is often used to reduce the number of entries on which to perform the subsequent search. In particular, when new entries of the database are continuously added, a fast classification based on simple threshold evaluation is desirable. In this work we present a CART-based (Classification And Regression Tree [1]) classification framework for audio streams belonging to multimedia databases. The database considered is the Archive of Ethnography and Social History (AESS) [2], which is mainly composed of popular songs and other audio records describing the popular traditions handed down generation by generation, such as traditional fairs, and customs. The peculiarities of this database are that it is continuously updated; the audio recordings are acquired in unconstrained environment; and for the non-expert human user is difficult to create the ground truth labels. In our experiments, half of all the available audio files have been randomly extracted and used as training set. The remaining ones have been used as test set. The classifier has been trained to distinguish among three different classes: speech, music, and song. All the audio files in the dataset have been previously manually labeled into the three classes above defined by domain experts.

Artese, M.; Bianco, S.; Gagliardi, I.; Gasparini, F.

2013-03-01

133

Supported eText: Effects of Text-to-Speech on Access and Achievement for High School Students with Disabilities  

ERIC Educational Resources Information Center

Students with disabilities often lack the skills required to access the general education curriculum and achieve success in school and postschool environments. Evidence suggests that using assistive technologies such as digital texts and translational supports enhances outcomes for these students (Anderson-Inman & Horney, 2007). The purpose of the…

Izzo, Margo Vreeburg; Yurick, Amanda; McArrell, Bianca

2009-01-01

134

AudioGene: Predicting Hearing Loss Genotypes from Phenotypes to Guide Genetic Screening  

PubMed Central

Autosomal Dominant Nonsyndromic Hearing Loss (ADNSHL) is a common and often progressive sensory deficit. ADNSHL displays a high degree of genetic heterogeneity, and varying rates of progression. Accurate, comprehensive and cost-effective genetic testing facilitates genetic counseling and provides valuable prognostic information to affected individuals. In this paper, we describe the algorithm underlying AudioGene, a software system employing machine-learning techniques that utilizes phenotypic information derived from audiograms to predict the genetic cause of hearing loss in persons segregating ADNSHL. Our data show that AudioGene has an accuracy of 68% in predicting the causative gene within its top three predictions, as compared to 44% for a Majority classifier. We also show that AudioGene remains effective for audiograms with high levels of clinical measurement noise. We identify audiometric outliers for each genetic locus and hypothesize that outliers may reflect modifying genetic effects. As personalized genomic medicine becomes more common, AudioGene will be increasingly useful as a phenotypic filter to assess pathogenicity of variants identified by massively parallel sequencing.

Taylor, Kyle R.; DeLuca, Adam P.; Shearer, A. Eliot; Hildebrand, Michael S.; Black-Ziegelbein, E. Ann; Anand, V. Nikhil; Sloan, Christina M.; Eppsteiner, Robert W.; Scheetz, Todd E.; Huygen, Patrick L. M.; Smith, Richard J. H.; Braun, Terry A.; Casavant, Thomas L.

2013-01-01

135

FM-MRR analog audio system  

NASA Astrophysics Data System (ADS)

In this work, we describe a hybrid free-space infrared communications link that supports audio transmission. The technique combines conventional frequency modulation (FM) techniques with optical amplitude modulation (AM) with a Multiple Quantum Well (MQW) Modulating Retroreflector (MRR) technology. The method has produced a robust, low power system capable of transmitting high quality audio information over a free space infrared link extending to multiple kilometers, depending on the characteristics of the Transmit/Receiver (interrogator) and the sensor/ MRR unit at the data source.

Murphy, J. L.; Gilbreath, G. C.; Rabinovich, W. S.; Sepantaie, M. M.; Goetz, P. G.

2005-08-01

136

DWT-Based High Capacity Audio Watermarking  

NASA Astrophysics Data System (ADS)

This letter suggests a novel high capacity robust audio watermarking algorithm by using the high frequency band of the wavelet decomposition, for which the human auditory system (HAS) is not very sensitive to alteration. The main idea is to divide the high frequency band into frames and then, for embedding, the wavelet samples are changed based on the average of the relevant frame. The experimental results show that the method has very high capacity (about 5.5kbps), without significant perceptual distortion (ODG in [-1, 0] and SNR about 33dB) and provides robustness against common audio signal processing such as added noise, filtering, echo and MPEG compression (MP3).

Fallahpour, Mehdi; Megías, David

137

Sparse audio representations using the MCLT  

Microsoft Academic Search

Abstract We consider sparse representations of audio based around,the modulated,complex,lapped transform,(MCLT) and a generalized,iteratively reweighted,least squares,algorithm,which,can,be interpreted as a variation of expectation maximization.,We compare,this mildly overcomplete,representation,to the more,traditional modified,discrete cosine transform,(MDCT) in terms of coding cost and explore the possibility of extending it to a dual-resolution analysis using a pair of MCLT transforms, illustrating its potential application for audio modification.

Mike E. Davies; Laurent Daudet

2006-01-01

138

High performance MPEG-audio decoder IC  

NASA Technical Reports Server (NTRS)

The emerging digital audio and video compression technology brings both an opportunity and a new challenge to IC design. The pervasive application of compression technology to consumer electronics will require high volume, low cost IC's and fast time to market of the prototypes and production units. At the same time, the algorithms used in the compression technology result in complex VLSI IC's. The conflicting challenges of algorithm complexity, low cost, and fast time to market have an impact on device architecture and design methodology. The work presented in this paper is about the design of a dedicated, high precision, Motion Picture Expert Group (MPEG) audio decoder.

Thorn, M.; Benbassat, G.; Cyr, K.; Li, S.; Gill, M.; Kam, D.; Walker, K.; Look, P.; Eldridge, C.; Ng, P.

1993-01-01

139

47 CFR 10.520 - Common audio attention signal.  

Code of Federal Regulations, 2012 CFR

...FEDERAL COMMUNICATIONS COMMISSION GENERAL COMMERCIAL MOBILE ALERT SYSTEM Equipment Requirements § 10.520 Common audio...d) The audio attention signal must be restricted to use for Alert Messages under part 10. (e) A device may include the...

2012-10-01

140

47 CFR 10.520 - Common audio attention signal.  

Code of Federal Regulations, 2011 CFR

...FEDERAL COMMUNICATIONS COMMISSION GENERAL COMMERCIAL MOBILE ALERT SYSTEM Equipment Requirements § 10.520 Common audio...d) The audio attention signal must be restricted to use for Alert Messages under part 10. (e) A device may include the...

2011-10-01

141

47 CFR 10.520 - Common audio attention signal.  

Code of Federal Regulations, 2010 CFR

...FEDERAL COMMUNICATIONS COMMISSION GENERAL COMMERCIAL MOBILE ALERT SYSTEM Equipment Requirements § 10.520 Common audio...d) The audio attention signal must be restricted to use for Alert Messages under part 10. (e) A device may include the...

2010-10-01

142

47 CFR 10.520 - Common audio attention signal.  

Code of Federal Regulations, 2013 CFR

...FEDERAL COMMUNICATIONS COMMISSION GENERAL WIRELESS EMERGENCY ALERTS Equipment Requirements § 10.520 Common audio attention...d) The audio attention signal must be restricted to use for Alert Messages under part 10. (e) A device may include the...

2013-10-01

143

Estimating tempo, swing and beat locations in audio recordings  

Microsoft Academic Search

The problem of estimating the tempo of audio recordings (the number of beats per minute, or BPM) has received an increasing amount of attention in the past few years. Applications include the synchronization of multiple audio tracks for simultaneous playback, \\

J. Laroche

2001-01-01

144

System and Method Using Blind Change Detection for Audio Segmentation.  

National Technical Information Service (NTIS)

A system, method and computer program product for performing blind change detection audio segmentation that combines hypothesized boundaries from several segmentation algorithms to achieve the final segmentation of the audio stream. Automatic segmentation...

G. N. Ramaswamy M. Kamal U. V. Chaudhari

2005-01-01

145

Highly realistic audio spatialization for multiparty conferencing using headphones  

Microsoft Academic Search

It is known that during multi-party conferencing spatialized audio which maps remote participants' voices to distinct virtual locations improves the listening experience. In this paper, we consider the case when the audio is rendered through headphones due to e.g. privacy reasons. Although existing headphone spatial audio techniques abound, most lack the desired realism dictated by listeners' expectation of naturalness in

Wei-Ge Chen; Zhengyou Zhang

2009-01-01

146

Audio content analysis for online audiovisual data segmentation and classification  

Microsoft Academic Search

While current approaches for audiovisual data segmentation and classification are mostly focused on visual cues, audio signals may actually play a more important role in content parsing for many applications. An approach to automatic segmentation and classification of audiovisual data based on audio content analysis is proposed. The audio signal from movies or TV programs is segmented and classified into

Tong Zhang; C.-C. Jay Kuo

2001-01-01

147

Speech Recognition on MPEG\\/Audio Encoded Files  

Microsoft Academic Search

A technique to peform speech recognition directly from audio files encoded using the MPEG\\/Audio coding standard is described. The technique works in the com- pressed domain and does not require the MPEG\\/Audio file to be decompressed. Only the encoded subband sam- ples are extracted and processed for training and recog- nition. The underlying speech recognition engine used is based on

Lawrence Yapp; Gregory L. Zick

1997-01-01

148

Content-Based Classification, Search, and Retrieval of Audio  

Microsoft Academic Search

Many audio and multimedia applications would benefit from the ability to classify and search for audio based on its characteristics. The audio analysis, search, and classification engine described here reduces sounds to perceptual and acoustical features. This lets users search or retrieve sounds by any one feature or a combination of them, by specifying previously learned classes based on these

Erling Wold; Thom Blum; Douglas Keislar; James Wheaton

1996-01-01

149

Analysis of Audio Packet Loss in the Internet  

Microsoft Academic Search

We consider the problem of distributing audio data over networks such as the Internet that do not provide support for real-time applications. Experiments with such networks indicate that audio quality is mediocre in large part because of excessive audio packet losses. In this paper, we show using measurements over the Internet as well as analytic modeling that the number of

Jean-chrysostome Bolot; Hugues Crépin; Andrés Vega-garcía

1995-01-01

150

Dynamic Soundscape: mapping time to space for audio browsing  

Microsoft Academic Search

Browsing audio data is not as easy as browsing printed documents because of the temporal nature of sound. This paper presents a browsing environment that provides a spatial interface for temporal navigation of audio data, taking advantage of human abilities of simultaneous listening and memory of spatial location. Instead of fast-forwarding or rewinding, users browse the audio data by switching

Minoru Kobayashi; Chris Schmandt

1997-01-01

151

Robust detection of audio watermarks after acoustic path transmission  

Microsoft Academic Search

This paper presents an algorithm to improve the robustness of audio watermarks after acoustic path transmission. This includes a brief presentation of the variety of signal alterations a watermarked audio track undergo in such a scenario. After presenting the underlying audio watermarking algorithm, the involved effects with respect to the correlation based detection will be discussed. Based on this observations

Michael Arnold; Peter G. Baum; Xiao-Ming Chen

2010-01-01

152

Unsupervised Audio Segmentation using Extended Baum-Welch Transformations  

Microsoft Academic Search

Audio segmentation has applications in a variety of contexts, such as audio information retrieval, automatic sound analysis, and as a pre-processing step in speech recognition. Extended Baum-Welch (EBW) transformations are most commonly used as a discriminative technique for estimating parameters of Gaussian mixtures. In this paper, we derive an unsupervised audio segmentation approach us- ing these transformations. Weønd that our

Tara N. Sainath; Dimitri Kanevsky; Giridharan Iyengar

2007-01-01

153

Reconstructing audio signals from modified non-coherent hilbert envelopes  

Microsoft Academic Search

In this paper, we present a speech and audio analysis-synthesis method based on a Basilar Membrane (BM) model. The audio signal is represented in this method by the Hilbert envelopes of the responses to complex gammatone filters uniformally spaced on a critical band scale. We show that for speech and audio sig- nals, a perceptually equivalent signal can be reconstructed

Joachim Thiemann; Peter Kabal

2007-01-01

154

Using the Fisher kernel method for Web audio classification  

Microsoft Academic Search

As the multimedia content of the Web increases techniques to automatically classify this content become more important. We present a system to classify audio files collected from the Web. The system classifies any audio file as belonging to one of three categories: speech, music and other. To classify the audio files, we use the technique of Fisher kernels. The technique

Pedro J. Moreno; Ryan Rifkin

2000-01-01

155

Optimizing high quality audio coding: advantages of full system observability  

Microsoft Academic Search

Perceptual audio coders rely on the efficient reduction of perceptually irrelevant components of the audio signal as well as on the removal of statistical signal redundancies to achieve good coding gains. In order to reach high compression ratios without reducing the subjective quality of the encoded audio signal, it is necessary to identify critically interdependent functional units of the encoding

A. J. S. Ferreira

1995-01-01

156

Mobile Video Capture Targeted Narrowband Audio Content Classification  

Microsoft Academic Search

Audio content analysis and automatic content management research field is facing new challenges from personal home video material created with mobile camera phones. We developed an audio content analysis and segmentation system that is robust to low sampling rate used in mobile phone camcorder tools and operates also well for AMR compressed data. Audio signal is segmented in five different

Satu-Marja Mäkelä; Johannes Peltola; Mikko Myllyniemi

2006-01-01

157

To Make a Long Story Short: Abridged Audio at 10.  

ERIC Educational Resources Information Center

Examines the history of abridged audio publishing 10 years after the formation of the Audio Publishers Association. Topics include abridged versus unabridged versions for bookstores and libraries; vendors and publishers; future possibilities for CDs and DVD (Digital Versatile Disc); and audio leasing for libraries. (LRW)

Annichiarico, Mark

1996-01-01

158

Spanish for Agricultural Purposes: The Audio Program.  

ERIC Educational Resources Information Center

The manual is meant to accompany and supplement the basic manual and to serve as support to the audio component of "Spanish for Agricultural Purposes," a one-semester course for North American agriculture specialists preparing to work in Latin America, consists of exercises to supplement readings presented in the course's basic manual and to…

Mainous, Bruce H.; And Others

159

Study of audio speakers containing ferrofluid  

NASA Astrophysics Data System (ADS)

This work validates a method for increasing the radial restoring force on the voice coil in audio speakers containing ferrofluid. In addition, a study is made of factors influencing splash loss of the ferrofluid due to shock. Ferrohydrodynamic analysis is employed throughout to model behavior, and predictions are compared to experimental data.

Rosensweig, R. E.; Hirota, Y.; Tsuda, S.; Raj, K.

2008-05-01

160

Musical audio analysis using sparse representations  

Microsoft Academic Search

Sparse representations are becoming an increasingly useful tool in the analysis of musical audio signals. In this paper we\\u000a will given an overview of work by ourselves and others in this area, to give a flavour of the work being undertaken, and to\\u000a give some pointers for further information about this interesting and challenging research topic.

Mark D. Plumbley; Samer A. Abdallah; Thomas Blumensath; Maria G. Jafari; Andrew Nesbit; Emmanuel Vincent; Beiming Wang

161

Bose Learning Center - Audio Demonstrator Technology  

NSDL National Science Digital Library

Here's information about a new audio technology created at the Bose Corporation, a large manufacturer of sound equipment. The new technology lets the building managers of large places like arenas, auditoriums and outdoor stadiums, preview how the sound system will sound before it's installed.

Corporation, Bose

162

Joint audio visual retrieval for tennis broadcasts  

Microsoft Academic Search

In recent years, there has been increasing work in the area of content retrieval for sports. The idea is generally to extract important events or create summaries to allow personalisation of the media stream. While previous work in sports analysis has employed either the audio or video stream to achieve some goal, there is little work that explores how much

Rozenn Dahyot; A. Kokaram; N. Rea; Hugh Denman

2003-01-01

163

INTERACTIVE AUDIO STRATEGIES FOR DEVELOPING LISTENING SKILLS  

Microsoft Academic Search

This article is based on a paper presented at the CALICO '89 Sixth Annual International Symposium, Colorado Springs, Colorado. This article describes and discusses some ideas for exercise types that can best exploit the capabilities of the computer-controlled laser audio disc in the design and development of foreign language listening comprehension learning activities. The exercises considered include vocabulary recognition, discovering

Jorge Salazar

164

Audio Mining: The Next Big Thing?  

NSDL National Science Digital Library

To get an idea of what audio mining is and how it can be used, people can read this article from the Cutter Consortium. It lists six broad areas that can benefit from using the technology and briefly discusses each one: technical support centers and help desks, call centers, broadcast media, conference managers, intelligence gathering, law enforcement, and security operations.

2008-01-30

165

Sparse audio representations using the MCLT  

Microsoft Academic Search

We consider sparse representations of audio based around the modulated complex lapped transform (MCLT) and a generalized iteratively reweighted least squares algorithm which can be interpreted as a variation of expectation maximization. We compare this mildly overcomplete representation to the more traditional modified discrete cosine transform (MDCT) in terms of coding cost and explore the possibility of extending it to

M. E. Davies; L. Daudet

166

Automatic Musical Genre Classification of Audio Signals  

Microsoft Academic Search

Musical genres are categorical descriptions that are used to describe music. They are c ommonly used to structure the increasing amounts of music available in digital form on the Web and are important for music information retrieval. Genre ca tegorization for audio has traditionally been performed manually. A particular musical genre is characterized by statistical properties related to the instrumentation,

George Tzanetakis

2001-01-01

167

Lessons from Developing Audio HTML Interfaces  

Microsoft Academic Search

In this paper, we discuss our previous research on the estab- lishment of guidelines and principles for choosing sounds to use in an audio interface to HTML, called the AHA frame- work. These principles, along with issues related to the tar- get audience such as user tasks, goals, and interests are factors that can help us to choose specific sounds

Frankie James

168

Audio-Visual Speech Perception Is Special  

ERIC Educational Resources Information Center

In face-to-face conversation speech is perceived by ear and eye. We studied the prerequisites of audio-visual speech perception by using perceptually ambiguous sine wave replicas of natural speech as auditory stimuli. When the subjects were not aware that the auditory stimuli were speech, they showed only negligible integration of auditory and…

Tuomainen, J.; Andersen, T.S.; Tiippana, K.; Sams, M.

2005-01-01

169

Memoirs of Togetherness from Audio Logs  

Microsoft Academic Search

In this paper, we propose a new concept how tempo-social information about moments of togetherness within a social group of people can be retrieved in the palm of the hand from social context. The social context is digitised by audio logging of the same user centric device such as mobile phone. Being asynchronously driven it allows automatically logging social events

Danil Korchagin

2009-01-01

170

Fast subband filtering in MPEG audio coding  

Microsoft Academic Search

Subband filtering is one of the most compute-intensive operations in the MPEG audio coding standard. The author proves that the matrixing operations in MPEG subband filtering can be efficiently computed using fast 32-point DCT or inverse DCT (IDCT) algorithms

K. Konstantinides

1994-01-01

171

Independent component analysis for audio classification  

Microsoft Academic Search

In this paper, we explore the performance gains achieved by performing independent component analysis (ICA) decomposition on speech features obtained from a model of the early auditory system. ICA projection achieves dimensionality reduction by reducing the redundancy in the feature set (the transformed features are statistically independent). Performance is evaluated for an audio classification environment using a Gaussian mixture model

Sunil Kamath; Sourabh Ravindran; David V. Anderson

2004-01-01

172

MARSYAS3D: A PROTOTYPE AUDIO BROWSER-EDITOR USING A LARGE SCALE IMMERSIVE VISUAL AND AUDIO DISPLAY  

Microsoft Academic Search

Most audio editing tools offer limited capabilities for browsing and editing large collections of files. Moreover working with many audio files tends to clutter the limited screen space of a desktop monitor. In this paper we describe MARSYAS3D, a prototype au- dio browser and editor for large audio collections. A variety of 2D and 3D graphics interfaces for working with

George Tzanetakis; Perry Cook

2001-01-01

173

Kid Machine  

NSDL National Science Digital Library

This activity is on page 3 (continued on page 2) of the pdf, part of the Simple Machines Discovery Box. In this fun activity, learners "create" a complex machine by simulating the parts in action. Learners move their bodies and make sounds as if they are individual parts of a moving machine. Then learners discover what happens when part of a machine is broken and problem solve ways to fix it.

Omsi

2004-01-01

174

Electrostatic Machines  

NSDL National Science Digital Library

This website from Antonio Carlos M. De Queiroz, an associate professor at the Federal University of Rio de Janeiro, illustrates a number of different electrostatic machines. The site includes details and images of machines built by the professor as well as many other historical machines of this type. Some information is also available in Portugese.

De Queiroz, Antonio C.

2011-07-13

175

Simple Machines  

NSDL National Science Digital Library

This is an online activity about simple machines. Learners will try their hand at putting these amazing devices to work. They will use several simple machines to help "build" a tree house. This is an excellent activity to demonstrate how science - in particular, simple machines - are at work in our everyday lives.

Cosi

2000-01-01

176

Three-dimensional audio using loudspeakers  

NASA Astrophysics Data System (ADS)

3-D audio systems, which can surround a listener with sounds at arbitrary locations, are an important part of immersive interfaces. A new approach is presented for implementing 3-D audio using a pair of conventional loudspeakers. The new idea is to use the tracked position of the listener's head to optimize the acoustical presentation, and thus produce a much more realistic illusion over a larger listening area than existing loudspeaker 3-D audio systems. By using a remote head tracker, for instance based on computer vision, an immersive audio environment can be created without donning headphones or other equipment. The general approach to a 3-D audio system is to reconstruct the acoustic pressures at the listener's ears that would result from the natural listening situation to be simulated. To accomplish this using loudspeakers requires that first, the ear signals corresponding to the target scene are synthesized by appropriately encoding directional cues, a process known as 'binaural synthesis,' and second, these signals are delivered to the listener by inverting the transmission paths that exist from the speakers to the listener, a process known as 'crosstalk cancellation.' Existing crosstalk cancellation systems only function at a fixed listening location; when the listener moves away from the equalization zone, the 3-D illusion is lost. Steering the equalization zone to the tracked listener preserves the 3-D illusion over a large listening volume, thus simulating a reconstructed soundfield, and also provides dynamic localization cues by maintaining stationary external sound sources during head motion. This dissertation will discuss the theory, implementation, and testing of a head-tracked loudspeaker 3-D audio system. Crosstalk cancellers that can be steered to the location of a tracked listener will be described. The objective performance of these systems has been evaluated using simulations and acoustical measurements made at the ears of human subjects. Many sound localization experiments were also conducted; the results show that head-tracking both significantly improves localization when the listener is displaced from the ideal listening location, and also enables dynamic localization cues. (Copies available exclusively from MIT Libraries, Rm. 14-0551, Cambridge, MA 02139-4307. Ph. 617-253-5668; Fax 617-253-1690.)

Gardner, William G.

1997-12-01

177

Coding of audio data in halftone images  

NASA Astrophysics Data System (ADS)

A new method for coding and decoding of audio data in halftone images is described and experimentally verified. Coding (modulation) is achieved by replacing printed dots by a set of two-dimensional characteristic symbols. Decoding (demodulation) is achieved by a bank of two-dimensional matched-filters. The system comprises source and channel coding, in order to compress and decompress the audio data and in order to avoid loss of data if the image is degraded physically. Up to 9 seconds of music in CD quality or 20 seconds of speech have been stored in an area of only 10cm2. Decoding is possible with a small hand-held scanner.

Wirnitzer, Bernhard; Dillmann, Vadim; Latorre, Javier

178

Building an Audio Visualizer in Flex  

Microsoft Academic Search

Prior to Flash Player 9, Flash developers had to rely on third-party applications to create equalizer-type displays or audio\\u000a visualizers. Now, thanks to improvements to the ActionScript language, you can create these experience enhancers natively.\\u000a Combine that with the relative ease of use of the Flex framework, and you can come up with some pretty amazing visualizations\\u000a with minimal effort.

Hasan Otuome

179

Extracting Keyphrases from Spoken Audio Documents  

Microsoft Academic Search

Spoken audio documents are becoming more and more common on the World Wide Web, and this is likely to be accelerated by the\\u000a widespread deployment of broadband technologies. Unfortunately, speech documents are inherently hard to browse because of\\u000a their transient nature. One approach to this problem is to label segments of a spoken document with keyphrases that summarise\\u000a them. In

Alain Désilets; Berry De Bruijn; Joel D. Martin

2001-01-01

180

Audio processing technology for law enforcement  

Microsoft Academic Search

The Air Force Research Laboratory Multi-Sensor Exploitation Branch (AFRL\\/IFEC) has been a Department of Defense leader in research and development (R&D) in speech and audio processing for over 25 years. Their primary thrust in these R&D areas has focused on developing technology to improve the collection, handling, identification, and intelligibility of military communication signals. The National Law Enforcement and Corrections

Sharon M. Walter; Maria Cofano; Roy J. Ratley

1999-01-01

181

Robust and Inaudible Multiecho Audio Watermarking  

Microsoft Academic Search

A novel echo embedding technique is proposed to overcome inherent trade-off between in-audibility and robustness in conventional\\u000a echo hiding. It makes use of masking model to embed two echoes by both positive and negative pulses (closely located) and\\u000a high energy to host audio signals. Subjective listening tests show that the proposed method could improve robustness to operations\\u000a of noise addition,

Dong-yan Huang; Theng Yee Yeo

2002-01-01

182

Capacity-optimized mp2 audio watermarking  

NASA Astrophysics Data System (ADS)

Today a number of audio watermarking algorithms have been proposed, some of them at a quality making them suitable for commercial applications. The focus of most of these algorithms is copyright protection. Therefore, transparency and robustness are the most discussed and optimised parameters. But other applications for audio watermarking can also be identified stressing other parameters like complexity or payload. In our paper, we introduce a new mp2 audio watermarking algorithm optimised for high payload. Our algorithm uses the scale factors of an mp2 file for watermark embedding. They are grouped and masked based on a pseudo-random pattern generated from a secret key. In each group, we embed one bit. Depending on the bit to embed, we change the scale factors by adding 1 where necessary until it includes either more even or uneven scale factors. An uneven group has a 1 embedded, an even group a 0. The same rule is later applied to detect the watermark. The group size can be increased or decreased for transparency/payload trade-off. We embed 160 bits or more in an mp2 file per second without reducing perceived quality. As an application example, we introduce a prototypic Karaoke system displaying song lyrics embedded as a watermark.

Steinebach, Martin; Dittmann, Jana

2003-06-01

183

Measuring human readability of machine generated text: three case studies in speech recognition and machine translation  

Microsoft Academic Search

We present highlights from three experiments that test the readability of current state-of-the art system output from: (1) an automated English speech-to-text (SST) system; (2) a text-based Arabic-to-English machine translation (MT) system; and (3) an audio-based Arabic-to-English MT process. We measure readability in terms of reaction time and passage comprehension in each case, applying standard psycholinguistic testing procedures and a

Douglas Jones; Edward Gibson; Wade Shen; Neil Granoien; Martha Herzog; Douglas Reynolds; Clifford Weinstein

2005-01-01

184

MRI with synchronized audio to evaluate velopharyngeal insufficiency.  

PubMed

Objective : To demonstrate the feasibility of simultaneous-acquired magnetic resonance imaging (MRI) and high-quality synchronized audio recording for evaluating velopharyngeal closure. Design : Institutional Review Board-approved case series. Setting : Tertiary care hospital. Patients : Three healthy adult volunteers with a normal speech pattern. Interventions : MRI with simultaneous recorded audio files evaluating velopharyngeal closure. Main outcome measure : Precise imaging and audio coordination of specific phonatory tasks. Results : Synchronization of MRI and audio in all three adults. Conclusion : Our novel imaging and audio protocol provides simultaneous acquired MRI with synchronized high quality audio for evaluating velopharyngeal closure. This technique may provide the opportunity to improve diagnosis and surgical planning in patients with velopharyngeal insufficiency. PMID:21740179

Maturo, Stephen; Silver, Amanda; Nimkin, Katherine; Sagar, Pallavi; Ashland, Jean; van der Kouwe, Andre J W; Hartnick, Christopher

2012-11-01

185

Precision Machining  

NSDL National Science Digital Library

Basic machining processes are introduced on a Web site that is devoted to engineering fundamentals (1). Descriptions and illustrations of drilling, turning, grinding, and other common processes are provided for people with little to no prior machining knowledge. A waterjet is a non-traditional machining technology that uses high pressure streams of water with abrasive additives rather than solid cutting instruments to slice through metal and other materials. An in-depth discussion of waterjet operation and applications is available from Southern Methodist University (2). Waterjets are often cited as being much more precise than traditional machining techniques. The Waterjet Video Vault (3) contains clips of waterjet machines in action. The video of the foam cutting procedure is especially interesting, as it shows how quick and accurate the machining process can be. An online guide to cross process machining, which incorporates elements from various conventional and unconventional techniques, is provided by the Mechanical Engineering Department at Columbia University (4). Some remarkable and innovative techniques that have surfaced over the past few years are outlined, including underwater laser machining and plasma-assisted machining. Entirely different and exotic machining techniques are required for creating microelectromechanical systems (MEMS) and other extremely small devices. The Caltech Micromachining Laboratory (5) maintains an archive of research highlights and papers on its homepage, including a paper on a MEMS-driven flapping wing for a palm-sized aerial vehicle. An online article from Modern Machine Shop (6) outlines some new technologies and research in the area of high speed machining. A particularly interesting section of the article describes a system developed at the University of Florida that aims to enable micromachining to achieve rotational speeds of standard machining processes, specifically up to a half million rotations per minute. Cutting edge waterjet innovations are the subject of a February 2003 feature from a publication of the Society of Manufacturing Engineers (7). Extremely high pressure nozzles are being developed to improve cutting speed, and enhanced software for controlling machine movements is also a focus of study. This news article (8) from June 20, 2003 describes an electrochemical machining process that is being used to fabricate complex nanostructures. The work, produced by German and U.S. researchers, has the potential to compete with current lithographic processes.

Leske, Cavin.

186

A new spectral enhancement algorithm in MP3 audio  

Microsoft Academic Search

This paper proposes a new blind method of spectral (high frequency) enhancement of MP3 audio. Due to the protocol constraint, the audio bandwidth of MP3 is restricted to 16 kHz at 128 kbps and 12 kHz at 64 kbps. Although band-restricted MP3 audio provides savings of storage space and network bandwidth, it suffers a major problem of a loss in

Sang-heon Oh; Won-Jung Yoon; Youn-ho Cho; Kyu-Sik Park; Ki-Man Kim

2006-01-01

187

Multistream Asynchrony Modeling for Audio-Visual Speech Recognition  

Microsoft Academic Search

In this paper, two multi-stream asynchrony Dynamic Bayesian Network models (MS-ADBN model and MM-ADBN model) are proposed for audio-visual speech recognition (AVSR). The proposed models, with different topology structures, loose the asynchrony of audio and visual streams to word level. For MS-ADBN model, both in audio stream and in visual stream, each word is composed of its corresponding phones, and

Guoyun Lv; Dongmei Jiang; Rongchun Zhao; Yunshu Hou

2007-01-01

188

DIGITAL AUDIO EFFECTS APPLIED DIRECTLY ON A DSD BITSTREAM  

Microsoft Academic Search

Digital audio effects are typically implemented on 16 or 24 bit signals sam- pled at 44.1 kHz. Yet high quality audio is often encoded in a one-bit, highly oversampled format , such as DSD. Processing of a bitstream, and the application of audio effects on a bitstream, requires special care and modification of existing methods. However, it has strong advantages

Josh Reiss; Mark Sandler

2004-01-01

189

Voice activity detection using audio-visual information  

Microsoft Academic Search

An audio-visual voice activity detector that uses sensors positioned distantly from the speaker is presented. Its constituting unimodal detectors are based on the modeling of the temporal variation of audio and visual features using hidden Markov models; their outcomes are fused using a post-decision scheme. The Mel-frequency cepstral coefficients and the vertical mouth opening are the chosen audio and visual

Theodoros Petsatodis; Aristodemos Pnevmatikakis; Christos Boukis

2009-01-01

190

Nonplanar machines  

SciTech Connect

This talk examines methods available to minimize, but never entirely eliminate, degradation of machine performance caused by terrain following. Breaking of planar machine symmetry for engineering convenience and/or monetary savings must be balanced against small performance degradation, and can only be decided on a case-by-case basis. 5 refs.

Ritson, D. (Stanford Linear Accelerator Center, Menlo Park, CA (USA))

1989-05-01

191

Electric machine  

DOEpatents

An interior permanent magnet electric machine is disclosed. The interior permanent magnet electric machine comprises a rotor comprising a plurality of radially placed magnets each having a proximal end and a distal end, wherein each magnet comprises a plurality of magnetic segments and at least one magnetic segment towards the distal end comprises a high resistivity magnetic material.

El-Refaie, Ayman Mohamed Fawzi (Niskayuna, NY); Reddy, Patel Bhageerath (Madison, WI)

2012-07-17

192

Excavating machines  

SciTech Connect

The excavating machine has a cutter carrying boom carried by a boom support member which can be swung about an axis extending in the direction of the roadway. The machine includes a cutter unit and a stay unit each of which is releasably anchorable in the roadway and each of which can be advanced relative to the other unit.

Plummer, D.

1980-10-21

193

Machine Learning.  

ERIC Educational Resources Information Center

As scientists seek to develop machines that can "learn," that is, solve problems by imitating the human brain, a gold mine of information on the processes of human learning is being discovered, expert systems are being improved, and human-machine interactions are being enhanced. (SK)

Kirrane, Diane E.

1990-01-01

194

Machine Learning  

Microsoft Academic Search

The purpose of this chapter is to present fundamental ideas and techniques of machine learning suitable for the field of this book, i.e., for automated scientific discovery. The chapter focuses on those symbolic machine learning methods, which produce results that are suitable to be interpreted and understood by humans. This is particularly important in the context of automated scientific discovery

Achim Hoffmann; Ashesh Mahidadia

2009-01-01

195

Marble track audio manipulator (MTAM): a tangible user interface for audio composition  

Microsoft Academic Search

We created a tangible user interface that allows children to create musical compositions through constructive play. Our Marble Track Audio Manipulator (MTAM) is an augmented marble tower construction kit where marbles represent sound clips and tracks represent different sound effects. To create musical compositions, children collaboratively build a marble tower and then play their compositions by dropping marbles into the

Alex Bean; Sabina Siddiqi; Anila Chowdhury; Billy Whited; Orit Shaer; Robert J. K. Jacob

2008-01-01

196

Instructional Audio Guidelines: Four Design Principles to Consider for Every Instructional Audio Design Effort  

ERIC Educational Resources Information Center

This article contends that instructional designers and developers should attend to four particular design principles when creating instructional audio. Support for this view is presented by referencing the limited research that has been done in this area, and by indicating how and why each of the four principles is important to the design process.…

Carter, Curtis W.

2012-01-01

197

Audio Adapted Assessment Data: Does the Addition of Audio to Written Items Modify the Item Calibration?  

ERIC Educational Resources Information Center

This dissertation research examined the changes in item RIT calibration that occurred when adding audio to a set of currently calibrated RIT items and then placing these new items as field test items in the modified assessments on the NWEA MAP test platform. The researcher used test results from over 600 students in the Poway School District in…

Snyder, James

2010-01-01

198

Quantization and psychoacoustic model in audio coding in advanced audio coding  

NASA Astrophysics Data System (ADS)

This paper presents complete optimized architecture of Advanced Audio Coder quantization with Huffman coding. After that psychoacoustic model theory is presented and few algorithms described: standard Two Loop Search, its modifications, Genetic, Just Noticeable Level Difference, Trellis-Based and its modification: Cascaded Trellis-Based Algorithm.

Brzuchalski, Grzegorz

2011-06-01

199

ABC News: Video and Audio Newsclips  

NSDL National Science Digital Library

ABC News has added a section of video and audio newsclips to its news service at the GO Network, InfoSeek Corporation's Internet portal. Users can see and listen to national headline news, such as a clip from Warren Beatty's speech at an awards dinner Wednesday night (sounding rather presidential). They can also search for additional video files using Videosearch, by Virage. Beatty as a search term turned up a clip about the Clinton family's summer vacation on Martha's Vineyard that included a mention of Beatty's presidential aspirations and opinions on the Democratic Party, but no additional pictures of Beatty.

200

Scribbling Machines  

NSDL National Science Digital Library

In this activity, learners explore electronics and motion by making a Scribbling Machine, a motorized contraption that moves in unusual ways and leaves a mark to trace its path. Itâs made from simple materials and is based on the idea of motion created by an offset motor. Try using harvested motors and switches from discarded toys and electronics to make your Scribbling Machine - this not only keeps costs down, but is a playful and inventive way to explore how everyday objects work. To take the activity further, you can also incorporate PicoCrickets to make your Scribbling Machine more intelligent and to explore computers.

Exploratorium

2013-01-30

201

Laboratory and in-Flight Experiments to Evaluate 3-D Audio Display Technology.  

National Technical Information Service (NTIS)

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 ...

M. Ericson R. Mckinley M. Kibbe D. Francis

1994-01-01

202

Electromechanical Machining.  

National Technical Information Service (NTIS)

The purpose of this work was to adapt the new process of electromechanical machining (EMM), in which metal is cut by direct contact of the tool with an electrochemically polarized workpiece, to turning and drilling. This investigation, which followed expe...

R. M. Latanision K. C. Nielsen

1976-01-01

203

Math Machines  

NSDL National Science Digital Library

The mission of the Math Machines organization is to "improve the quality of mathematical education, enhance the transfer of mathematical thinking into other classes, and increase students' ability to apply rigorous mathematics outside the classroom." Their website supports a National Science Foundation ATE grant-supported project designed to improve teaching in the areas of Mathematics, Science, and Technology at the high school and college levels. This improved learning results from using math, science, and technology principles to build and control various machines such as pointers and robots or "math machines", which are simple devices that provide an immediate, physical, dynamic expression to abstract mathematical equations. The website provides information links on Educational Theory, Classroom Activities, Project Workshops, Calculators & Programs, and Machine Construction Instructions for Building: Closed Circuits, Servo Motors, Controllers, Robot Boards and more. There is also contact information, an FAQ section, as well as upcoming events.

2010-05-18

204

Mining machine  

SciTech Connect

A haulage system for a mining machine comprises a mining machine mounted on and/or guided by a conveyor and reciprocable with respect thereto, the conveyor being provided with a rack having plural rows of teeth of identical pitch, with the teeth of one row staggered with respect to an adjacent row(s), and the machine being provided with at least one power driven haulage sprocket comprising plural sets of peripherally arranged teeth of identical pitch, one set being angularly staggered with respect to an adjacent set(s), whereby one set is engageable with each row of teeth of the rack. The invention also includes a mining machine provided with such a power driven haulage sprocket, and a rack as above described and provided with end fittings for securing in articulated manner to an adjacent rack.

Parrott, G.A.

1985-05-07

205

Structuring Soccer Video Based on Audio Classification and Segmentation Using Hidden Markov Model  

Microsoft Academic Search

\\u000a This paper presents a novel scheme for indexing and segmentation of video by analyzing the audio track using Hidden Markov\\u000a Model. This analysis is then applied to structuring the soccer video. Based on the attributes of soccer video, we define three\\u000a audio classes in soccer video, namely Game-audio, Advertisement-audio and Studio-audio. For each audio class, a HMM is built using

Jianyun Chen; Yunhao Li; Songyang Lao; Ling-da Wu; Liang Bai

2004-01-01

206

Using Audio Books to Improve Reading and Academic Performance  

ERIC Educational Resources Information Center

This article highlights significant research about what below grade-level reading means in middle school classrooms and suggests a tested approach to improve reading comprehension levels significantly by using audio books. The use of these audio books can improve reading and academic performance for both English language learners (ELLs) and for…

Montgomery, Joel R.

2009-01-01

207

Audio Podcasting in a Tablet PC-Enhanced Biochemistry Course  

ERIC Educational Resources Information Center

This report describes the effects of making audio podcasts of all lectures in a large, basic biochemistry course promptly available to students. The audio podcasts complement a previously described approach in which a tablet PC is used to annotate PowerPoint slides with digital ink to produce electronic notes that can be archived. The fundamentals…

Lyles, Heather; Robertson, Brian; Mangino, Michael; Cox, James R.

2007-01-01

208

The Use of Audio and Animation in Computer Based Instruction.  

ERIC Educational Resources Information Center

This study investigated the effects of audio, animation, and spatial ability in a computer-based instructional program for biology. The program presented instructional material via test or audio with lean text and included eight instructional sequences presented either via static illustrations or animations. High school students enrolled in a…

Koroghlanian, Carol; Klein, James D.

209

The Effect of Audio and Animation in Multimedia Instruction  

ERIC Educational Resources Information Center

This study investigated the effects of audio, animation, and spatial ability in a multimedia computer program for high school biology. Participants completed a multimedia program that presented content by way of text or audio with lean text. In addition, several instructional sequences were presented either with static illustrations or animations.…

Koroghlanian, Carol; Klein, James D.

2004-01-01

210

Use of Audio Modification in Science Vocabulary Assessment  

ERIC Educational Resources Information Center

The purposes of this study were to examine the utilization of audio modification in vocabulary assessment in school subject areas, specifically in elementary science, and to present a web-based key vocabulary assessment tool for the elementary school level. Audio-recorded readings were used to replace independent student readings as the task…

Adiguzel, Tufan

2011-01-01

211

Audio-Tutorial Practices in California Community Colleges. Preliminary Report.  

ERIC Educational Resources Information Center

This study surveys the audio-tutorial practices and evaluations at 91 California and 25 other junior colleges in the United States. Seventy of the California colleges indicate they are now or will be using the audio-tutorial method in the near future. A large majority of them indicate that they prepare their materials locally. Many of the colleges…

Diablo Valley Coll. Pleasant Hill, CA.

212

Early investigations into subjective audio quality assessment using brainwave responses  

Microsoft Academic Search

In this work, we take the first steps towards quantifying changes in the perceived quality of audio by directly measuring human brainwave responses using a high-resolution electroencephelograph (EEG). Specifically, human subjects are presented with audio whose quality varies with time while being monitored by a 128-channel EEG; some of the time, they move a slider bar up and down to

Charles D. Creusere; Srikant R. Siddenki; Joe Hardin; Jim Kroger

2011-01-01

213

Automotive audio noise assessment. human hearing factors and complexity  

Microsoft Academic Search

Vehicle audio subsystems are becoming increasingly complex. In addition to providing audio entertainment with higher fidelity at higher power levels vehicles have become generally quieter with respect to road, wind and power train noise. The overall effect is an expanded dynamic range. When the engineer is faced with reducing EMC related noise in an entertainment system what metrics are to

Richard Wiese

2006-01-01

214

Audio Frequency Coders Ky-430/Srm and Ky-431/Srm.  

National Technical Information Service (NTIS)

These procedures describe the calibration of Audio Frequency Coder KY-430/SRM and KY-431/SRM. The KY-430/SRM, which is contained in Target Control System Test Set AN/SRM-2 and AN/SRM-5, is a fixed-wing coder that generates the first twenty IRIG audio-freq...

1967-01-01

215

Audio-visual speaker identification using coupled hidden Markov models  

Microsoft Academic Search

In this paper, we investigate the use of the coupled hid- den Markov models (CHMM) for the task of audio-visual text dependent speaker identification. Our system deter- mines the identity of the user from a temporal sequence of audio and visual observations obtained from the acous- tic speech and the shape of the mouth, respectively. The multi modal observation sequences

Tieyan Fu; Xiao Xing Liu; Lu Hong Liang; Xiaobo Pi; Ara V. Nefian

2003-01-01

216

Improved implementation of MDCT in MP3 audio coding  

Microsoft Academic Search

Forward and inverse modified discrete cosine transform (MDCT) are two of the most computational intensive operations in the MPEG audio coding standard. So, their efficient implementations are very important. We present an improved algorithm for fast implementation of forward and inverse MDCT for layer III in MPEG-1 and MPEG-2 audio coding standards. The proposed algorithm requires fewer operations than recently

Vladimir Nikolajevic; G. Fettweis

2004-01-01

217

CLAM, an Object Oriented Framework for Audio and Music  

Microsoft Academic Search

CLAM is a C++ framework that is being developed at the Music Technology Group of the Universitat Pompeu Fabra (Barcelona, Spain). The framework oers a complete development and research platform for the audio and music domain. Apart from oering an abstract model for audio systems, it also includes a repository of processing algorithms and data types as well as a

Pau Arum; Xavier Amatriain

218

Control Mechanisms for Packet Audio in the Internet  

Microsoft Academic Search

The Internet provides a single class best effort service. From an application's point of view, this service amounts in practice to providing channels with time-varying characteristics such as delay and loss distributions. One way to support real time applications such as interactive audio given this service is to use control mechanisms that adapt the audio coding and decoding processes based

Jean-chrysostome Bolot; Andrés Vega-garcía

1996-01-01

219

CIC interpolation filter design in the audio decoder  

Microsoft Academic Search

In order to get high-quality audio output in the audio decoder, analog low-pass filter to reduce the design difficulty, then the interpolation filter as its digital signal processing part of an integral important part of CIC filters are commonly used in the interpolation process highly efficient filters, with a simple structure, easy to implement advantages. This article is in the

Zhou Jinglei; Li Chengliang; Qi Bo; Wei Yanhui

2010-01-01

220

Overcoming asynchrony in Audio-Visual Speech Recognition  

Microsoft Academic Search

In this paper we propose two alternatives to overcome the natural asynchrony of modalities in Audio-Visual Speech Recognition. We first investigate the use of asynchronous statistical models based on Dynamic Bayesian Networks with different levels of asynchrony. We show that audio-visual models should consider asynchrony within word boundaries and not at phoneme level. The second approach to the problem includes

Virginia Estellers; Jean-Philippe Thiran

2010-01-01

221

RTP Profile for Audio and Video Conferences with Minimal Control  

Microsoft Academic Search

This memo describes a profile for the use of the real-time transport protocol (RTP), version 2, and the associated control protocol, RTCP, within audio and video multiparticipant conferences with minimal control. It provides interpretations of generic fields within the RTP specification suitable for audio and video conferences. In particular, this document defines a set of default mappings from payload type

H. Schulzrinne

1996-01-01

222

Content-Based Retrieval of Music and Audio  

Microsoft Academic Search

Though many systems exist for content-based retrieval of images, little work has been done on the audio portionof the multimedia stream. This paper presents a system to retrieve audio documents by acoustic similarity. Thesimilarity measure is based on statistics derived from a supervised vector quantizer, rather than matching simple pitchor spectral characteristics. The system is thus able to learn distinguishing

Jonathan T. Foote

1997-01-01

223

Visualizing music and audio using self-similarity  

Microsoft Academic Search

This paper presents a novel approach to visualizing the time structure of music and audio. The acoustic similarity between any two instants of an audio recording is displayed in a 2D representation, allowing identification of structural and rhythmic characteristics. Examples are presented for classical and popular music. Applications include content-based analysis and segmentation, as well as tempo and structure extraction.

Jonathan Foote

1999-01-01

224

Tune in the Net with RealAudio.  

ERIC Educational Resources Information Center

Describes how to connect to the RealAudio Web site to download a player that provides sound from Web pages to the computer through streaming technology. Explains hardware and software requirements and provides addresses for other RealAudio Web sites are provided, including weather information and current news. (LRW)

Buchanan, Larry

1997-01-01

225

A Detection Algorithm of Audio Spread Spectrum Data Hiding  

Microsoft Academic Search

In this paper, a method of passive steganalysis is proposed. We focus on detecting the existing of data hidden in audio files with spread spectrum (SS) data hiding. SS data hiding is considered as a process of adding noise. The technology of classifier and feature vector extraction are used to achieve the detection. First, we divide an audio signal into

S. Gao; R. M. Hu; W. Zeng; H. J. Ai; C. R. Li

2008-01-01

226

Exposing audio data to the web: an API and prototype  

Microsoft Academic Search

The HTML5 specification introduces the audio and video media elements, and with them the opportunity to change the way media is integrated on the web. The current HTML5 media API provides ways to play and get limited information about audio and video, but no way to programatically access or create such media. In this paper we present an enhanced API

David Humphrey; Corban Brook; Alistair MacDonald

2010-01-01

227

Error Concealment of MPEG-2 AAC Audio Using Modulo Watermarks.  

National Technical Information Service (NTIS)

We propose an error concealment scheme for MPEG-2 compressed (AAC) audio using a novel modulo watermarking technique. It can be used on top of other error control schemes. After the modulo watermark is embedded, an MPEG-2 AAC audio only shows negligible f...

S. Cheng H. Yu Z. Xiong

2002-01-01

228

Jitter-free Audio Playout over Best Effort Packet Networks  

Microsoft Academic Search

The aim of this research is to explore important issues in play- out of audio over best effort audio networks and the improvements that can be made at the upper layers without changing the underlying network. This paper starts with a review of existing methods, the experiments conducted based on them and comments on their performance. The point of departure

Aman Kansal; Abhay Karandikar

229

SKEW DETECTION AND COMPENSATION FOR INTERNET AUDIO APPLICA TIONS  

Microsoft Academic Search

Long lived audio streams, such as music broadcasts, and small differences in clock rates lead to buffer underflow or overflo w events in receiving applications that manifest themselves as au- dible interruptions. We present a low complexity algorithm for detecting clock skew in network audio applications that function with local clocks and in the absence of a synchronization mech- anism.

Orion Hodson; Colin Perkins; Vicky Hardman; Playout Delay

230

Language and genre detection in audio content analysis  

Microsoft Academic Search

This paper presents an audio genre detection framework that can be used for a multi-language audio corpus. Cepstral coefficients are considered and analyzed as the feature set for both a language dependent and language independent genre identification (GID) task. Language information is found to increase the overall detection accuracy on an average by at least 2.6% from its language independent

Vikramjit Mitra; Daniel Garcia-Romero; Carol Y. Espy-Wilson

2008-01-01

231

Informed audio watermarking scheme using digital chaotic signals  

Microsoft Academic Search

An informed audio watermarking scheme using chaotic waveforms is proposed. The watermark consists of an attenuated signal embedded in the original audio signal. A modulation scheme based on vector quantization is used to achieve blind detection of the watermark signal. A method for self-synchronization of the watermark detector is proposed using chaotic orthogonal waveforms. A perceptual model is employed to

G. C. M. Silvestret; N. J. Hurleyt; G. S. Hanaut; W. J. Dowlingt

2001-01-01

232

Psychoacoustic Principles and Genetic Algorithms in Audio Compression  

Microsoft Academic Search

High audio data compression can be achieved by removing irrelevant signal information that is not detectable by even a well- trained or sensitive listener. Contemporary audio coding schemes like MP3, AAC, and Ogg Vorbis identify the irrelevant information during signal analysis by incorporating into the coder several psychoacoustic principles, including absolute hearing thresholds, critical band analysis, simultaneous masking, and temporal

Mandeep Singh Wali; Balwant Singh; Amit Gupta

2007-01-01

233

Comparison of psychoacoustic principles and genetic algorithms in audio compression  

Microsoft Academic Search

High audio data compression can be achieved by removing irrelevant signal information that is not detectable by even a well-trained or sensitive listener. Contemporary audio coding schemes like MP3, AAC, and Ogg Vorbis identify the irrelevant information during signal analysis by incorporating into the coder several psychoacoustic principles, including absolute hearing thresholds, critical band analysis, simultaneous masking, and temporal masking

Howard Chen; T. L. Yu

2005-01-01

234

Getting Started with CD Audio in HyperCard.  

ERIC Educational Resources Information Center

This article examines the use of the Voyager Compact Disk (CD) AudioStack to provide HyperCard stacks designed to promote language learning with the ability to play on common precisely specified portions of off-the-shelf audio compact disks in a CD-ROM drive. Four German and Russian HyperCard stacks are described and their construction outlined.…

Decker, Donald A.

1992-01-01

235

The Audio-Visual Equipment Directory. Seventeenth Edition.  

ERIC Educational Resources Information Center

The following types of audiovisual equipment are catalogued: 8 mm. and 16 mm. motion picture projectors, filmstrip and sound filmstrip projectors, slide projectors, random access projection equipment, opaque, overhead, and micro-projectors, record players, special purpose projection equipment, audio tape recorders and players, audio tape…

Herickes, Sally, Ed.

236

Monel Machining  

NASA Astrophysics Data System (ADS)

Castle Industries, Inc. is a small machine shop manufacturing replacement plumbing repair parts, such as faucet, tub and ballcock seats. Therese Castley, president of Castle decided to introduce Monel because it offered a chance to improve competitiveness and expand the product line. Before expanding, Castley sought NERAC assistance on Monel technology. NERAC (New England Research Application Center) provided an information package which proved very helpful. The NASA database was included in NERAC's search and yielded a wealth of information on machining Monel.

1983-01-01

237

Systematic acquisition of audio classes for elevator surveillance  

NASA Astrophysics Data System (ADS)

We present a systematic framework for arriving at audio classes for detection of crimes in elevators. We use a time series analysis framework to analyze the low-level features extracted from the audio of an elevator surveillance content to perform an inlier/outlier based temporal segmentation. Since suspicious events in elevators are outliers in a background of usual events, such a segmentation help bring out such events without any a priori knowledge. Then, by performing an automatic clustering on the detected outliers, we identify consistent patterns for which we can train supervised detectors. We apply the proposed framework to a collection of elevator surveillance audio data to systematically acquire audio classes such as banging, footsteps, non-neutral speech and normal speech etc. Based on the observation that the banging audio class and non-neutral speech class are indicative of suspicious events in the elevator data set, we are able to detect all of the suspicious activities without any misses.

Radhakrishnan, Regunathan; Divakaran, Ajay

2005-03-01

238

50 CFR 27.71 - Commercial filming and still photography and audio recording.  

Code of Federal Regulations, 2013 CFR

... Commercial filming and still photography and audio recording. 27...Disturbing Violations: Filming, Photography, and Light and Sound Equipment... Commercial filming and still photography and audio recording....

2013-10-01

239

Horatio Audio-Describes Shakespeare's "Hamlet": Blind and Low-Vision Theatre-Goers Evaluate an Unconventional Audio Description Strategy  

ERIC Educational Resources Information Center

Audio description (AD) has been introduced as one solution for providing people who are blind or have low vision with access to live theatre, film and television content. However, there is little research to inform the process, user preferences and presentation style. We present a study of a single live audio-described performance of Hart House…

Udo, J. P.; Acevedo, B.; Fels, D. I.

2010-01-01

240

Audio-visual biofeedback for respiratory-gated radiotherapy: Impact of audio instruction and audio-visual biofeedback on respiratory-gated radiotherapy  

SciTech Connect

Purpose: Respiratory gating is a commercially available technology for reducing the deleterious effects of motion during imaging and treatment. The efficacy of gating is dependent on the reproducibility within and between respiratory cycles during imaging and treatment. The aim of this study was to determine whether audio-visual biofeedback can improve respiratory reproducibility by decreasing residual motion and therefore increasing the accuracy of gated radiotherapy. Methods and Materials: A total of 331 respiratory traces were collected from 24 lung cancer patients. The protocol consisted of five breathing training sessions spaced about a week apart. Within each session the patients initially breathed without any instruction (free breathing), with audio instructions and with audio-visual biofeedback. Residual motion was quantified by the standard deviation of the respiratory signal within the gating window. Results: Audio-visual biofeedback significantly reduced residual motion compared with free breathing and audio instruction. Displacement-based gating has lower residual motion than phase-based gating. Little reduction in residual motion was found for duty cycles less than 30%; for duty cycles above 50% there was a sharp increase in residual motion. Conclusions: The efficiency and reproducibility of gating can be improved by: incorporating audio-visual biofeedback, using a 30-50% duty cycle, gating during exhalation, and using displacement-based gating.

George, Rohini [Department of Radiation Oncology, Virginia Commonwealth University, Richmond, VA (United States); Department of Biomedical Engineering, Virginia Commonwealth University, Richmond, VA (United States); Chung, Theodore D. [Department of Radiation Oncology, Virginia Commonwealth University, Richmond, VA (United States); Vedam, Sastry S. [Department of Radiation Oncology, Virginia Commonwealth University, Richmond, VA (United States); Ramakrishnan, Viswanathan [Department of Biostatistics, Virginia Commonwealth University, Richmond, VA (United States); Mohan, Radhe [Department of Radiation Physics, University of Texas M.D. Anderson Cancer Center, Houston, TX (United States); Weiss, Elisabeth [Department of Radiation Oncology, Virginia Commonwealth University, Richmond, VA (United States); Department of Radiation Oncology, Georg-August-Universitaet, Goettingen (Germany); Keall, Paul J. [Department of Radiation Oncology, Virginia Commonwealth University, Richmond, VA (United States)]. E-mail: pjkeall@vcu.edu

2006-07-01

241

Robust audio-visual speech recognition under noisy audio-video conditions.  

PubMed

This paper presents the maximum weighted stream posterior (MWSP) model as a robust and efficient stream integration method for audio-visual speech recognition in environments, where the audio or video streams may be subjected to unknown and time-varying corruption. A significant advantage of MWSP is that it does not require any specific measurements of the signal in either stream to calculate appropriate stream weights during recognition, and as such it is modality-independent. This also means that MWSP complements and can be used alongside many of the other approaches that have been proposed in the literature for this problem. For evaluation we used the large XM2VTS database for speaker-independent audio-visual speech recognition. The extensive tests include both clean and corrupted utterances with corruption added in either/both the video and audio streams using a variety of types (e.g., MPEG-4 video compression) and levels of noise. The experiments show that this approach gives excellent performance in comparison to another well-known dynamic stream weighting approach and also compared to any fixed-weighted integration approach in both clean conditions or when noise is added to either stream. Furthermore, our experiments show that the MWSP approach dynamically selects suitable integration weights on a frame-by-frame basis according to the level of noise in the streams and also according to the naturally fluctuating relative reliability of the modalities even in clean conditions. The MWSP approach is shown to maintain robust recognition performance in all tested conditions, while requiring no prior knowledge about the type or level of noise. PMID:23757540

Stewart, Darryl; Seymour, Rowan; Pass, Adrian; Ming, Ji

2014-02-01

242

Is audio useful in immersive visualization?  

NASA Astrophysics Data System (ADS)

In this article I provide from localization experiments in virtual environment. I define common tasks (orientation, localization, and navigation) in immersive visualization. The above mentioned tasks will be examined with user tests. Two localization experiments have been accomplished. In the first localization experiment the localization accuracy was significantly better (p<<0.01 in ANOVA) with loudspeaker reproduction than with headphone reproduction (nonindividualized HRTF's). The second experiment indicated, that localization accuracy is depending on signal (P<<0.01 in ANOVA). Although the absolute lower limit for auditory localization accuracy in front is one degree for the azimuth, the reality is much worse. For example the screens and room reverberation deteriorate loudspeaker reproduction accuracy. Current results suggest that at least in some tasks the audio is useful addition to immersive visualization tasks.

Groehn, Matti

2002-05-01

243

The Fields Institute: Lecture Audio and Slides  

NSDL National Science Digital Library

The Fields Institute for Research in Mathematical Sciences aims to "enhance mathematical activity in Canada by bringing together mathematicians from Canada and abroad, and by promoting contact and collaboration between professional mathematicians and the increasing numbers of users of mathematics." They support research in pure and applied mathematics, statistics and computer science, as well as collaborative projects between mathematicians and those applying mathematics in areas such as engineering, the physical and biological sciences, medicine, economics and finance, telecommunications and information systems. They offer this website with audio files and slides from events and lectures at the Fields Institute. The lectures, given by scientists from around the world, address such topics as Quantitative Finance, String Theory, Homological Algebra, Combinatorics, and much more. The files are organized by academic year and series title. In cases where the files are not available to download, they provide information on how to obtain the files.

244

A direct broadcast satellite-audio experiment  

NASA Technical Reports Server (NTRS)

System studies have been carried out over the past three years at the Jet Propulsion Laboratory (JPL) on digital audio broadcasting (DAB) via satellite. The thrust of the work to date has been on designing power and bandwidth efficient systems capable of providing reliable service to fixed, mobile, and portable radios. It is very difficult to predict performance in an environment which produces random periods of signal blockage, such as encountered in mobile reception where a vehicle can quickly move from one type of terrain to another. For this reason, some signal blockage mitigation techniques were built into an experimental DAB system and a satellite experiment was conducted to obtain both qualitative and quantitative measures of performance in a range of reception environments. This paper presents results from the experiment and some conclusions on the effectiveness of these blockage mitigation techniques.

Vaisnys, Arvydas; Abbe, Brian; Motamedi, Masoud

1992-01-01

245

Philadelphia Museum of Art: Audio Tours  

NSDL National Science Digital Library

Going to the Philadelphia Museum of Art and wandering around can be a great experience. But what if there were also some audio podcasts to enhance this experience? This site provides visitors access to short podcasts that can be used while in the museum, or just while sitting in front of one's computer screen. The podcasts are organized into thematic categories that include "Arms and Armor", "Modern and Contemporary Art", and "Constantine Tapestries". Many of the podcasts include digitized images of the object in question, along with information about its provenance and country of origin. It's easy to see how an assemblage of these podcasts could be organized for use by an art history class or someone who's just developing an interest about a certain aspect of art.

246

Noise-Canceling Helmet Audio System  

NASA Technical Reports Server (NTRS)

A prototype helmet audio system has been developed to improve voice communication for the wearer in a noisy environment. The system was originally intended to be used in a space suit, wherein noise generated by airflow of the spacesuit life-support system can make it difficult for remote listeners to understand the astronaut s speech and can interfere with the astronaut s attempt to issue vocal commands to a voice-controlled robot. The system could be adapted to terrestrial use in helmets of protective suits that are typically worn in noisy settings: examples include biohazard, fire, rescue, and diving suits. The system (see figure) includes an array of microphones and small loudspeakers mounted at fixed positions in a helmet, amplifiers and signal-routing circuitry, and a commercial digital signal processor (DSP). Notwithstanding the fixed positions of the microphones and loudspeakers, the system can accommodate itself to any normal motion of the wearer s head within the helmet. The system operates in conjunction with a radio transceiver. An audio signal arriving via the transceiver intended to be heard by the wearer is adjusted in volume and otherwise conditioned and sent to the loudspeakers. The wearer s speech is collected by the microphones, the outputs of which are logically combined (phased) so as to form a microphone- array directional sensitivity pattern that discriminates in favor of sounds coming from vicinity of the wearer s mouth and against sounds coming from elsewhere. In the DSP, digitized samples of the microphone outputs are processed to filter out airflow noise and to eliminate feedback from the loudspeakers to the microphones. The resulting conditioned version of the wearer s speech signal is sent to the transceiver.

Seibert, Marc A.; Culotta, Anthony J.

2007-01-01

247

Culture Machine  

NSDL National Science Digital Library

Culture Machine is a new, refereed, electronic journal encompassing cultural studies and cultural theory. The international editorial board of the interactive journal aims to "generat[e] research in culture and theory" by promoting and publishing "the most provocative of new work." The theme of the inaugural issue is Taking Risks with the Future. Content includes articles such as Life After Death of the Text by Johan Fornas, Cultural Studies in the Clouds: Mourning for Detail by Tadeusz Slawek, and The Future States of Politics by Kenneth Surin. Culture Machine is hosted by the University of Teesside, England.

248

MPEG-4 low-delay general audio coding  

NASA Astrophysics Data System (ADS)

Traditionally, speech coding for communication purposes and perceptual audio coding have been separate worlds. On one hand, speech coders provide acceptable speech quality at very low data rates and low delays which are suitable for two-way communication applications, such as Voice over IP (VoIP) or teleconferencing. Due to the underlying coding paradigm, however, such coders do not perform well for non-speech signals (e.g.~music and environmental noise). Furthermore, the sound quality and naturalness is severely limited by the fact that most coders are working in narrow-band mode, i.e. with a bandwidth below 4 kHz. On the other hand, perceptual audio codecs provide excellent subjective audio quality for a broad range of signals including speech at bit rates down to 16 kbit/s. The delay of such a coder/decoder chain, however, usually exceeds 200 ms at very low data rates and in this way is not acceptable for interactive two-way communication. This paper describes a coding scheme which is designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. The codec was standardized within MPEG-4 Version 2 Audio under the work item ``Low Delay Audio Coding'' and is derived from the ISO/MPEG-2/4 Advanced Audio Coding (AAC) algorithm. The algorithm provides modes operating at algorithmic delay as low as 20 ms and is equipped to handle all full-bandwidth high-quality audio signals, both in monophonic, stereophonic and even multi-channel format. Despite of the low algorithmic delay, the codec delivers better audio quality than MPEG-1 Layer-3 (MP3) at the same bit rate. The paper also addresses issues pertaining to the integration of the coder into H.32x and SDP applications.

Sporer, Thomas; Grill, Bernhard; Herre, Juergen

2001-07-01

249

Robust Detection Algorithm for Spread Spectrum Audio Watermarking  

NASA Astrophysics Data System (ADS)

In this letter we propose a robust detection algorithm for audio watermarking for copyright protection. The watermark is embedded in the time domain of an audio signal by the normally used spread spectrum technique. The scheme of detection is an improvement of the conventional correlation detector. A high-pass filter is applied along with the linear prediction error filter for whitening the audio signal and an adaptive threshold is chosen for decision comparing. Experimental results show that our detection algorithm outperforms the conventional one not only because it improves the robustness to normal attacks but also because it can provide the robustness to time-invariant pitch-scale modification.

Li, Lili; Fang, Xiangzhong

250

Methods and apparatuses for recording and playing back audio signals  

US Patent & Trademark Office Database

Methods and apparatus for recreating audio signals to sound as though the signals had been recorded in a different acoustic environment is provided. The methods and apparatus may include one or more inputs that receive an audio signal and an input that receives a selected acoustic environment signal, as well as processing circuitry that produces one or more output signals representative of the audio signal being played in the selected acoustic environment. The input, output and characterization signals may be processed and recorded to storage media, either individually or together. The circuitry may interface with other technology and circuitry or may be a complete stand-alone system.

2007-02-27

251

Wacky Machines  

ERIC Educational Resources Information Center

Collectors everywhere know that local antique shops and flea markets are treasure troves just waiting to be plundered. Science teachers might take a hint from these hobbyists, for the next community yard sale might be a repository of old, quirky items that are just the things to get students thinking about simple machines. By introducing some…

Fendrich, Jean

2002-01-01

252

Simple Machines  

NSDL National Science Digital Library

A webquest about simple machines Please open microsoft word and re-type or copy and paste each question and then type your answer. Be sure to include your name at the top of the page!!! . . Follow the link below and click on the Start button. Go to either the house or the tool shed and go ...

Mr.rebello

2008-11-24

253

XML Machines  

NASA Astrophysics Data System (ADS)

In order to capture the dynamics of XML databases a general model of tree-based database transformations is required. In this paper such an abstract computational model is presented, which brings together ideas from Abstract State Machines and monadic second-order logic. The model captures all XML database transformations.

Wang, Qing; Ferrarotti, Flavio A.

254

Function Machine  

NSDL National Science Digital Library

This Java applet allows learners to explore simple linear functions. Students determine the algebraic form of a linear equation by entering inputs into the machine and by looking for patterns in the outputs. The function rules available are: integers from -10 to 10 are either added to, subtracted from, or multiplied by the input x to yield the output y.

2011-01-01

255

Gaussian Mixture Modeling Using Short Time Fourier Transform Features for Audio Fingerprinting  

Microsoft Academic Search

In audio fingerprinting, an audio clip must be recognized by matching an extracted fingerprint to a database of previously computed fingerprints. The fingerprints should reduce the dimensionality of the input significantly, provide discrimination among different audio clips, and at the same time, invariant to the distorted versions of the same audio clip. In this paper, we design fingerprints addressing the

Arunan Ramalingam; Sridhar Krishnan

2005-01-01

256

Advances in unsupervised audio classification and segmentation for the broadcast news and NGSW corpora  

Microsoft Academic Search

The problem of unsupervised audio classification and segmentation continues to be a challenging research problem which significantly impacts automatic speech recognition (ASR) and spoken document retrieval (SDR) performance. This paper addresses novel advances in 1) audio classification for speech recognition and 2) audio segmentation for unsupervised multispeaker change detection. A new algorithm is proposed for audio classification, which is based

Rongqing Huang; John H. L. Hansen

2006-01-01

257

Gaussian Mixture Modeling of Short-Time Fourier Transform Features for Audio Fingerprinting  

Microsoft Academic Search

In audio fingerprinting, an audio clip must be recognized by matching an extracted fingerprint to a database of previously computed fingerprints. The fingerprints should reduce the dimensionality of the input significantly, provide discrimination among different audio clips, and, at the same time, be invariant to distorted versions of the same audio clip. In this paper, we design fingerprints addressing the

Arunan Ramalingam; Sridhar Krishnan

2006-01-01

258

Drilling Machines: Vocational Machine Shop.  

ERIC Educational Resources Information Center

The lessons and supportive information in this field tested instructional block provide a guide for teachers in developing a machine shop course of study in drilling. The document is comprised of operation sheets, information sheets, and transparency masters for 23 lessons. Each lesson plan includes a performance objective, material and tools,…

Thomas, John C.

259

A compressed domain beat detector using MP3 audio bitstreams  

Microsoft Academic Search

This paper presents a novel beat detector that processes MPEG-1 Layer III (known as MP3) encoded audio bitstreams directly in the compressed domain. Most previous beat detection or tracking systems dealing with MIDI or PCM signals are not directly applicable to compressed audio bitstreams, such as MP3 bitstreams. We have developed the beat detector as a part of a beat-pattern

Ye Wang; Miikka Vilermo

2001-01-01

260

AN ASYNCHRONOUS DBN FOR AUDIO-VISUAL SPEECH RECOGNITION  

Microsoft Academic Search

We investigate an asynchronous two-stream dynamic Bayesian network-based model for audio-visual speech recognition. The model allows the audio and visual streams to de-synchronize within the boundaries of each word. The probability of de-synchronization by a given number of states is learned during training. This type of asynchrony has been previously used for pronunciation modeling and for visual speech recognition (lipreading);

Kate Saenko; Karen Livescu

2006-01-01

261

Audio watermarking method robust against time- and frequency-fluctuation  

Microsoft Academic Search

In this paper, we describe an audio watermarking algorithm that can embed a multiple-bit message which is robust against wow-and-flutter, cropping, noise-addition, pitch-shift, and audio compressions such as MP3. The algorithm calculates and manipulates the magnitudes of segmented areas in the time-frequency plane of the content using short-term DFTs. The detection algorithm correlates the magnitudes with a pseudo-random array that

Ryuki Tachibana; Shuichi Shimizu; Taiga Nakamura; Seiji Kobayashi

2001-01-01

262

Robust Audio Watermarking Based on Low-Order Zernike Moments  

Microsoft Academic Search

Extensive testing shows that the audio Zernike moments in lower orders are very robust to common signal processing operations, such as MP3 compression, low-pass flltering, etc. Based on the observations, in this paper, a robust watermark scheme is proposed by embedding the bits into the low-order moments. By analyzing and deducting the linear relationship between the audio amplitude and moments,

Shijun Xiang; Jiwu Huang; Rui Yang; Chuntao Wang; Hongmei Liu

2006-01-01

263

Structural and Semantic Modeling of Audio for Content-Based Querying and Browsing  

Microsoft Academic Search

\\u000a A typical content-based audio management system deals with three aspects namely audio segmentation and classification, audio\\u000a analysis, and content-based retrieval of audio. In this paper, we integrate the three aspects of content-based audio management\\u000a into a single framework and propose an efficient method for flexible querying and browsing of auditory data. More specifically,\\u000a we utilize two robust feature sets namely

Mustafa Sert; Buyurman Baykal; Adnan Yazici

2006-01-01

264

Robust Audio Watermarking Based on Log-Polar Frequency Index  

NASA Astrophysics Data System (ADS)

In this paper, we analyze the audio signal distortions introduced by pitch-scaling, random cropping and DA/AD conversion, and find a robust feature, average Fourier magnitude over the log-polar frequency index(AFM), which can resist these attacks. Theoretical analysis and extensive experiments demonstrate that AFM is an appropriate embedding region for robust audio watermarking. This is the first work on applying log-polar mapping to audio watermark. The usage of log-polar mapping in our work is basically different from the existing works in image watermarking. The log-polar mapping is only applied to the frequency index, not to the transform coefficients, which avoids the reconstruction distortion of inverse log-polar transform and reduces the computation cost. Comparison with the existing methods, the proposed AFM-based watermarking scheme has the outstanding performance on resisting pitch-scaling and random cropping, as well as very approving robustness to DA/AD conversion and TSM (Time-Scale Modification). The watermarked audio achieves high auditory quality. Experimental results show that the scheme is very robust to common audio signal processing and distortions introduced in Stirmark for Audio.

Yang, Rui; Kang, Xiangui; Huang, Jiwu

265

Minimally radiating sources for personal audio.  

PubMed

In order to reduce annoyance from the audio output of personal devices, it is necessary to maintain the sound level at the user position while minimizing the levels elsewhere. If the dark zone, within which the sound is to be minimized, extends over the whole far field of the source, the problem reduces to that of minimizing the radiated sound power while maintaining the pressure level at the user position. It is shown analytically that the optimum two-source array then has a hypercardioid directivity and gives about 7 dB reduction in radiated sound power, compared with a monopole producing the same on-axis pressure. The performance of other linear arrays is studied using monopole simulations for the motivating example of a mobile phone. The trade-off is investigated between the performance in reducing radiated noise, and the electrical power required to drive the array for different numbers of elements. It is shown for both simulations and experiments conducted on a small array of loudspeakers under anechoic conditions, that both two and three element arrays provide a reasonable compromise between these competing requirements. The implementation of the two-source array in a coupled enclosure is also shown to reduce the electrical power requirements. PMID:20968345

Elliott, Stephen J; Cheer, Jordan; Murfet, Harry; Holland, Keith R

2010-10-01

266

Fullerene Machines  

NASA Technical Reports Server (NTRS)

Recent computational efforts at NASA Ames Research Center and computation and experiment elsewhere suggest that a nanotechnology of machine phase functionalized fullerenes may be synthetically accessible and of great interest. We have computationally demonstrated that molecular gears fashioned from (14,0) single-walled carbon nanotubes and benzyne teeth should operate well at 50-100 gigahertz. Preliminary results suggest that these gears can be cooled by a helium atmosphere and a laser motor can power fullerene gears if a positive and negative charge have been added to form a dipole. In addition, we have unproven concepts based on experimental and computational evidence for support structures, computer control, a system architecture, a variety of components, and manufacture. Combining fullerene machines with the remarkable mechanical properties of carbon nanotubes, there is some reason to believe that a focused effort to develop fullerene nanotechnology could yield materials with tremendous properties.

Globus, Al; Saini, Subhash

1998-01-01

267

Fullerene Machines  

NASA Technical Reports Server (NTRS)

Fullerenes possess remarkable properties and many investigators have examined the mechanical, electronic and other characteristics of carbon SP2 systems in some detail. In addition, C-60 can be functionalized with many classes of molecular fragments and we may expect the caps of carbon nanotubes to have a similar chemistry. Finally, carbon nanotubes have been attached to t he end of scanning probe microscope (Spill) tips. Spills can be manipulated with sub-angstrom accuracy. Together, these investigations suggest that complex molecular machines made of fullerenes may someday be created and manipulated with very high accuracy. We have studied some such systems computationally (primarily functionalized carbon nanotube gears and computer components). If such machines can be combined appropriately, a class of materials may be created that can sense their environment, calculate a response, and act. The implications of such hypothetical materials are substantial.

Globus, Al; Saini, Subhash (Technical Monitor)

1998-01-01

268

Induction machine  

DOEpatents

A polyphase rotary induction machine for use as a motor or generator utilizing a single rotor assembly having two series connected sets of rotor windings, a first stator winding disposed around the first rotor winding and means for controlling the current induced in one set of the rotor windings compared to the current induced in the other set of the rotor windings. The rotor windings may be wound rotor windings or squirrel cage windings.

Owen, Whitney H. (Ogden, UT)

1980-01-01

269

Simple Machines  

NSDL National Science Digital Library

Can you identify the six types of simple machines? 1. What do you know about Inclined Planes? Draw an example on your graphic organizer and state one fact.Inclined Plane 2. What do you know about levers? Draw an example on your graphic organizer and state one fact.Lever. 3. What do you know about pulleys? Draw an example on your graphic organizer and ...

Stewart, Miss

2010-03-24

270

Living Machines  

Microsoft Academic Search

\\u000a Ecological studies have revealed that nature has an in-built system to restore itself, thereby sustaining its continuity.\\u000a In other words, natural ecosystems can act as “Living Machines” in keeping the ecosystems habitable. The biological communities\\u000a – microbes, plants, and animals – serve as the driving force of several living technological innovations – constructed wetlands,\\u000a Lake Restores, Eco-Restorers, and Reedbeds. These

Yung-Tse Hung; Joseph F. Hawumba; Lawrence K. Wang

271

Applying Spatial Audio to Human Interfaces: 25 Years of NASA Experience  

NASA Technical Reports Server (NTRS)

From the perspective of human factors engineering, the inclusion of spatial audio within a human-machine interface is advantageous from several perspectives. Demonstrated benefits include the ability to monitor multiple streams of speech and non-speech warning tones using a cocktail party advantage, and for aurally-guided visual search. Other potential benefits include the spatial coordination and interaction of multimodal events, and evaluation of new communication technologies and alerting systems using virtual simulation. Many of these technologies were developed at NASA Ames Research Center, beginning in 1985. This paper reviews examples and describes the advantages of spatial sound in NASA-related technologies, including space operations, aeronautics, and search and rescue. The work has involved hardware and software development as well as basic and applied research.

Begault, Durand R.; Wenzel, Elizabeth M.; Godfrey, Martine; Miller, Joel D.; Anderson, Mark R.

2010-01-01

272

TEMPO machine  

SciTech Connect

TEMPO is a transformer powered megavolt pulse generator with an output pulse of 100 ns duration. The machine was designed for burst mode operation at pulse repetition rates up to 10 Hz with minimum pulse-to-pulse voltage variations. To meet the requirement for pulse duration a nd a 20-..omega.. output impedance within reasonable size constraints, the pulse forming transmission line was designed as two parallel water-insulated, strip-type Blumleins. Stray capacitance and electric fields along the edges of the line elements were controlled by lining the tank with plastic sheet.

Rohwein, G.J.; Lancaster, K.T.; Lawson, R.N.

1986-06-01

273

An Adaptive Robust Watermarking Algorithm for Audio Signals Using SVD  

NASA Astrophysics Data System (ADS)

This paper proposes an efficient watermarking algorithm which embeds watermark data adaptively in the audio signal. The algorithm embeds the watermark in the host audio signal in such a way that the degree of embedding (DOE) is adaptive in nature and is chosen in a justified manner according to the localized content of the audio. The watermark embedding regions are selectively chosen in the high energy regions of the audio signal which make the embedding process robust to synchronization attacks. Synchronization codes are added along with the watermark in the wavelet domain and hence the embedded data can be subjected to self synchronization and the synchronization code can be used as a check to combat false alarm that results from data modification due to watermark embedding. The watermark is embedded by quantization of the singular value decompositions in the wavelet domain which makes the process perceptually transparent. The experimental results suggest that the proposed algorithm maintains a good perceptual quality of the audio signal and maintains good robustness against signal processing attacks. Comparative analysis indicates that the proposed algorithm of adaptive DOE has superior performance in comparison to existing uniform DOE.

Dutta, Malay Kishore; Pathak, Vinay K.; Gupta, Phalguni

274

Talker variability in audio-visual speech perception  

PubMed Central

A change in talker is a change in the context for the phonetic interpretation of acoustic patterns of speech. Different talkers have different mappings between acoustic patterns and phonetic categories and listeners need to adapt to these differences. Despite this complexity, listeners are adept at comprehending speech in multiple-talker contexts, albeit at a slight but measurable performance cost (e.g., slower recognition). So far, this talker variability cost has been demonstrated only in audio-only speech. Other research in single-talker contexts have shown, however, that when listeners are able to see a talker’s face, speech recognition is improved under adverse listening (e.g., noise or distortion) conditions that can increase uncertainty in the mapping between acoustic patterns and phonetic categories. Does seeing a talker’s face reduce the cost of word recognition in multiple-talker contexts? We used a speeded word-monitoring task in which listeners make quick judgments about target word recognition in single- and multiple-talker contexts. Results show faster recognition performance in single-talker conditions compared to multiple-talker conditions for both audio-only and audio-visual speech. However, recognition time in a multiple-talker context was slower in the audio-visual condition compared to audio-only condition. These results suggest that seeing a talker’s face during speech perception may slow recognition by increasing the importance of talker identification, signaling to the listener a change in talker has occurred.

Heald, Shannon L. M.; Nusbaum, Howard C.

2014-01-01

275

Parallel machines: Parallel machine languages  

SciTech Connect

This book presents a framework for understanding the tradeoffs between the conventional view and the dataflow view with the objective of discovering the critical hardware structures which must be present in any scalable, general-purpose parallel computer to effectively tolerate latency and synchronization costs. The author presents an approach to scalable general purpose parallel computation. Linguistic Concerns, Compiling Issues, Intermediate Language Issues, and hardware/technological constraints are presented as a combined approach to architectural Develoement. This book presents the notion of a parallel machine language.

Iannucci, R.A. (IBM (US))

1990-01-01

276

Mind & Machine  

NSDL National Science Digital Library

Mind & Machine is a weekly column provided by Ashley Dunn for the New York Times Cybertimes that discusses topics related to computing, technology, and the Internet. Recent columns have addressed the topics of the development of Internet telephony, possible futures of user interfaces, the history of technology and standards, and the Internet as a vehicle for community. Articles are well written, opinionated, and thought provoking. Mr. Dunn is a free lance writer who has written for such papers as the New York Times, the Los Angeles Times, the Seattle Post-Intelligencer, and the South China Morning Post. Note that the site is available only upon registration and is free of charge only in the US.

Dunn, Ashley.

1996-01-01

277

Multi-channel spatialization system for audio signals  

NASA Technical Reports Server (NTRS)

Synthetic head related transfer functions (HRTF's) for imposing reprogramable spatial cues to a plurality of audio input signals included, for example, in multiple narrow-band audio communications signals received simultaneously are generated and stored in interchangeable programmable read only memories (PROM's) which store both head related transfer function impulse response data and source positional information for a plurality of desired virtual source locations. The analog inputs of the audio signals are filtered and converted to digital signals from which synthetic head related transfer functions are generated in the form of linear phase finite impulse response filters. The outputs of the impulse response filters are subsequently reconverted to analog signals, filtered, mixed and fed to a pair of headphones.

Begault, Durand R. (inventor)

1995-01-01

278

Multi-channel spatialization systems for audio signals  

NASA Technical Reports Server (NTRS)

Synthetic head related transfer functions (HRTF's) for imposing reprogrammable spatial cues to a plurality of audio input signals included, for example, in multiple narrow-band audio communications signals received simultaneously are generated and stored in interchangeable programmable read only memories (PROM's) which store both head related transfer function impulse response data and source positional information for a plurality of desired virtual source locations. The analog inputs of the audio signals are filtered and converted to digital signals from which synthetic head related transfer functions are generated in the form of linear phase finite impulse response filters. The outputs of the impulse response filters are subsequently reconverted to analog signals, filtered, mixed, and fed to a pair of headphones.

Begault, Durand R. (inventor)

1993-01-01

279

Say What? The Role of Audio in Multimedia Video  

NASA Astrophysics Data System (ADS)

Audio, including interviews, ambient sounds, and music, is a critical-yet often overlooked-part of an effective multimedia video. In February 2010, Linder joined scientists working on the Global Rivers Observatory Project for two weeks of intensive fieldwork in the Congo River watershed. The team's goal was to learn more about how climate change and deforestation are impacting the river system and coastal ocean. Using stills and video shot with a lightweight digital SLR outfit and audio recorded with a pocket-sized sound recorder, Linder documented the trials and triumphs of working in the heart of Africa. Using excerpts from the six-minute Congo multimedia video, this presentation will illustrate how to record and edit an engaging audio track. Topics include interview technique, collecting ambient sounds, choosing and using music, and editing it all together to educate and entertain the viewer.

Linder, C. A.; Holmes, R. M.

2011-12-01

280

Music Identification System Using MPEG-7 Audio Signature Descriptors  

PubMed Central

This paper describes a multiresolution system based on MPEG-7 audio signature descriptors for music identification. Such an identification system may be used to detect illegally copied music circulated over the Internet. In the proposed system, low-resolution descriptors are used to search likely candidates, and then full-resolution descriptors are used to identify the unknown (query) audio. With this arrangement, the proposed system achieves both high speed and high accuracy. To deal with the problem that a piece of query audio may not be inside the system's database, we suggest two different methods to find the decision threshold. Simulation results show that the proposed method II can achieve an accuracy of 99.4% for query inputs both inside and outside the database. Overall, it is highly possible to use the proposed system for copyright control.

You, Shingchern D.; Chen, Wei-Hwa; Chen, Woei-Kae

2013-01-01

281

Virtual environment display for a 3D audio room simulation  

NASA Technical Reports Server (NTRS)

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.; Foster, Scott H.

1992-01-01

282

Multi-channel spatialization systems for audio signals  

NASA Astrophysics Data System (ADS)

Synthetic head related transfer functions (HRTF's) for imposing reprogrammable spatial cues to a plurality of audio input signals included, for example, in multiple narrow-band audio communications signals received simultaneously are generated and stored in interchangeable programmable read only memories (PROM's) which store both head related transfer function impulse response data and source positional information for a plurality of desired virtual source locations. The analog inputs of the audio signals are filtered and converted to digital signals from which synthetic head related transfer functions are generated in the form of linear phase finite impulse response filters. The outputs of the impulse response filters are subsequently reconverted to analog signals, filtered, mixed, and fed to a pair of headphones.

Begault, Durand R.

1993-10-01

283

Multi-channel spatialization system for audio signals  

NASA Astrophysics Data System (ADS)

Synthetic head related transfer functions (HRTF's) for imposing reprogramable spatial cues to a plurality of audio input signals included, for example, in multiple narrow-band audio communications signals received simultaneously are generated and stored in interchangeable programmable read only memories (PROM's) which store both head related transfer function impulse response data and source positional information for a plurality of desired virtual source locations. The analog inputs of the audio signals are filtered and converted to digital signals from which synthetic head related transfer functions are generated in the form of linear phase finite impulse response filters. The outputs of the impulse response filters are subsequently reconverted to analog signals, filtered, mixed and fed to a pair of headphones.

Begault, Durand R.

1995-08-01

284

Highlight summarization in golf videos using audio signals  

NASA Astrophysics Data System (ADS)

In this paper, we present an automatic summarization of highlights in golf videos based on audio information alone without video information. The proposed highlight summarization system is carried out based on semantic audio segmentation and detection on action units from audio signals. Studio speech, field speech, music, and applause are segmented by means of sound classification. Swing is detected by the methods of impulse onset detection. Sounds like swing and applause form a complete action unit, while studio speech and music parts are used to anchor the program structure. With the advantage of highly precise detection of applause, highlights are extracted effectively. Our experimental results obtain high classification precision on 18 golf games. It proves that the proposed system is very effective and computationally efficient to apply the technology to embedded consumer electronic devices.

Kim, Hyoung-Gook; Kim, Jin Young

2008-01-01

285

Influence of audio triggered emotional attention on video perception  

NASA Astrophysics Data System (ADS)

Perceptual video coding methods attempt to improve compression efficiency by discarding visual information not perceived by end users. Most of the current approaches for perceptual video coding only use visual features ignoring the auditory component. Many psychophysical studies have demonstrated that auditory stimuli affects our visual perception. In this paper we present our study of audio triggered emotional attention and it's applicability to perceptual video coding. Experiments with movie clips show that the reaction time to detect video compression artifacts was longer when video was presented with the audio information. The results reported are statistically significant with p=0.024.

Torres, Freddy; Kalva, Hari

2014-02-01

286

Animation, audio, and spatial ability: Optimizing multimedia for scientific explanations  

NASA Astrophysics Data System (ADS)

This study investigated the effects of audio, animation and spatial ability in a computer based instructional program for biology. The program presented instructional material via text or audio with lean text and included eight instructional sequences presented either via static illustrations or animations. High school students enrolled in a biology course were blocked by spatial ability and randomly assigned to one of four treatments (Text-Static Illustration Audio-Static Illustration, Text-Animation, Audio-Animation). The study examined the effects of instructional mode (Text vs. Audio), illustration mode (Static Illustration vs. Animation) and spatial ability (Low vs. High) on practice and posttest achievement, attitude and time. Results for practice achievement indicated that high spatial ability participants achieved more than low spatial ability participants. Similar results for posttest achievement and spatial ability were not found. Participants in the Static Illustration treatments achieved the same as participants in the Animation treatments on both the practice and posttest. Likewise, participants in the Text treatments achieved the same as participants in the Audio treatments on both the practice and posttest. In terms of attitude, participants responded favorably to the computer based instructional program. They found the program interesting, felt the static illustrations or animations made the explanations easier to understand and concentrated on learning the material. Furthermore, participants in the Animation treatments felt the information was easier to understand than participants in the Static Illustration treatments. However, no difference for any attitude item was found for participants in the Text as compared to those in the Audio treatments. Significant differences were found by Spatial Ability for three attitude items concerning concentration and interest. In all three items, the low spatial ability participants responded more positively than high spatial ability participants. In addition, low spatial ability participants reported greater mental effort than high spatial ability participants. Findings for time-in-program and time-in-instruction indicated that participants in the Animation treatments took significantly more time than participants in the Static Illustration treatments. No time differences of any type were found for participants in the Text versus Audio treatments. Implications for the design of multimedia instruction and topics for future research are included in the discussion.

Koroghlanian, Carol May

287

Temporal Asynchronicity Modeling by Product HMMS for Audio-Visual Speech Recognition.  

National Technical Information Service (NTIS)

There have been higher demands recently for Automatic Speech Recognition (ASR) systems able to operate robustly in acoustically noisy environments. This paper proposes a method to effectively integrate audio and visual information in audio-visual (bi-moda...

S. Nakamura

2002-01-01

288

76 FR 591 - Determination of Rates and Terms for Preexisting Subscription and Satellite Digital Audio Radio...  

Federal Register 2010, 2011, 2012, 2013

...Docket No. 2011-1 CRB PSS/Satellite II] Determination of Rates and Terms for Preexisting Subscription and Satellite Digital Audio Radio Services AGENCY...for preexisting subscription and satellite digital audio radio services...

2011-01-05

289

Study of the Practical Real-Time Implementation of High-Performance Text-to-Speech Translation Based on the MIT (Allen/Klat) Rule Programs.  

National Technical Information Service (NTIS)

Development of a system capable of generating high quality speech from full-word English text in real time, is reported. This system is based upon rules comprising the MIT Modular Speech System (MSS) using a general-purpose minicomputer and special hardwa...

J. L. Caldwell

1979-01-01

290

Iowa Virtual Literacy Protocol: A Pre-Experimental Design Using Kurzweil 3000 Text-to-Speech Software with Incarcerated Adult Learners  

ERIC Educational Resources Information Center

The problem: The increasingly competitive global economy demands literate, educated workers. Both men and women experience the effects of education on employment rates and income. Racial and ethnic minorities, English language learners, and especially those with prison records are most deeply affected by the economic consequences of dropping out…

McCulley, Yvette K.

2012-01-01

291

Design and VLSI implementation of a digital audio-specific DSP core for MP3\\/AAC  

Microsoft Academic Search

We present a digital audio-specific DSP core designed for a dual decoder for MPEG\\/audio layer-3 (MP3) and MPEG-2 advanced audio coding (AAC). The processing core is a 20-bit fixed-point programmable DSP having an architecture suitable for audio signal processing. It supports special instructions such as UNPACK and Huffman as well as general arithmetic and logical instructions including pipelined-MAC. All instructions

Kyoung Ho Bang; Nam Hun Jeong; Joon Seok Kim; Young Cheol Park; Dae Hee Youn

2002-01-01

292

Machine musicianship  

NASA Astrophysics Data System (ADS)

The training of musicians begins by teaching basic musical concepts, a collection of knowledge commonly known as musicianship. Computer programs designed to implement musical skills (e.g., to make sense of what they hear, perform music expressively, or compose convincing pieces) can similarly benefit from access to a fundamental level of musicianship. Recent research in music cognition, artificial intelligence, and music theory has produced a repertoire of techniques that can make the behavior of computer programs more musical. Many of these were presented in a recently published book/CD-ROM entitled Machine Musicianship. For use in interactive music systems, we are interested in those which are fast enough to run in real time and that need only make reference to the material as it appears in sequence. This talk will review several applications that are able to identify the tonal center of musical material during performance. Beyond this specific task, the design of real-time algorithmic listening through the concurrent operation of several connected analyzers is examined. The presentation includes discussion of a library of C++ objects that can be combined to perform interactive listening and a demonstration of their capability.

Rowe, Robert

2002-05-01

293

Individual audio channels with single display groupware: effects on communication and task strategy  

Microsoft Academic Search

We introduce a system that allows four users to each receive sound from a private audio channel while using a shared tabletop display. In order to explore how private audio channels affect a collaborative work environment, we conducted a user study with this system. The results reveal differences in work strategies when groups are presented with individual versus public audio,

Meredith Ringel Morris; Dan Morris; Terry Winograd

2004-01-01

294

A flexible framework for key audio effects detection and auditory context inference  

Microsoft Academic Search

Key audio effects are those special effects that play critical roles in human's perception of an auditory context in audiovisual materials. Based on key audio effects, high-level semantic inference can be carried out to facilitate various con- tent-based analysis applications, such as highlight extraction and video summarization. In this paper, a flexible framework is pro- posed for key audio effect

Rui Cai; Lie Lu; Alan Hanjalic; Hong-Jiang Zhang; Lian-hong Cai

2006-01-01

295

A probabilistic principal component analysis based hidden Markov model for audio-visual speech recognition  

Microsoft Academic Search

Lipreading is an efficient method among those proposed to improve the performance of speech recognition systems, especially in acoustic noisy environments. This paper proposes a simple audio-visual speech recognition (AVSR) system, which could improve the robustness and accuracy of audio speech recognition by integrating the synchronous audio and visual information. We propose a hidden Markov model (HMM) based on the

Zhanyu Ma; Arne Leijon

2008-01-01

296

Audio-visual continuous speech recognition using a coupled hidden Markov model  

Microsoft Academic Search

With the increase in the computational complexity of recent computers, audio-visual speech recognition (AVSR) became an attractive research topic that can lead to a robust solution for speech recognition in noisy environments. In the audio visual continuous speech recognition system presented in this paper, the audio and visual observation sequences are integrated using a coupled hidden Markov model (CHMM). The

Xiaoxing Liu; Yibao Zhao; Xiaobo Pi; Luhong Liang; Ara V. Nefian

2002-01-01

297

Design and Usability Testing of an Audio Platform Game for Players with Visual Impairments  

ERIC Educational Resources Information Center

This article reports on the evaluation of a novel audio platform game that creates a spatial, interactive experience via audio cues. A pilot study with players with visual impairments, and usability testing comparing the visual and audio game versions using both sighted players and players with visual impairments, revealed that all the…

Oren, Michael; Harding, Chris; Bonebright, Terri L.

2008-01-01

298

DEVELOPING CROSS-PLATFORM AUDIO AND MUSIC APPLICATIONS WITH THE CLAM FRAMEWORK  

Microsoft Academic Search

CLAM is a C++ framework that offers a complete devel- opment and research platform for the audio and music do- main. Apart from offering an abstract model for audio systems, it also includes a repository of processing algo- rithms and data types as well as a number of tools such as audio or MIDI input\\/output. All these features can be

Xavier Amatriain; Pau Arum ´ i

299

The Case for FEC-based Error Control for Packet Audio in the Internet  

Microsoft Academic Search

We consider the problem of distributing real-time packet audio overnetworks such as the Internet which do not provide support for real-timeapplications. Experiments with such networks indicate that audio qualityis mediocre in large part because of excessive audio packet losses. In thispaper, we show using measurements over the Internet as well as analyticmodeling that most loss periods involve a small number

Andrs Vega-garca; Jean-chrysostome Bolot

1997-01-01

300

High quality digital audio experiments for a signal processing first course  

Microsoft Academic Search

In this paper we provide the details of a final project in audio signal processing, used in a signal processing first course taught to second year ECE students. High quality audio signal processing covers a range of applications from the musical recording studio, musical performance, home theatre, home stereo, car stereo system, or even a personal audio player. The design\\/build\\/test

Mark A. Wickert

2011-01-01

301

Exploring spatial audio conferencing functionality in multiuser virtual environments (poster session)  

Microsoft Academic Search

A chatspace was developed that allows conversation with 3D sound using networked streaming in a shared virtual environment. The system provides an interface to advanced audio features, such as a “whisper function” for conveying a confided audio stream. This study explores the use of spatial audio to enhance a user's experience in multiuser virtual environments.

Yasuhiro Yamazaki; Jens Herder

2000-01-01

302

Hearing You Loud and Clear: Student Perspectives of Audio Feedback in Higher Education  

ERIC Educational Resources Information Center

The use of audio feedback for students in a full-time community nursing degree course is appraised. The aim of this mixed methods study was to examine student views on audio feedback for written assignments. Questionnaires and a focus group were used to capture student opinion of this pilot project. The majority of students valued audio feedback…

Gould, Jill; Day, Pat

2013-01-01

303

A phone-viseme dynamic Bayesian network for audio-visual automatic speech recognition  

Microsoft Academic Search

This work extends and improves a recently intro- duced (Dec. 2007) dynamic Bayesian network (DBN) based audio-visual automatic speech recognition (AV- ASR) system. That system models the audio and visual components of speech as being composed of the same sub-word units when, in fact, this is not psycholinguisti- cally true. We extend the system to model the audio and visual

Louis H. Terry; Aggelos K. Katsaggelos

2008-01-01

304

Rich System Combination For Keyword Spotting In Noisy and Acoustically Heterogeneous Audio Streams.  

National Technical Information Service (NTIS)

We address the problem of retrieving spoken information from noisy and heterogeneous audio archives using a rich system combination with a diverse set of noise-robust modules and audio characteriza- tion. Audio search applications so far have focused on c...

J. van Hout L. Burget M. Akbacak W. Wang

2013-01-01

305

Hearing you loud and clear: student perspectives of audio feedback in higher education  

Microsoft Academic Search

The use of audio feedback for students in a full-time community nursing degree course is appraised. The aim of this mixed methods study was to examine student views on audio feedback for written assignments. Questionnaires and a focus group were used to capture student opinion of this pilot project. The majority of students valued audio feedback as more detailed, personalised

Jill Gould; Pat Day

2012-01-01

306

Development and Assessment of Web Courses That Use Streaming Audio and Video Technologies.  

ERIC Educational Resources Information Center

Iowa State University, through a program called Project BIO (Biology Instructional Outreach), has been using RealAudio technology for about 2 years in college biology courses that are offered entirely via the World Wide Web. RealAudio is a type of streaming media technology that can be used to deliver audio content and a variety of other media…

Ingebritsen, Thomas S.; Flickinger, Kathleen

307

Combined video and audio watermarking: embedding content information in multimedia data  

Microsoft Academic Search

Audio and video watermarking enable the copyright protection with owner or customer authentication and the detection of media manipulations. The available watermarking technology concentrates on single media like audio or video. But the typical multimedia stream consists of both video and audio data. Our goal is to provide a solution with robust and fragile aspects to guarantee authentication and integrity

Jana Dittmann; Martin Steinebach; Ivica Rimac; Stephan Fischer; Ralf Steinmetz

2000-01-01

308

Content based audio classification and retrieval using joint time-frequency analysis  

Microsoft Academic Search

We present an audio classification and retrieval technique that exploits the non-stationary behavior of music signals and extracts features that characterize their spectral change over time. Audio classification provides a solution to incorrect and inefficient manual labelling of audio files on computers by allowing users to extract music files based on content similarity rather than labels. In our technique, classification

S. Esmaili; S. Krishnan; K. Raahemifar

2004-01-01

309

Integrating Additional Chord Information Into HMM-Based Lyrics-to-Audio Alignment  

Microsoft Academic Search

Aligning lyrics to audio has a wide range of ap- plications such as the automatic generation of karaoke scores, song-browsing by lyrics, and the generation of audio thumbnails. Existing methods are restricted to using only lyrics and match them to phoneme features extracted from the audio (usually mel-frequency cepstral coefficients). Our novel idea is to integrate the textual chord information

Matthias Mauch; Hiromasa Fujihara; Masataka Goto

2012-01-01

310

An Improved Psychoacoustic Model for Audio Coding Based on Wavelet Packet  

Microsoft Academic Search

This paper describes a new design of a psychoacoustic model for audio coding following the model used in the standard MPEG-1 audio layer 3 using an appropriate wavelet packet decomposition of the speech\\/audio signal. The design of a psychoacoustic model is achieved by wavelet packet decomposition whose connections are selected in such a way that sub bands correspond to the

Samar Krimi; Kaïs Ouni; Noureddine Ellouze

2007-01-01

311

Low-delay predictive audio coding for the HIVITS HDTV codec  

NASA Astrophysics Data System (ADS)

The status of work relating to predictive audio coding, as part of the European project on High Quality Video Telephone and HD(TV) Systems (HIVITS), is reported. The predictive coding algorithm is developed, along with six-channel audio coding and decoding hardware. Demonstrations of the audio codec operating in conjunction with the video codec, are given.

McParland, A. K.; Gilchrist, N. H. C.

1995-01-01

312

Business Information Audio Cassettes: Their Care and Feeding  

ERIC Educational Resources Information Center

Audio cassettes raise new types of problems for special librarians. Confronted with sparse information about the media, the author, through trial and error, eventually produced basic guidelines and criteria for bibliographic control, storage, circulation, and material sources. These are described and workable solutions are presented. (5…

Noble, Valerie

1973-01-01

313

The Effect of Features on Clustering in Audio Surveillance  

Microsoft Academic Search

The effect of the choice of features on unsupervised clustering in audio surveillance is investigated. The importance of individual features in a larger feature set is first analyzed by examining the component loadings in principal component analysis (PCA). The individual sound events are then assigned into clusters using the self-tuning spectral clustering and the classical K-means algorithms. A weighted version

Sampo Vesa

314

Infant Perception of Audio-Visual Speech Synchrony  

ERIC Educational Resources Information Center

Three experiments investigated perception of audio-visual (A-V) speech synchrony in 4- to 10-month-old infants. Experiments 1 and 2 used a convergent-operations approach by habituating infants to an audiovisually synchronous syllable (Experiment 1) and then testing for detection of increasing degrees of A-V asynchrony (366, 500, and 666 ms) or by…

Lewkowicz, David J.

2010-01-01

315

HRIR~: modulating range in headphone-reproduced spatial audio  

Microsoft Academic Search

HRIR~, a new software audio filter for Head-Related Impulse Response (HRIR) convolution is presented. The filter, implemented as a Pure-Data object, allows dynamic modification of a sound source's apparent location by modulating its virtual azimuth, elevation, and range in realtime, the last attribute being missing in surveyed similar applications. With hrir~ users can virtually localize monophonic sources around a listener's

Julián Villegas; Michael Cohen

2010-01-01

316

The Use of Audio in Computer-Based Instruction.  

ERIC Educational Resources Information Center

This study investigated the effects of audio and text density on the achievement, time-in-program, and attitudes of 134 undergraduates. Data concerning the subjects' preexisting computer skills and experience, as well as demographic information, were also collected. The instruction in visual design principles was delivered by computer and included…

Koroghlanian, Carol M.; Sullivan, Howard J.

317

Audio-based radio and TV broadcast monitoring  

Microsoft Academic Search

This paper describes a scalable real-time audio fingerprinting system developed by IBOPE Midia for radio and TV broadcast monitoring. A special temporal feature extraction strategy based on the Short-Time Fourier Transform has been designed. When given an input stream to analyse, the system matches it ag ainst the database and automatically recognizes instances of the previously registered samples within the

Bruno Oliveira; Alexandre Crivellaro; Roberto Marcondes Cesar Junior

2006-01-01

318

Decaptcha: Breaking 75% of eBay Audio CAPTCHAs  

Microsoft Academic Search

CAPTCHA tests aim at preventing attackers from per- forming automatic website registration. In this paper we show that our prototype Decaptcha is able to success- fully break 75% of eBay audio captchas. We compare its performance with the state of the art, readily available speech recognition system Sphinx and discuss the impli- cations for eBay security.

Elie Bursztein; Steven Bethard

319

Learning from Animated Concept Maps with Concurrent Audio Narration  

ERIC Educational Resources Information Center

An animated concept map is a presentation of a network diagram in which nodes and links are sequentially added or modified. An experiment compared learning from animated concept maps and text by randomly assigning 133 undergraduates to study 1 of 4 narrated animations presenting semantically equivalent information accompanied by identical audio

Nesbit, John C.; Adesope, Olusola O.

2011-01-01

320

Multimedia content analysis-using both audio and visual clues  

Microsoft Academic Search

Multimedia content analysis refers to the computerized understanding of the semantic meanings of a multimedia document, such as a video sequence with an accompanying audio track. With a multimedia document, its semantics are embedded in multiple forms that are usually complimentary of each other, Therefore, it is necessary to analyze all types of data: image frames, sound tracks, texts that

Yao Wang; Zhu Liu; Jin-Cheng Huang

2000-01-01

321

A collaborative computing model for audio post-production  

Microsoft Academic Search

Networked systems for audio post-production provide solutions to several persistent problems in the cinema industry. By enabling remote collaboration between media professionals, networked computing increases efficiency and reduces costs. This practice creates a virtual organization in which media and media editing devices are tightly synchronized between remote locations. An experimental system meeting these needs is described. Two successful demonstrations of

Nathan Brock; Michelle Daniels; Steve Morris; Peter Otto

2011-01-01

322

Analysis of an Audio Control Protocol with Bus Collision  

Microsoft Academic Search

We analyze the data link layer of a protocol developed by Philips toconnect the devices of an audio set. The protocol allows significant timeuncertainty, and because several componets are conected to one bus theprotocol has to deal with bus collisions. Besides a proof of correctness themaximal tolerance on timing for which the protocol still functions correctlyis given. For the verification

W. o. d. Griffioen

1994-01-01

323

Evaluation of an Audio Cassette Tape Lecture Course  

ERIC Educational Resources Information Center

An audio-cassette continuing education course (Selected Topics in Pharmacology) from Extension Services in Pharmacy at the University of Wisconsin was offered to a selected test market of pharmacists and evaluated using a pre-, post-test design. Results showed significant increase in cognitive knowledge and strong approval of students. (JT)

Blank, Jerome W.

1975-01-01

324

Multicamera Audio-Visual Analysis of Dance Figures  

Microsoft Academic Search

We present an automated system for multicamera motion capture and audio-visual analysis of dance figures. The multiview video of a dancing actor is acquired using 8 synchronized cameras. The motion capture technique is based on 3D tracking of the markers attached to the person's body in the scene, using stereo color information without need for an explicit 3D model. The

Ferda Ofli; Yasemin Demir; Engin Erzin; Yücel Yemez; A. Murat Tekalp

2007-01-01

325

Training Ircam's score follower [audio to musical score alignment system  

Microsoft Academic Search

This paper describes our attempt to make the hidden Markov model (HMM) score following system, developed at Ircam, sensible to past experiences in order to obtain better audio to score real-time alignment for musical applications. A new observation modeling based on Gaussian mixture models is developed which is trainable using a learning algorithm we would call automatic discriminative training. The

Arshia Cont; Diemo Schwarz; Norbert Schnell

2005-01-01

326

MP3 Audio Quality for Single and Multiple Encoding  

Microsoft Academic Search

The results of several measurements of audio quality of single and multiple MPEG-Layer 3 encodings will be presented. Subjective and objective tests have been done. A comparisons between the scores of the objective difference grade (ODG) and the subjective difference grade (SDG) prove a more accuracy of the PEAQ method at higher bit-rate. The recent versions of the LAME encoder

Giancarlo Vercellesi; Andrea Vitali; Martino Zerbini

2007-01-01

327

An Efficient Watermarking Method for MP3 Audio Files  

Microsoft Academic Search

In this work, we present for the first time in our perception an efficient digital watermarking scheme for mpeg audio layer 3 files that operates directly in the compressed data domain, while manipulating the time and subband\\/channel domain. In addition, it does not need the original signal to detect the watermark. Our scheme was implemented taking special care for the

Dimitrios Koukopoulos; Yannis C. Stamatiou

2005-01-01

328

An Adaptive Watermarking Algorithm for MP3 Compressed Audio Signals  

Microsoft Academic Search

MP3 has been generating a significant popularity for distributing digital audio signals over the Internet. However, copyright issue has been raised because of illegal copy and redistribution. Digital watermarking is a technology that allows users to embed watermark into digital contents to identify the copyright holder, to prevent illegal copy, and to verily modification to the original content. This paper

Bingwei Chen; Jiying Zhao; Dali Wang

2008-01-01

329

Mobile Audio - from MP3 to AAC and further  

Microsoft Academic Search

The purpose of this paper is to evaluate the advanced audio codec's and reflect over their suitability for mobile needs of today and tomorrow. The historical development of different codec's for different purposes is analyzed. The features of the most common codec's are discussed in parallel with performance and other criteria. The capabilities of mobile devices and the telecommunication possibilities

Henri Autti; Johnny Biström

330

Design of elevator audio player controlled by wireless communication  

Microsoft Academic Search

The objective of elevator audio playing system mainly concerns the device installed in elevator to provide music, advertisement and notifications. In this paper, GPRS is utilized for mode control and wireless instruction sequence update. As the core of the whole system, STM32 Microcontroller unit can browse and manage the catalog of files in SD card. Additionally, through building-up FAT 16

Xintong Zhang; Chengdong Wu; Yuanlong Wang; Bingyang Li; Yunzhou Zhang

2010-01-01

331

HMM Based Falling Person Detection Using Both Audio and Video  

Microsoft Academic Search

Automatic detection of a falling person in video is an important problem with applications in security and safety areas including supportive home environments and CCTV surveillance systems. Human motion in video is modeled using Hidden Markov Models (HMM) in this paper. In addition, the audio track of the video is also used to distinguish a person simply sitting on a

B. Ugur Töreyin; Yigithan Dedeoglu; A. Enis Çetin

2005-01-01

332

Synchronization of Multiple Camera Videos Using Audio-Visual Features  

Microsoft Academic Search

Digital video capturing is getting popular with the decreasing price of camcorders and the increasing availability of devices with embedded video cameras such as digital-still cameras, mobile phones and PDAs. While a raw home video is considered as visually non-appealing, having multiple recordings of the same event provides the opportunity to combine audio and video segments from different cameras for

Prarthana Shrestha; Mauro Barbieri; Hans Weda; Dragan Sekulovski

2010-01-01

333

Wavelet analysis for audio signals with music classification applications  

Microsoft Academic Search

Audio data like speech and music can be analyzed and processed with Fourier methods, having as one constraint the constant product of time and frequency resolutions. This problem can be avoided applying the wavelet transform, ensuring good resolutions on both time and frequency supports. We propose in this paper to determine features of music in a combined framework using multi-resolution

Anca Popescu; Inge Gavat; Mihai Datcu

2009-01-01

334

Packet Audio Playout Delay Adjustment: Performance Bounds and Algorithms  

Microsoft Academic Search

In packet audio applications, packets are buffered at a receiving site and their playout delayed in order to com- pensate for variable network delays. In this paper, we con- sider the problem of adaptively adjusting the playout delay in order to keep this delay as small as possible, while at the same time avoiding excessive \\

Sue B. Moon; James F. Kurose; Donald F. Towsley

1998-01-01

335

Finding Copyright-free Images, Audio and Video  

Microsoft Academic Search

This hands-on workshop takes you on a tour of the web sites and online resources that provide a variety of media (text, images, audio, video, etc.) that can be used in educational materials without the typical restrictions of copyright. Some are public domain, while others are special resources available only for non-profit or educational use. Please register for this event

Fred Zinn

2009-01-01

336

An experimental comparison of audio tempo induction algorithms  

Microsoft Academic Search

We report on the tempo induction contest organ- ised during the International Conference on Music Information Retrieval (ISMIR 2004) held at the University Pompeu Fabra in Barcelona in October 2004. The goal of this contest was to evaluate some state-of-the-art algorithms in the task of inducing the basic tempo (as a scalar, in beats per minute) from musical audio signals.

Fabien Gouyon; Anssi Klapuri; Simon Dixon; Miguel Alonso; George Tzanetakis; Christian Uhle; Pedro Cano

2006-01-01

337

Audio-visual perception system for a humanoid robotic head.  

PubMed

One of the main issues within the field of social robotics is to endow robots with the ability to direct attention to people with whom they are interacting. Different approaches follow bio-inspired mechanisms, merging audio and visual cues to localize a person using multiple sensors. However, most of these fusion mechanisms have been used in fixed systems, such as those used in video-conference rooms, and thus, they may incur difficulties when constrained to the sensors with which a robot can be equipped. Besides, within the scope of interactive autonomous robots, there is a lack in terms of evaluating the benefits of audio-visual attention mechanisms, compared to only audio or visual approaches, in real scenarios. Most of the tests conducted have been within controlled environments, at short distances and/or with off-line performance measurements. With the goal of demonstrating the benefit of fusing sensory information with a Bayes inference for interactive robotics, this paper presents a system for localizing a person by processing visual and audio data. Moreover, the performance of this system is evaluated and compared via considering the technical limitations of unimodal systems. The experiments show the promise of the proposed approach for the proactive detection and tracking of speakers in a human-robot interactive framework. PMID:24878593

Viciana-Abad, Raquel; Marfil, Rebeca; Perez-Lorenzo, Jose M; Bandera, Juan P; Romero-Garces, Adrian; Reche-Lopez, Pedro

2014-01-01

338

The Audio-Visual Marketing Handbook for Independent Schools.  

ERIC Educational Resources Information Center

This how-to booklet offers specific advice on producing video or slide/tape programs for marketing independent schools. Five chapters present guidelines for various stages in the process: (1) Audio-Visual Marketing in Context (aesthetics and economics of audiovisual marketing); (2) A Question of Identity (identifying the audience and deciding on…

Griffith, Tom

339

Media Convergence: Grand Theft Audio: Negotiating Copyright as Composers  

Microsoft Academic Search

Today, writing often requires composers to draw upon multiple modes of meaning making. Today's computers and robust networks allow writers to choreograph audio, video, other visual elements, text, and more. This is new. Admittedly, some professionals have been mixing media for years to create advertisements, movies, and CDs, for instance, but access to these technologies is now available in ways

Dànielle Nicole DeVoss; Suzanne Webb

2008-01-01

340

Experiments in Composing Proxy Audio Services for Mobile Users  

Microsoft Academic Search

Abstract: This paper describes an experimental study in the use of acomposable proxy framework to improve the quality of interactive audiostreams delivered to mobile hosts. Two forward error correction (FEC)proxylets are developed, one using block erasure codes, and the otherusing the GSM 06.10 encoding algorithm. Separately, each type of FECimproves the ability of the audio stream to tolerate errors in

Philip K. Mckinley; Udiyan I. Padmanabhan; Nandagopal Ancha

2001-01-01

341

Streaming Audio and Video: New Challenges and Opportunities for Museums.  

ERIC Educational Resources Information Center

Streaming audio and video present new challenges and opportunities for museums. Streaming media is easier to author and deliver to Internet audiences than ever before; digital video editing is commonplace now that the tools--computers, digital video cameras, and hard drives--are so affordable; the cost of serving video files across the Internet…

Spadaccini, Jim

342

Dynamic Mechanical Properties Tester for Low Audio and Subaudio Frequencies  

Microsoft Academic Search

A dynamic mechanical properties tester is described for use in the low audio and subaudio frequency range on various types of flabby material, including biological tissue. The moving element of the device is supported by a gas bearing with negligible springiness and friction and contains about 2.5 g of mass. A single coil in a magnetic field is used for

David A. Keiper

1962-01-01

343

Libraries and Audio-Visual Center Cost Allocation Study.  

National Technical Information Service (NTIS)

A 1975 cost analysis study of the Purdue University Library and Audio-Visual Center collected data to ascertain the cost of services and materials during the fiscal year 1975 and to identify the allocation of costs among user groups, among library functio...

M. A. Drake

1976-01-01

344

Survey of the State of Audio Collections in Academic Libraries  

ERIC Educational Resources Information Center

The goal of this survey was to collect and analyze baseline information about the status of audio collections held by a set of research institutions. This information can help shape the national preservation plan now being developed by the National Recording Preservation Board (NRPB) and the Library of Congress to preserve "sound recordings that…

Smith, Abby; Allen, David Randal; Allen, Karen

2004-01-01

345

Modern Construction Practices for CBS Audio-Video Systems  

Microsoft Academic Search

Equipment Construction practices developed in recent years by the CBS Television Network have contributed to improving the reliability of audio-video systems, and to reducing installation and maintenance costs. The procedures developed to obtain this result, as well as the materials and techniques used, are described. Although the subject matter is of specific interest to broadcasters, much of it applies to

Charles J. Neenan

1963-01-01

346

Artifact-free asynchronous geometry-based audio rendering  

Microsoft Academic Search

Most audio rendering systems include a geometry-based simulation engine which computes and updates sound propagation paths and an auralization engine which renders audible the resulting sound field. In the case of dynamic environments the attributes of the sound paths vary over time, which can cause severe artifacts in the auralized sound signal. To avoid this problem, geometrical calculations have to

Nicolas Tsingos

2001-01-01

347

A chip set for a digital audio broadcasting channel decoder  

Microsoft Academic Search

In this paper the design of two chips for an ASIC based channel decoder for a Digital Audio Broadcasting (DAB) system is discussed. The ASIC solution is a follow-up to an expensive implementation which is based on general purpose DSP processors. Both ASICs are used in a test receiver and a precursor consumer DAB receiver

A. Delaruelle; J. Huisken; J. Van Loon; F. Welten

1995-01-01

348

Adaptive delay estimation for low jitter audio over Internet  

Microsoft Academic Search

Real time voice applications typically produce uniformly spaced voice packets and faithful reconstruction demands that these be played out at the same intervals. Best effort packet networks, however, produce variable delays on different packets and the receiver is required to buffer the received packets before playout. Excessive buffering delays deteriorate the system performance for interactive audio and so intelligent algorithms

Aman Kansal; Abhay Karandikar

2001-01-01

349

Low-power audio classification for ubiquitous sensor networks  

Microsoft Academic Search

In the past researchers have proposed a variety of features that are based on the human auditory system. However none of these features have been able to replace mel-frequency cepstral coefficients (MFCC) as the preferred feature for audio classification problems, either because of computational costs involved or because of their poor performance in the presence of noise. In this paper

Sourabh Ravindran; David Anderson; Malcolm Slaney

2004-01-01

350

A lossless audio compression software for Windows application  

Microsoft Academic Search

This paper focuses on the design of an adaptive lossless compression algorithm for audio waveforms, which aims to be used in Internet, PC game, library archive applications, etc. Windows based software is developed which can also be used in DOS or UNIX operating system. The lossless compression is based on the Rice code (Rice and Plant, 1971), and a scheme

Lin Xiao; Tan Wee Hong; Gee Chein Woei; Li Gang

1998-01-01

351

An iterated rational filter bank for audio coding  

Microsoft Academic Search

This paper proposes a regular third-of-an-octave filter bank for high fidelity audio coding. The originality here is twofold: first, the filter bank is an iterated orthonormal rational filter bank for which the generating filters have been designed so that its outputs closely approximate a wavelet transform. This is different from the known coding algorithms which all use an integer filter

T. Blu

1996-01-01

352

Data Hiding in Digital Audio by Frequency Domain Dithering  

Microsoft Academic Search

A technique that inserts data densely into short frames in a digital audio signal by frequency domain dithering is described. With the proposed method, large embedding capacity can be realized, and the presence of the hid- den data is imperceptible. Synchronization in detection is achieved by using a two-step search process that accurately locates a PN sequence-based pilot signal attached

Shuozhong Wang; Xinpeng Zhang; Kaiwen Zhang

2003-01-01

353

Sounds in CD-ROM--Integrating Audio in Multimedia Products.  

ERIC Educational Resources Information Center

Describes how audio technology is being integrated into CD-ROMs to create multimedia products. Computer hardware and software are discussed, including the use of HyperCard to combine still pictures, moving video pictures, and sound; and specific new multimedia products produced by the Voyager Company are described. (LRW)

Rosebush, Judson

1992-01-01

354

Robust video and audio-based synchronization of multimedia files  

Microsoft Academic Search

This paper addresses the problem of robust and automated synchronization of multiple audio and video signals. The input signals are from a set of independent multimedia recordings coming from several camcorders and microphones. While the camcorders are static, the microphones are mobile as they are attached to people. The motivation for synchronization of all signals is to support studies and

Benjamin A. Raichel; Peter Bajcsy

2010-01-01

355

Towards practical deployment of audio-visual speech recognition  

Microsoft Academic Search

Much progress has been achieved during the past two decades in audio-visual automatic speech recognition (AVASR). However, challenges persist that hinder AVASR deployment in practical situations, most notably, robust and fast extraction of visual speech features. We review our efforts in overcoming this problem, based on an appearance-based visual feature representation of the speaker's mouth region. We cover three topics

G. Potamianos; C. Neti; J. Huang; J. H. Connell; S. Chu; V. Libal; E. Marcheret; N. Haas; J. Jiang

2004-01-01

356

Sony's Data Discman: A Look at These New Portable Information Machines and What They Mean for CD-ROM Developers.  

ERIC Educational Resources Information Center

Describes a portable CD-ROM machine intended for the mass market that provides access to searchable text, graphics, and audio through a user-friendly interface. Six search modes and other system features are reviewed, and electronic texts for the unit are introduced. A table compares features of the two available models. (NRP)

Bonime, Andrew

1992-01-01

357

[Ventriloquism and audio-visual integration of voice and face].  

PubMed

Presenting synchronous auditory and visual stimuli in separate locations creates the illusion that the sound originates from the direction of the visual stimulus. Participants' auditory localization bias, called the ventriloquism effect, has revealed factors affecting the perceptual integration of audio-visual stimuli. However, many studies on audio-visual processes have focused on performance in simplified experimental situations, with a single stimulus in each sensory modality. These results cannot necessarily explain our perceptual behavior in natural scenes, where various signals exist within a single sensory modality. In the present study we report the contributions of a cognitive factor, that is, the audio-visual congruency of speech, although this factor has often been underestimated in previous ventriloquism research. Thus, we investigated the contribution of speech congruency on the ventriloquism effect using a spoken utterance and two videos of a talking face. The salience of facial movements was also manipulated. As a result, when bilateral visual stimuli are presented in synchrony with a single voice, cross-modal speech congruency was found to have a significant impact on the ventriloquism effect. This result also indicated that more salient visual utterances attracted participants' auditory localization. The congruent pairing of audio-visual utterances elicited greater localization bias than did incongruent pairing, whereas previous studies have reported little dependency on the reality of stimuli in ventriloquism. Moreover, audio-visual illusory congruency, owing to the McGurk effect, caused substantial visual interference to auditory localization. This suggests that a greater flexibility in responding to multi-sensory environments exists than has been previously considered. PMID:22764349

Yokosawa, Kazuhiko; Kanaya, Shoko

2012-07-01

358

Deutsch Durch Audio-Visuelle Methode: An Audio-Lingual-Oral Approach to the Teaching of German.  

ERIC Educational Resources Information Center

This teaching guide, designed to accompany Chilton's "Deutsch Durch Audio-Visuelle Methode" for German 1 and 2 in a three-year secondary school program, focuses major attention on the operational plan of the program and a student orientation unit. A section on teaching a unit discusses four phases: (1) presentation, (2) explanation, (3)…

Dickinson Public Schools, ND. Instructional Media Center.

359

Maintaining high-quality IP audio services in lossy IP network environments  

NASA Astrophysics Data System (ADS)

In this paper we present our research activities in the area of digital audio processing and transmission. Today's available teleconference audio solutions are lacking in flexibility, robustness and fidelity. There was a need for enhancing the quality of audio for IP-based applications to guarantee optimal services under varying conditions. Multiple tests and user evaluations have shown that a reliable audio communication toolkit is essential for any teleconference application. This paper summarizes our research activities and gives an overview of developed applications. In a first step the parameters, which influence the audio quality, were evaluated. All of these parameters have to be optimized in order to result into the best achievable quality. Therefore it was necessary to enhance existing schemes or develop new methods. Applications were developed for Internet-Telephony, broadcast of live music and spatial audio for Virtual Reality environments. This paper describes these applications and issues of delivering high quality digital audio services over lossy IP networks.

Barton, Robert J.; Chodura, Hartmut

2000-07-01

360

The method of narrow-band audio classification based on universal noise background model  

NASA Astrophysics Data System (ADS)

Audio classification is the basis of content-based audio analysis and retrieval. The conventional classification methods mainly depend on feature extraction of audio clip, which certainly increase the time requirement for classification. An approach for classifying the narrow-band audio stream based on feature extraction of audio frame-level is presented in this paper. The audio signals are divided into speech, instrumental music, song with accompaniment and noise using the Gaussian mixture model (GMM). In order to satisfy the demand of actual environment changing, a universal noise background model (UNBM) for white noise, street noise, factory noise and car interior noise is built. In addition, three feature schemes are considered to optimize feature selection. The experimental results show that the proposed algorithm achieves a high accuracy for audio classification, especially under each noise background we used and keep the classification time less than one second.

Rui, Rui; Bao, Chang-chun

2013-03-01

361

Machine and process characterization  

SciTech Connect

A study was conducted to statistically characterize 11 precision machining centers to determine their operating characteristics and process capabilities. Measurement probes and a ball plate were used for measurement analysis. A generic test part designed with geometric features that the department typically manufactures was machined using various machining processes. A better understanding of each machine's characteristics and process capability was realized through repeating these methods on each machine.

Love, L.W.

1992-12-01

362

Automatic audio signing. Volume 2: Review, analysis and design  

NASA Astrophysics Data System (ADS)

Automati Audio Signing, also referred to as an 'automatic highway advisory radio system' (AHAR) provides appropriately equipped motor vehicles one way non commerical communications pertaining to traffic, road and weather conditions, travel advisories, directions, tourist information and other matters of interest to the traveling public. The automatic audio signing project reduces accidents by providing advance warning of hazardous traffic, weather and road conditions; saves the motorists' time and fuel, and reduces motorist irritation by improving traffic control and provides route diversion information when justified by traffic congestion or road blockage; and provides directions, locations of tourist facilities, descriptions of points of interest, and other messages intended to enhance the convenience and enjoyment of the traveling public.

1981-11-01

363

Robust Audio Watermarking by Using Low-Frequency Histogram  

NASA Astrophysics Data System (ADS)

In continuation to earlier work where the problem of time-scale modification (TSM) has been studied [1] by modifying the shape of audio time domain histogram, here we consider the additional ingredient of resisting additive noise-like operations, such as Gaussian noise, lossy compression and low-pass filtering. In other words, we study the problem of the watermark against both TSM and additive noises. To this end, in this paper we extract the histogram from a Gaussian-filtered low-frequency component for audio watermarking. The watermark is inserted by shaping the histogram in a way that the use of two consecutive bins as a group is exploited for hiding a bit by reassigning their population. The watermarked signals are perceptibly similar to the original one. Comparing with the previous time-domain watermarking scheme [1], the proposed watermarking method is more robust against additive noise, MP3 compression, low-pass filtering, etc.

Xiang, Shijun

364

VLSI Implementation for Portable Application Oriented MPEG4 Audio Codec  

Microsoft Academic Search

In this paper, we present an audio codec for portable application, which supports MPEG-2\\/MPEG-4 AAC LC profile. The presented codec has been implemented using UMC\\/HJTC 0.18mum CMOS technology. In this implementation, optimized algorithms are developed and most memory and calculation units are shared between encoder and decoder so that the features of low power and low cost are achieved. The

Peilin Liu; Lingzhi Liu; Ning Deng; Xuan Fu; Jiayan Liu; Qianru Liu; Guocheng Zhang; Bin He

2007-01-01

365

An audio DSP Toolkit for rapid application development in Flash  

Microsoft Academic Search

The Adobe Flash platform has become the de facto standard for developing and deploying media rich Web applications and games. The relative ease-of-development and cross-platform architecture of Flash enables designers to rapidly prototype graphically rich interactive applications, but comprehensive support for audio and signal processing has been lacking. ActionScript, the primary development language used for Flash, is poorly suited for

Travis M. Doll; Raymond Migneco; Jeff J. Scott; Youngmoo E. Kim

2009-01-01

366

Sensory dominance in combinations of audio, visual and haptic stimuli  

Microsoft Academic Search

Participants presented with auditory, visual, or bi-sensory audio–visual stimuli in a speeded discrimination task, fail to\\u000a respond to the auditory component of the bi-sensory trials significantly more often than they fail to respond to the visual\\u000a component—a ‘visual dominance’ effect. The current study investigated further the sensory dominance phenomenon in all combinations\\u000a of auditory, visual and haptic stimuli. We found

David Hecht; Miriam Reiner

2009-01-01

367

Specication and Design of a MP3 Audio Decoder  

Microsoft Academic Search

In an effort to understand, experience and prove the benets of automated SoC design, this report describes the specication modeling, design space exploration and implementation of a real world example using SpecC based System on Chip Environment (SCE). The report covers a complete description of developing the specication model of a MPEG-1 Layer 3 (MP3) audio decoder in SpecC language

Pramod Chandraiah; Rainer D

368

Transform coding of audio signals using perceptual noise criteria  

Microsoft Academic Search

A 4-b\\/sample transform coder is designed using a psychoacoustically derived noise-making threshold that is based on the short-term spectrum of the signal. The coder has been tested in a formal subjective test involving a wide selection of monophonic audio inputs. The signals used in the test were of 15-kHz bandwidth, sampled at 32 kHz. The bit rate of the resulting

JAMES D. JOHNSTON

1988-01-01

369

jetAudio 6.2.8 Basic  

NSDL National Science Digital Library

Contained within a streamlined silver package is jetAudio 6.2.8 Basic. It presents a welcome alternative to other multimedia players, and it allows users to utilize a number of equalizers, speed controls, and of course, the cross-fade option. Additionally, for the truly brave, the application also includes a synchronized lyrics display for karaoke. This version is compatible with computers running Windows 98, Me, 2000, and XP.

2007-01-01

370

A Distributed Real-Time MPEG Video Audio Player  

Microsoft Academic Search

. This paper presents the design, implementation and experimentalanalysis of a distributed, real-time MPEG video and audio player.The player is designed for use across the Internet, a shared environmentwith variable traffic and with great diversity in network bandwidth andhost processing speed. We use a novel toolkit approach to build softwarefeedback mechanisms for client\\/server synchronization, dynamicQuality-of-Service control, and system adaptiveness. Our

Shanwei Cen; Calton Pu; Richard Staehli; Crispin Cowan; Jonathan Walpole

1995-01-01

371

Entropy and Dynamism Criteria for Speech and Audio Classification Applications  

Microsoft Academic Search

\\u000a We describe the audio classification system that uses entropy and dynamism criteria as discrimination features. The main idea\\u000a of this approach is that the input neural net is considered as a informational channel. Channel tuned to the certain type\\u000a of information transmits it best of all according to the informational criterion. In our case a multilayer perceptron (MLP)\\u000a emitted posterior

Igor E. Kheidorov; Hanna M. Lukashevich; Denis L. Mitrofanov

2003-01-01

372

Audio communications with a mid-IR laser  

NASA Astrophysics Data System (ADS)

We have demonstrated audio communications with a mid-IR laser. The laser is a frequency doubled Q-switched CO2 system producing approximately 12ns pulses at 4.6?m. The audio signal was encoded on the beam by means of pulse frequency modulation (PFM) with a carrier frequency of 37kHz. A 1mm diameter, low noise thermoelectrically cooled IR photovoltaic detector with electrical bandwidth 250MHz was used to detect the laser beam. A custom-built circuit stretched the resultant electrical pulses to approximately 1.5?s, before being demodulated. High quality audio signals were received and recorded, and still images were successfully transmitted using slow scan television techniques. The demonstration was conducted at the Defence Science & Technology Organisation's laser range at Edinburgh, South Australia in July 2008. The distance was 1.5km, with a slant path (8m to 1.5m). The maximum range using this system is estimated to be tens of kilometres.

Grant, Kenneth J.; Clare, Bradley A.; Martinsen, Wayne; Dubovinsky, Miro; Isterling, William; Wright, Daniel; Mudge, Kerry A.

2008-09-01

373

Audio conferencing using dynamic channel-assignment system  

NASA Astrophysics Data System (ADS)

This study examines dynamic channel assignment techniques to demonstrate their feasibility in audio teleconferencing. Computer simulation of dynamic assignment is used to investigate the rules for controlling the bit-rate and is used to subjectively evaluate the speech quality of audio conferencing system. Efficient encoding of speech signals requires the removal of redundancies in the speech waveform. Predictive codings techniques are applicable to dynamic channel assignment. Adaptive Predictive Coding at 16 kbps is suitable for a high quality/high rate channel. The closely related technique of Linear Predictive Coding is appropriate for a low quality/low rate channel. A dual-mode coder combines both the above techniques together with a switching control algorithm. The control algorithm selects the mode of the dual-mode coder based on the relative amplitudes of the conference participants. Computer simulations of a dual-mode coder indicate that the technique is well suited to audio conferencing. Dynamic assignment of the coding mode introduces only minor quality degradations when proper control algorithms are used. The overall impression is one of high quality coding for most of the duration of the conference.

1982-04-01

374

Robust video and audio-based synchronization of multimedia files  

NASA Astrophysics Data System (ADS)

This paper addresses the problem of robust and automated synchronization of multiple audio and video signals. The input signals are from a set of independent multimedia recordings coming from several camcorders and microphones. While the camcorders are static, the microphones are mobile as they are attached to people. The motivation for synchronization of all signals is to support studies and understanding of human interaction in a decision support environment that have been limited so far due to the difficulties in automated processing of any observations during the decision making sessions. The application of our work is to environments supporting decisions. The data sets for this work have been acquired during training exercises of response teams, rescue workers, and fire fighters at multiple locations. The developed synchronization methodology for a set of independent multimedia recordings is based on introducing aural and visual landmarks with a bell and room light switches. Our approach to synchronization is based on detecting the landmarks in audio and video signals per camcorder and per microphone, and then fusing the results to increase robustness and accuracy of the synchronization. We report synchronization results that demonstrate accuracy of synchronization based on video and audio.

Raichel, Benjamin A.; Bajcsy, Peter

2010-02-01

375

Audio recording and reproduction in CARROUSO: Getting closer to perfection?  

NASA Astrophysics Data System (ADS)

State-of-the-art systems for spatial audio reproduction utilize two to six discrete playback channels. A problem inherent to these systems is the relatively small area where the listener is able to experience a true 3-D sound sensation. This so-called ``sweet spot'' can be significantly enlarged by using loudspeaker arrays in combination with wave field synthesis (WFS) technology, initially developed at Delft University. By following this approach, actual sonic spaces can be reproduced in their entirety and not only discrete multichannel representations thereof. While loudspeaker arrays can be used to reproduce sound fields, microphone arrays can be used for sound field capture and analysis. Having high-quality audio reproduction in mind, microphone array designs are presented that need to fulfill stricter requirements than what has been traditionally considered for microphone array applications. Information on acoustic source position is essential for WFS-based rendering techniques. As will be shown, joint audio-video object tracking proves to be efficient for this task. Moreover, full-duplex applications based on WFS technology, like high-quality teleconferencing or remote music teaching, call for sophisticated multichannel acoustic echo cancellation algorithms. The European project ``CARROUSO'' aims at developing, integrating, and building a real-time system that embraces all previously described technologies in an MPEG-4 context.

Teutsch, Heinz; Spors, Sascha; Buchner, Herbert; Rabenstein, Rudolf; Kellermann, Walter

2002-05-01

376

A method to synchronise video cameras using the audio band.  

PubMed

This paper proposes and evaluates a novel method for synchronisation of video cameras using the audio band. The method consists in generating and transmitting an audio signal through radio frequency for receivers connected to the microphone input of the cameras and inserting the signal in the audio band. In a software environment, the phase differences among the video signals are calculated and used to interpolate the synchronous 2D projections of the trajectories. The validation of the method was based on: (1) Analysis of the phase difference changes as a function of time of two video signals. (2) Comparison between the values measured with an oscilloscope and by the proposed method. (3) Estimation of the improvement in the accuracy in the measurements of the distance between two markers mounted on a rigid body during movement applying the method. The results showed that the phase difference changes in time slowly (0.150 ms/min) and linearly, even when the same model of cameras are used. The values measured by the proposed method and by oscilloscope showed equivalence (R2=0.998), the root mean square of the difference between the measurements was 0.10 ms and the maximum difference found was 0.31 ms. Applying the new method, the accuracy of the 3D reconstruction had a statistically significant improvement. The accuracy, simplicity and wide applicability of the proposed method constitute the main contributions of this work. PMID:16439248

Leite de Barros, Ricardo Machado; Guedes Russomanno, Tiago; Brenzikofer, René; Jovino Figueroa, Pascual

2006-01-01

377

Hard Machinable Machining of Cobalt Super Alloys  

NASA Astrophysics Data System (ADS)

The article deals with difficult-to-machine cobalt super alloys. The main aim is to test the basic properties of cobalt super alloys and propose suitable cutting materials and machining parameters under the designation 188 when machining. Although the development of technology in chipless machining such as moulding, precision casting and other manufacturing methods continues to advance, machining is still the leading choice for piece production, typical for energy and chemical engineering. Nowadays, super alloys are commonly used in turbine engines in regions that are subject to high temperatures, which require high strength, high temperature resistance, phase stability, as well as corrosion or oxidation resistance.

?ep, Robert; Janásek, Adam; Petr?, Jana; ?epová, Lenka; Sadílek, Marek; Kratochvíl, Ji?í

2012-12-01

378

Stirling machine operating experience  

NASA Technical Reports Server (NTRS)

Numerous Stirling machines have been built and operated, but the operating experience of these machines is not well known. It is important to examine this operating experience in detail, because it largely substantiates the claim that Stirling machines are capable of reliable and lengthy lives. The amount of data that exists is impressive, considering that many of the machines that have been built are developmental machines intended to show proof of concept, and were not expected to operate for any lengthy period of time. Some Stirling machines (typically free-piston machines) achieve long life through non-contact bearings, while other Stirling machines (typically kinematic) have achieved long operating lives through regular seal and bearing replacements. In addition to engine and system testing, life testing of critical components is also considered.

Ross, Brad; Dudenhoefer, James E.

1991-01-01

379

Nontraditional Machining of Beryllium.  

National Technical Information Service (NTIS)

The report deals with electrichemical machining (ECM), chemical milling, and electric-discharge machining (EDM). The general characteristics of these processes and their applications to the processing of beryllium parts are presented and covered in detail...

J. A. Gurklis

1972-01-01

380

36 CFR 1194.4 - Definitions.  

Code of Federal Regulations, 2010 CFR

...not limited to, Braille, ASCII text, large print, recorded audio...Internet posting, captioning, text-to-speech synthesis, and audio description. Assistive...equipment that employs interactive text based communications...

2010-07-01

381

36 CFR 1194.4 - Definitions.  

Code of Federal Regulations, 2010 CFR

...not limited to, Braille, ASCII text, large print, recorded audio...Internet posting, captioning, text-to-speech synthesis, and audio description. Assistive...equipment that employs interactive text based communications...

2009-07-01

382

AudioSense: Enabling Real-time Evaluation of Hearing Aid Technology In-Situ  

PubMed Central

AudioSense integrates mobile phones and web technology to measure hearing aid performance in real-time and in-situ. Measuring the performance of hearing aids in the real world poses significant challenges as it depends on the patient's listening context. AudioSense uses Ecological Momentary Assessment methods to evaluate both the perceived hearing aid performance as well as to characterize the listening environment using electronic surveys. AudioSense further characterizes a patient's listening context by recording their GPS location and sound samples. By creating a time-synchronized record of listening performance and listening contexts, AudioSense will allow researchers to understand the relationship between listening context and hearing aid performance. Performance evaluation shows that AudioSense is reliable, energy-efficient, and can estimate Signal-to-Noise Ratio (SNR) levels from captured audio samples.

Hasan, Syed Shabih; Lai, Farley; Chipara, Octav; Wu, Yu-Hsiang

2014-01-01

383

The Advantage of Machines  

NSDL National Science Digital Library

In this lesson, students learn about work as defined by physical science and see that work is made easier through the use of simple machines. Already encountering simple machines everyday, students will be learn about their widespread uses in improving everyday life. This lesson serves as the starting point for the Simple Machines Unit.

Integrated Teaching And Learning Program

384

Your Sewing Machine.  

ERIC Educational Resources Information Center

The programed instruction manual is designed to aid the student in learning the parts, uses, and operation of the sewing machine. Drawings of sewing machine parts are presented, and space is provided for the student's written responses. Following an introductory section identifying sewing machine parts, the manual deals with each part and its…

Peacock, Marion E.

385

RRA: an audio format for single-source music and lyrics  

Microsoft Academic Search

Karaoke music has world-wide appeal, especially for non-professional singers. However, most karaoke-audio architectures involve separate text and audio data streams which run in different threads. Such an approach suffers from timing synchronization problems. Another drawback is the need for a karaoke system to process at least two different formats: one used for karaoke-text and other used for audio data. To

M. Rao; J. C. Lusth

2012-01-01

386

MultiStream Asynchrony Dynamic Bayesian Network Model for Audio-Visual Continuous Speech Recognition  

Microsoft Academic Search

How best to describe the asynchrony of the speech and lip motion is a key problem of audio-visual speech recognition model. A multi-stream asynchrony dynamic Bayesian network (MS-ADBN) model is brought forward for audio-visual speech recognition, and in this model, audio stream and visual stream are synchronous in word node, while between the word nodes, each stream has its own

Guoyun Lv; Dongmei Jiang; Rongchun Zhao; Xiaoyue Jiang; H. Sahli

2007-01-01

387

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

Microsoft Academic Search

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

Mark Ericson; Richard McKinley; Marion Kibbe; Daniel Francis

1994-01-01

388

Autocalibration of an SMT machine by machine vision  

Microsoft Academic Search

An SMT machine has many working coordinate frames—the fiducial mark camera frame, component camera frame, machine table frame, PCB frame, and reference frame. Because of many influences such as mechanical dimension errors, machine assembling errors, and camera lens distortions, all frames on the SMT machine must be calibrated to compensate for these machine errors. This paper applies machine vision techniques

C.-L. Shih; C.-W. Ruo

2005-01-01

389

ASTP video tape recorder ground support equipment (audio/CTE splitter/interleaver). Operations manual  

NASA Technical Reports Server (NTRS)

A descriptive handbook for the audio/CTE splitter/interleaver (RCA part No. 8673734-502) was presented. This unit is designed to perform two major functions: extract audio and time data from an interleaved video/audio signal (splitter section), and provide a test interleaved video/audio/CTE signal for the system (interleaver section). It is a rack mounting unit 7 inches high, 19 inches wide, 20 inches deep, mounted on slides for retracting from the rack, and weighs approximately 40 pounds. The following information is provided: installation, operation, principles of operation, maintenance, schematics and parts lists.

1974-01-01

390

A Virtual Audio Guidance and Alert System for Commercial Aircraft Operations  

NASA Technical Reports Server (NTRS)

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.; Wenzel, Elizabeth M.; Shrum, Richard; Miller, Joel; Null, Cynthia H. (Technical Monitor)

1996-01-01

391

Edheads: The Compound Machine  

NSDL National Science Digital Library

This resource is a collection of interactive animations designed to help kids learn how forces and simple machines can work together to create the compound machine. Child-centered animated activities enhance understanding of how compound machines function and how they are differentiated from simple machines. Additionally the site includes a glossary of important terms, lesson plans, a teacher's guide and information about professionals who work with compound machines. This page is part of a larger collection of animated education resources for the elementary level.

2007-09-18

392

The Monitoring of Machines Surveillance des Machines.  

National Technical Information Service (NTIS)

The application of classical vibration analysis to the surveillance of the mechanical condition of machines is described. Automated procedures handled by specialized computers, using advanced signal processing techniques, are sought. Vibration measurement...

M. Gaillochet

1980-01-01

393

Informed spectral analysis: audio signal parameter estimation using side information  

NASA Astrophysics Data System (ADS)

Parametric models are of great interest for representing and manipulating sounds. However, the quality of the resulting signals depends on the precision of the parameters. When the signals are available, these parameters can be estimated, but the presence of noise decreases the resulting precision of the estimation. Furthermore, the Cramér-Rao bound shows the minimal error reachable with the best estimator, which can be insufficient for demanding applications. These limitations can be overcome by using the coding approach which consists in directly transmitting the parameters with the best precision using the minimal bitrate. However, this approach does not take advantage of the information provided by the estimation from the signal and may require a larger bitrate and a loss of compatibility with existing file formats. The purpose of this article is to propose a compromised approach, called the 'informed approach,' which combines analysis with (coded) side information in order to increase the precision of parameter estimation using a lower bitrate than pure coding approaches, the audio signal being known. Thus, the analysis problem is presented in a coder/decoder configuration where the side information is computed and inaudibly embedded into the mixture signal at the coder. At the decoder, the extra information is extracted and is used to assist the analysis process. This study proposes applying this approach to audio spectral analysis using sinusoidal modeling which is a well-known model with practical applications and where theoretical bounds have been calculated. This work aims at uncovering new approaches for audio quality-based applications. It provides a solution for challenging problems like active listening of music, source separation, and realistic sound transformations.

Fourer, Dominique; Marchand, Sylvain

2013-12-01

394

Incorporating Auditory Models in Speech/Audio Applications  

NASA Astrophysics Data System (ADS)

Following the success in incorporating perceptual models in audio coding algorithms, their application in other speech/audio processing systems is expanding. In general, all perceptual speech/audio processing algorithms involve minimization of an objective function that directly/indirectly incorporates properties of human perception. This dissertation primarily investigates the problems associated with directly embedding an auditory model in the objective function formulation and proposes possible solutions to overcome high complexity issues for use in real-time speech/audio algorithms. Specific problems addressed in this dissertation include: 1) the development of approximate but computationally efficient auditory model implementations that are consistent with the principles of psychoacoustics, 2) the development of a mapping scheme that allows synthesizing a time/frequency domain representation from its equivalent auditory model output. The first problem is aimed at addressing the high computational complexity involved in solving perceptual objective functions that require repeated application of auditory model for evaluation of different candidate solutions. In this dissertation, a frequency pruning and a detector pruning algorithm is developed that efficiently implements the various auditory model stages. The performance of the pruned model is compared to that of the original auditory model for different types of test signals in the SQAM database. Experimental results indicate only a 4-7% relative error in loudness while attaining up to 80-90 % reduction in computational complexity. Similarly, a hybrid algorithm is developed specifically for use with sinusoidal signals and employs the proposed auditory pattern combining technique together with a look-up table to store representative auditory patterns. The second problem obtains an estimate of the auditory representation that minimizes a perceptual objective function and transforms the auditory pattern back to its equivalent time/frequency representation. This avoids the repeated application of auditory model stages to test different candidate time/frequency vectors in minimizing perceptual objective functions. In this dissertation, a constrained mapping scheme is developed by linearizing certain auditory model stages that ensures obtaining a time/frequency mapping corresponding to the estimated auditory representation. This paradigm was successfully incorporated in a perceptual speech enhancement algorithm and a sinusoidal component selection task.

Krishnamoorthi, Harish

395

Audio spectrum and sound pressure levels vary between pulse oximeters  

Microsoft Academic Search

Purpose  The variable-pitch pulse oximeter is an important intraoperative patient monitor. Our ability to hear its auditory signal\\u000a depends on its acoustical properties and our hearing. This study quantitatively describes the audio spectrum and sound pressure\\u000a levels of the monitoring tones produced by five variable-pitch pulse oximeters.\\u000a \\u000a \\u000a \\u000a \\u000a Methods  We compared the Datex-Ohmeda Capnomac Ultima, Hewlett-Packard M1166A, Datex-Engstrom AS\\/3, Ohmeda Biox 3700, and

Deven Chandra; Michael J. Tessler; John Usher

2006-01-01

396

TV audio and video on the same channel  

NASA Technical Reports Server (NTRS)

Transmitting technique adds audio to video signal during vertical blanking interval. SIVI (signal in the vertical interval) is used by TV networks and stations to transmit cuing and automatic-switching tone signals to augment automatic and manual operations. It can also be used to transmit one-way instructional information, such as bulletin alerts, program changes, and commercial-cutaway aural cues from the networks to affiliates. Additonally, it can be used as extra sound channel for second-language transmission to biligual stations.

Hopkins, J. B.

1979-01-01

397

A compact electroencephalogram recording device with integrated audio stimulation system.  

PubMed

A compact (96 x 128 x 32 mm(3), 374 g), battery-powered, eight-channel electroencephalogram recording device with an integrated audio stimulation system and a wireless interface is presented. The recording device is capable of producing high-quality data, while the operating time is also reasonable for evoked potential studies. The effective measurement resolution is about 4 nV at 200 Hz sample rate, typical noise level is below 0.7 microV(rms) at 0.16-70 Hz, and the estimated operating time is 1.5 h. An embedded audio decoder circuit reads and plays wave sound files stored on a memory card. The activities are controlled by an 8 bit main control unit which allows accurate timing of the stimuli. The interstimulus interval jitter measured is less than 1 ms. Wireless communication is made through bluetooth and the data recorded are transmitted to an external personal computer (PC) interface in real time. The PC interface is implemented with LABVIEW and in addition to data acquisition it also allows online signal processing, data storage, and control of measurement activities such as contact impedance measurement, for example. The practical application of the device is demonstrated in mismatch negativity experiment with three test subjects. PMID:20590254

Paukkunen, Antti K O; Kurttio, Anttu A; Leminen, Miika M; Sepponen, Raimo E

2010-06-01

398

Audio-Tactile Integration and the Influence of Musical Training  

PubMed Central

Perception of our environment is a multisensory experience; information from different sensory systems like the auditory, visual and tactile is constantly integrated. Complex tasks that require high temporal and spatial precision of multisensory integration put strong demands on the underlying networks but it is largely unknown how task experience shapes multisensory processing. Long-term musical training is an excellent model for brain plasticity because it shapes the human brain at functional and structural levels, affecting a network of brain areas. In the present study we used magnetoencephalography (MEG) to investigate how audio-tactile perception is integrated in the human brain and if musicians show enhancement of the corresponding activation compared to non-musicians. Using a paradigm that allowed the investigation of combined and separate auditory and tactile processing, we found a multisensory incongruency response, generated in frontal, cingulate and cerebellar regions, an auditory mismatch response generated mainly in the auditory cortex and a tactile mismatch response generated in frontal and cerebellar regions. The influence of musical training was seen in the audio-tactile as well as in the auditory condition, indicating enhanced higher-order processing in musicians, while the sources of the tactile MMN were not influenced by long-term musical training. Consistent with the predictive coding model, more basic, bottom-up sensory processing was relatively stable and less affected by expertise, whereas areas for top-down models of multisensory expectancies were modulated by training.

Kuchenbuch, Anja; Paraskevopoulos, Evangelos; Herholz, Sibylle C.; Pantev, Christo

2014-01-01

399

Improvement of information fusion-based audio steganalysis  

NASA Astrophysics Data System (ADS)

In the paper we extend an existing information fusion based audio steganalysis approach by three different kinds of evaluations: The first evaluation addresses the so far neglected evaluations on sensor level fusion. Our results show that this fusion removes content dependability while being capable of achieving similar classification rates (especially for the considered global features) if compared to single classifiers on the three exemplarily tested audio data hiding algorithms. The second evaluation enhances the observations on fusion from considering only segmental features to combinations of segmental and global features, with the result of a reduction of the required computational complexity for testing by about two magnitudes while maintaining the same degree of accuracy. The third evaluation tries to build a basis for estimating the plausibility of the introduced steganalysis approach by measuring the sensibility of the models used in supervised classification of steganographic material against typical signal modification operations like de-noising or 128kBit/s MP3 encoding. Our results show that for some of the tested classifiers the probability of false alarms rises dramatically after such modifications.

Kraetzer, Christian; Dittmann, Jana

2010-02-01

400

A compact electroencephalogram recording device with integrated audio stimulation system  

NASA Astrophysics Data System (ADS)

A compact (96×128×32 mm3, 374 g), battery-powered, eight-channel electroencephalogram recording device with an integrated audio stimulation system and a wireless interface is presented. The recording device is capable of producing high-quality data, while the operating time is also reasonable for evoked potential studies. The effective measurement resolution is about 4 nV at 200 Hz sample rate, typical noise level is below 0.7 ?Vrms at 0.16-70 Hz, and the estimated operating time is 1.5 h. An embedded audio decoder circuit reads and plays wave sound files stored on a memory card. The activities are controlled by an 8 bit main control unit which allows accurate timing of the stimuli. The interstimulus interval jitter measured is less than 1 ms. Wireless communication is made through bluetooth and the data recorded are transmitted to an external personal computer (PC) interface in real time. The PC interface is implemented with LABVIEW® and in addition to data acquisition it also allows online signal processing, data storage, and control of measurement activities such as contact impedance measurement, for example. The practical application of the device is demonstrated in mismatch negativity experiment with three test subjects.

Paukkunen, Antti K. O.; Kurttio, Anttu A.; Leminen, Miika M.; Sepponen, Raimo E.

2010-06-01

401

Audio annotation watermarking with robustness against DA/AD conversion  

NASA Astrophysics Data System (ADS)

In the paper we present a watermarking scheme developed to meet the specific requirements of audio annotation watermarking robust against DA/AD conversion (watermark detection after playback by loudspeaker and recording with a microphone). Additionally the described approach tries to achieve a comparably low detection complexity, so it could be embedded in the near future in low-end devices (e.g. mobile phones or other portable devices). We assume in the field of annotation watermarking that there is no specific motivation for attackers to the developed scheme. The basic idea for the watermark generation and embedding scheme is to combine traditional frequency domain spread spectrum watermarking with psychoacoustic modeling to guarantee transparency and alphabet substitution to improve the robustness. The synchronization and extraction scheme is designed to be much less computational complex than the embedder. The performance of the scheme is evaluated in the aspects of transparency, robustness, complexity and capacity. The tests reveals that 44% out of 375 tested audio files pass the simulation test for robustness, while the most appropriate category shows even 100% robustness. Additionally the introduced prototype shows an averge transparency of -1.69 in SDG, while at the same time having a capacity satisfactory to the chosen application scenario.

Qian, Kun; Kraetzer, Christian; Biermann, Michael; Dittmann, Jana

2010-02-01

402

Internet-oriented visualization with audio presentation of speech signals  

NASA Astrophysics Data System (ADS)

Visualization of speech signals, including the capability to visualize the waveforms while simultaneously hearing the speech, is among the essential requirements in speech processing research. In tasks related to labeling of speech signals, visualization activities may have to be performed by multiple users upon a centralized collection of speech data. When speech labeling activities involve perceptual issues, the human factors issues including functionality tradeoffs are particularly important, since the user's burden (tiredness, annoyance) can affect the perceptual responses. We developed VideVox (pronounced 'Veedeh-Vox'), a speech visualization facility, in which the visualization activities may be performed by a large number of users in geographically, dialectically and linguistically diverse locations. Developed in Java, and capable of operating both as an Internet Java applet and a Java application, VideVox is platform independent. Using the client-server architecture paradigm, it allows distributed visualization work. The Internet orientation makes VideVox a promising direction for speech signal visualization in speech labeling activities that require a large number of users in multiple locations. In the paper, we describe our approach, VideVox features, modes of audio data exploration and audio-synchronous animation for speech visualization, operations related to identification of perceptual events, and the human factors issues related to perception-oriented visualizations of speech.

Braun, Jerome J.; Levkowitz, Haim

1998-05-01

403

Characteristics of the audio sound generated by ultrasound imaging systems  

NASA Astrophysics Data System (ADS)

Medical ultrasound scanners use high-energy pulses to probe the human body. The radiation force resulting from the impact of such pulses on an object can vibrate the object, producing a localized high-intensity sound in the audible range. Here, a theoretical model for the audio sound generated by ultrasound scanners is presented. This model describes the temporal and spectral characteristics of the sound. It has been shown that the sound has rich frequency components at the pulse repetition frequency and its harmonics. Experiments have been conducted in a water tank to measure the sound generated by a clinical ultrasound scanner in various operational modes. Results are in general agreement with the theory. It is shown that a typical ultrasound scanner with a typical spatial-peak pulse-average intensity value at 2 MHz may generate a localized sound-pressure level close to 100 dB relative to 20 ?Pa in the audible (<20 kHz) range under laboratory conditions. These findings suggest that fetuses may become exposed to a high-intensity audio sound during maternal ultrasound examinations. Therefore, contrary to common beliefs, ultrasound may not be considered a passive tool in fetal imaging..

Fatemi, Mostafa; Alizad, Azra; Greenleaf, James F.

2005-03-01

404

Machine tool locator  

DOEpatents

Machine tools can be accurately measured and positioned on manufacturing machines within very small tolerances by use of an autocollimator on a 3-axis mount on a manufacturing machine and positioned so as to focus on a reference tooling ball or a machine tool, a digital camera connected to the viewing end of the autocollimator, and a marker and measure generator for receiving digital images from the camera, then displaying or measuring distances between the projection reticle and the reference reticle on the monitoring screen, and relating the distances to the actual position of the autocollimator relative to the reference tooling ball. The images and measurements are used to set the position of the machine tool and to measure the size and shape of the machine tool tip, and examine cutting edge wear. patent

Hanlon, John A. (Los Alamos, NM); Gill, Timothy J. (Stanley, NM)

2001-01-01

405

Combining audio-based similarity with web-based data to accelerate automatic music playlist generation  

Microsoft Academic Search

We present a technique for combining audio signal-based music similarity with web-based musical artist similarity to accelerate the task of automatic playlist generation. We demonstrate the applicability of our proposed method by extending a recently published interface for music players that benefits from intelligent structuring of audio collections. While the original approach involves the calculation of sim- ilarities between every

Peter Knees; Tim Pohle; Markus Schedl; Gerhard Widmer

2006-01-01

406

Phantom Materialization: A Novel Method to Enhance Stereo Audio Reproduction on Headphones  

Microsoft Academic Search

EDICs : AUD-SMCA Abstract— Loudspeaker reproduction systems are sub- ject to a compromise between spatial realism and cost. By simulating loudspeaker reproduction on headphones, the resulting spatial realism is limited accordingly, despite the virtually unlimited spatial imaging capabilities of binaural audio rendering technology. More particularly, phantom imaging as often used for stereo audio mate- rial intended for loudspeaker reproduction is

Jeroen Breebaart; Erik Schuijers

2008-01-01

407

Enhancing recognition systems through an integrated processing of visual and audio information  

Microsoft Academic Search

The AVIS framework for integrated audio and visual information processing is applied to the problem of person identification. An instantiation of the AVIS framework, called PIAVI, is based on a fuzzy neural network (FuNN) model of audio-visual person identification. In PIAVI's unimodal (visual) mode of operation, only dynamic visual features are used, whereas in the bimodal mode of operation, dynamic

Eric Postma; N. Kasabov; Jaap van den Herik

1998-01-01

408

LiveDescribe: Can Amateur Describers Create High-Quality Audio Description?  

ERIC Educational Resources Information Center

Introduction: The study presented here evaluated the usability of the audio description software LiveDescribe and explored the acceptance rates of audio description created by amateur describers who used LiveDescribe to facilitate the creation of their descriptions. Methods: Twelve amateur describers with little or no previous experience with…

Branje, Carmen J.; Fels, Deborah I.

2012-01-01

409

A Survey of MPEG1 Audio, Video and Semantic Analysis Techniques  

Microsoft Academic Search

Digital audio & video data have become an integral part of multimedia information systems. To reduce storage and bandwidth requirements, they are commonly stored in a compressed format, such as MPEG-1. Increasing amounts of MPEG encoded audio and video documents are available online and in proprietary collections. In order to effectively utilise them, we need tools and techniques to automatically

Uma Srinivasan; Silvia Pfeiffer; Surya Nepal; Michael Lee; Lifang Gu; Stephen Barrass

2005-01-01

410

Development and Evaluation of Automatic Speaker based- Audio Identification and Segmentation for Broadcast News Recordings Indexation  

Microsoft Academic Search

In this paper, we describe an automatic- speaker based- audio segmentation and identification system for broadcasted news indexation purposes. We specifically focus on speaker identification and audio scene detection. Speaker identification (SI) is based on the state of the art Gaussian mixture models, whereas scene change detection process uses the classical Bayesian Information Criteria (BIC) and the recently proposed DISTBIC

Messaoud Bengherabi; Abdenour Sehad

2006-01-01

411

Histogram Specification-based Audio Watermarking Technology against Filtering Attacks in Time Domain  

Microsoft Academic Search

Filtering processing is a kind of typical and serious attack on the audio watermarking algorithm. Based on the computation in time domain before and after attacking on the audio, the mean and the standard deviation show good invariant statistical feature. The relation of four consecutive bins in a histogram can keep the change within plusmn5%. A data range selection approach,

Xiaoming Zhang; Xiong Yin; Zhaoyang Yu

2008-01-01

412

Associating audio-visual activity cues in a dominance estimation framework  

Microsoft Academic Search

We address the problem of both estimating the dominant person in a meeting from a single audio source and identifying them visually in a multi-camera setting. We use a speaker diarization algorithm to perform speaker segmentation and clustering, representing when they spoke. Using a greedy ordered audio-visual association algorithm, we investigate using the speaker clusters to find the corresponding person

Hayley Hung; Yan Huang; Chuohao Yeo; Daniel Gatica-Perez

2008-01-01

413

Some methodological aspects for measuring asynchrony detection in audio-visual stimuli  

Microsoft Academic Search

For audio-visual stimuli with a clear temporal structure, like impact events, a difference exists between physical and perceived synchrony. We compare the results of different methods to establish the point of subjective equality (PSE). These methods differ in the type of response categories subjects can use: 1) 3 categories: audio first, synchronous, video first, 2) 2 categories: synchronous, asynchronous, 3)

Armin Kohlrausch; Lawrence KS

414

Effects of Audio-Visual Information on the Intelligibility of Alaryngeal Speech  

ERIC Educational Resources Information Center

Background: There is minimal research on the contribution of visual information on speech intelligibility for individuals with a laryngectomy (IWL). Aims: The purpose of this project was to determine the effects of mode of presentation (audio-only, audio-visual) on alaryngeal speech intelligibility. Method: Twenty-three naive listeners were…

Evitts, Paul M.; Portugal, Lindsay; Van Dine, Ami; Holler, Aline

2010-01-01

415

The modulated lapped transform, its time-varying forms, and its applications to audio coding standards  

Microsoft Academic Search

The modulated lapped transform (MLT) is used in both audio and video data compression schemes. This paper describes its properties and how it can be used to generate a time-varying filterbank. Examples of its implementation in two audio coding standards are presented

Seymour Shlien

1997-01-01

416

HMM based structuring of tennis videos using visual and audio cues  

Microsoft Academic Search

This paper focuses on the use of hidden Markov models (HMMs) for structure analysis of videos, and demonstrates how they can be efficiently applied to merge audio and visual cues. Our approach is validated in the particular domain of tennis videos. The basic temporal unit is the video shot. Visual features describe the audio events within a video shot. The

E. Kijak; G. Gravier; P. Gros; L. Oisel; F. Bimbot

2003-01-01

417

Audio-Based versus Text-Based Asynchronous Online Discussion: Two Case Studies  

ERIC Educational Resources Information Center

The main objective of this paper is to examine the use of audio- versus text-based asynchronous online discussions. We report two case studies conducted within the context of semester-long teacher education courses at an Asian Pacific university. Forty-one graduate students participated in Study I. After the online discussions (both audio-based as…

Hew, Khe Foon; Cheung, Wing Sum

2013-01-01

418

A fuzzy approach towards perceptual classification and segmentation of MP3\\/AAC audio  

Microsoft Academic Search

The paper presents a novel perceptual based fuzzy approach towards classification and segmentation for MP3 and AAC audio in the compressed domain. The input audio is split into segments, which are classified as speech, music, fuzzy or silent. The proposed method minimizes critical errors of misclassification by fuzzy region modeling, thus increasing the efficiency of both pure and fuzzy classification.

Serkan Kiranyaz; Ahmad Farooq Qureshi; Moncef Gabbouj

2004-01-01

419

Unsupervised Segmentation and Classification over MP3 and AAC Audio Bitstreams  

Microsoft Academic Search

The paper presents a novel classification and segmentation scheme for MP3 and AAC audio in the compressed domain. The input audio is split into speech, music and silent segments using features such as total energy, band energy ratio, pause rate, subband centroid and fundamental frequency. Simulation results show the efficiency of the proposed algorithm.

Serkan Kiranyaz; Mathieu Aubazac; Moncef Gabbouj

2003-01-01

420

Audio-text synchronization inside mp3 files: a new approach and its implementation  

Microsoft Academic Search

The large usage of multimedia portable devices has contributed to a rapid increase in the demand for multimedia entertainment services. We focus on the karaoke service: several systems have been proposed but they are too difficult to be used directly over audio devices. Conversely, we propose a very simple approach to providing a karaoke-like service over any audio device that

Marco Furini; Lorenzo Alboresi

2004-01-01

421

Content-based Classification and Retrieval of Audio Using the Nearest Feature Line Method  

Microsoft Academic Search

A method is presented for content-based audio classification and retrieval. It is based on a new pattern classification method called the nearest Feature Line (NFL). In the NFL, information provided by multiple prototypes per class is explored. This contrasts to the nearest neighbor (NN) classification in which the query is compared to each prototype individually. Regarding audio representation, perceptual and

Stan. Z. Li

2000-01-01

422

Content-based audio classification and retrieval using the nearest feature line method  

Microsoft Academic Search

A method is presented for content-based audio classification and retrieval. It is based on a new pattern classification method called the nearest feature line (NFL). In the NFL, information provided by multiple prototypes per class is explored. This contrasts to the nearest neighbor (NN) classification in which the query is compared to each prototype individually. Regarding audio representation, perceptual and

Stan Z. Li

2000-01-01

423

EFFECTIVENESS OF AUDIO AND VISUAL TRAINING PRESENTATION MODES FOR GLUCOMETER CALIBRATION  

Microsoft Academic Search

This experiment investigated whether different presentation modes of instructional materials are differentially effective for older and younger adults learning to calibrate a glucometer. Glucometers are complex and require serial, sequential steps to calibrate them successfully. Some previous studies have failed to find a difference for older adults between instructions presented via audio and instructions presented with both audio and video

Anne C. McLaughlin; Wendy A. Rogers; Arthur D. Fisk

2002-01-01

424

Description of Audio-Visual Recording Equipment and Method of Installation for Pilot Training.  

ERIC Educational Resources Information Center

The Audio-Video Recorder System was developed to evaluate the effectiveness of in-flight audio/video recording as a pilot training technique for the U.S. Air Force Pilot Training Program. It will be used to gather background and performance data for an experimental program. A detailed description of the system is presented and construction and…

Neese, James A.

425

On the impact of loss and delay variation on Internet packet audio transmission  

Microsoft Academic Search

The quality of audio in IP telephony is significantly influenced by various factors, including type of encoder, delay, delay variation, rate and distribution of packet loss, and type of error concealment. Hence, the performance of IP telephony systems is highly dependent on understanding the contribution of these factors to audio quality, and their impact on adaptive transport mechanisms such as

Lopamudra Roychoudhuri; Ehab S. Al-shaer; Gregory B. Brewster

2006-01-01

426

CUAVE: A new audio-visual database for multimodal human-computer interface research  

Microsoft Academic Search

Multimodal signal processing has become an important topic of research for overcoming certain problems of audio-only speech processing. Audio-visual speech recognition is one area with great potential. Difficulties due to background noise and multiple speakers are significantly reduced by the additional information provided by extra visual features. Despite a few efforts to create databases in this area, none has emerged

E. K. Patterson; S. Gurbuz; Z. Tufekci; J. N. Gowdy

2002-01-01

427

"Listen to This!" Utilizing Audio Recordings to Improve Instructor Feedback on Writing in Mathematics  

ERIC Educational Resources Information Center

Providing audio files in lieu of written remarks on graded assignments is arguably a more effective means of feedback, allowing students to better process and understand the critique and improve their future work. With emerging technologies and software, this audio feedback alternative to the traditional paradigm of providing written comments…

Weld, Christopher

2014-01-01

428

Temporal Interval Discrimination Thresholds Depend on Perceived Synchrony for Audio-Visual Stimulus Pairs  

ERIC Educational Resources Information Center

Audio-visual stimulus pairs presented at various relative delays, are commonly judged as being "synchronous" over a range of delays from about -50 ms (audio leading) to +150 ms (video leading). The center of this range is an estimate of the point of subjective simultaneity (PSS). The judgment boundaries, where "synchronous" judgments yield to a…

van Eijk, Rob L. J.; Kohlrausch, Armin; Juola, James F.; van de Par, Steven

2009-01-01

429

Facilitating Discourse and Enhancing Teaching Presence: Using Mini Audio Presentations in Online Forums  

ERIC Educational Resources Information Center

The purpose of this pilot study was to determine if instructors' use of mini audio presentations (MAPs) in online discussions serves as an effective facilitation method, particularly when the content contains specific facilitation markers including reinforcement, recognition, and reward (three Rs). Instructors posted MAPs as audio file attachments…

Dringus, Laurie P.; Snyder, Martha M.; Terrell, Steven R.

2010-01-01

430

Age Matters: Student Experiences with Audio Learning Guides in University-Based Continuing Education  

ERIC Educational Resources Information Center

The primary objective of this research was to explore the experiences of undergraduate distance education students using sample audio versions (provided on compact disc) of the learning guides for their courses. The results of this study indicated that students responded positively to the opportunity to have word-for-word audio versions of their…

Mercer, Lorraine; Pianosi, Birgit

2012-01-01

431

Students' Attitudes to and Usage of Academic Feedback Provided via Audio Files  

ERIC Educational Resources Information Center

This study explores students' attitudes to the provision of formative feedback on academic work using audio files together with the ways in which students implement such feedback within their learning. Fifteen students received audio file feedback on written work and were subsequently interviewed regarding their utilisation of that feedback within…

Merry, Stephen; Orsmond, Paul

2008-01-01

432

Active Learning in the Online Environment: The Integration of Student-Generated Audio Files  

ERIC Educational Resources Information Center

Educators have integrated instructor-produced audio files in a variety of settings and environments for purposes such as content presentation, lecture reviews, student feedback, and so forth. Few instructors, however, require students to produce audio files and share them with peers. The purpose of this study was to obtain empirical data on…

Bolliger, Doris U.; Armier, David Des, Jr.

2013-01-01

433

Power-aware bandwidth and stereo-image scalable audio decoding  

Microsoft Academic Search

We propose a new workload-scalable audio decoding scheme that would enable users to control the tradeoff between playback quality and power consumption in battery-powered portable audio players. Our objective is to give users a control at the decoder side, similar to the Long Play (LP) recording mode at the encoder side in many media recording devices. The main contribution of

Wendong Huang; Ye Wang; Samarjit Chakraborty

2005-01-01

434

Application of simulated three-dimensional audio displays to fighter cockpits: a user survey  

Microsoft Academic Search

User opinion was solicited on the application of simulated three dimensional (3-D) audio displays to fighter cockpits. Seventy-six experienced military pilots were surveyed (72 had experience in tactical aircraft, 53 had Navy or Air Force fixed wing fighter experience, and average flight hours greater than 3000). Pilot opinion was collected about the utility and applicability of 3-D audio within broad

M. D. Lee; R. W. Patterson; D. J. Folds; D. A. Dotson

1993-01-01

435

One solution of audio\\/video grabber architecture used for automatic testing of Set Top Boxes  

Microsoft Academic Search

Modern TV technology, especially digital television today is equipped witn smart computer based TV signal receivers. This document shortly describes the hardware architecture and base functional principle of audio\\/video grabber device, used for automated testing of Set Top Boxes. Audio\\/video grabber is a part of BBT testing environment.

Miodrag Culum; Ivan Resetar; Dragan Cuca; Zoran Marceta

2011-01-01

436

WP9.06 Skew Detection and Compensation for Internet Audio Applications  

Microsoft Academic Search

Long lived audio streams, such as music broadcasts, and small differences in clock rates lead to buffer underflo w or overflo w events in receiving applications that manifest themselves as au- dible interruptions. We present a low complexity algorithm for detecting clock skew in network audio applications that function with local clocks and in the absence of a synchronization mech-

Orion Hodson; Colin Perkins; Vicky Hardman

2000-01-01

437

Calibration and 3-D sound reproduction in the Immersive Audio Environment  

Microsoft Academic Search

The effectiveness of virtual environments depends largely on how efficiently they recreate the real world. In the case of auditory virtual environments, the importance of accurate recreation is enhanced since there are no visual cues to assist perception, as in the case of audio-visual virtual environments. In this paper, we present the Immersive Audio Environment (IAE), an easily constructible and

Pratik Shah; Steven Grant; William Chapin

2011-01-01

438

AUDIO AND SPEECH SIGNAL PROCESSING FOR SECURITY AND SAFETY APPLICATION IN PUBLIC TRANSPORT  

Microsoft Academic Search

This paper addresses the issue of automatic audio and speech analysis in public transport. For many years, automatic surveillance systems aimed at detecting and identifying abnormal or critical situations using video analysis tools. Nevertheless, deployed in embedded areas, these video systems are placed in unfavourable situations characterised by mobility constraints. Many researchers leaded works on audio scene analysis, in particular

Sodoyer David; Ambellouis Sébastien

439

When I Stopped Writing on Their Papers: Accommodating the Needs of Student Writers with Audio Comments  

ERIC Educational Resources Information Center

The author finds using software to make audio comments on students' writing improves students' understanding of her responses and increases their willingness to take her suggestions for revision more seriously. In the process of recording audio comments, she came to a new understanding of her students' writing needs and her responsibilities as…

Bauer, Sara

2011-01-01

440

Virtual bass for home entertainment, multimedia PC, game station and portable audio systems  

Microsoft Academic Search

In digital audio playback systems for home entertainment, multimedia PC, game station and portable audio devices, there is a strong demand to produce deep bass using small multimedia speakers and earphones, without the need for additional subwoofer or expensive speakers\\/earphones. This paper describes a technique in synthesizing psycho-acoustic bass over a pair of stereo speakers using the pitch perception theory.

Woon S. Gan; S. M. Kuo; C. W. Toh

2001-01-01

441

Foundations for Creating Effective Two-Way Audio/Video Distance Education Environments.  

ERIC Educational Resources Information Center

This study examined the relationship between students' perceptions of the two-way audio/video classroom and their anxiety, as well as their satisfaction with their distance learning experience. Students (n=222) in 12 two-way audio/video distance classes at two major midwestern universities and two midwestern community colleges completed…

Reinhart, Julie; Schneider, Paul

442

Discrete Model to Estimate Lifetime of a Wireless Sensor Network for Audio Storage  

Microsoft Academic Search

Wireless sensor networks (WSNs) can be used to record and store audio data at remote and inaccessible places. However, audio data adds an additional concern to the design of the WSN; the motes need a larger amount of memory resources to be able to store the collected data. In this paper, we evaluate the ideas already presented, specifically the EnviroMic,

Sajid Hussain; Patrick Drane; Michael Mallinson

2008-01-01

443

Universal synchronization scheme for distributed audio-video capture on heterogeneous computing platforms  

Microsoft Academic Search

We propose a universal synchronization scheme for distributed audio-video capture on heterogeneous computing devices such as laptops, tablets, PDAs, cellular phones, audio recorders, and camcorders. These devices typically possess sensors such as microphones and possibly cameras. In order to combine them wirelessly into a distributed sensing and computing system, it is necessary to provide relative time synchronization among the distributed

Rainer Lienhart; Igor Kozintsev; Stefan Wehr

2003-01-01

444

Gunshot detection in audio streams from movies by means of dynamic programming and Bayesian networks  

Microsoft Academic Search

This paper treats gunshot detection in audio streams from movies as a maximization task, where the solution is obtained by means of dynamic programming. The proposed method seeks the sequence of segments and respective class labels, i.e., gunshots vs. all other audio types, that maximize the product of posterior class label probabilities, given the segments' data. The required posterior probabilities

Aggelos Pikrakis; Theodoros Giannakopoulos; Sergios Theodoridis

2008-01-01

445

Straightening-stretching machine  

SciTech Connect

A straightening-stretching machine has been designed at the All Union Design and Research Institute for Chemical Engineering. It straightens metal strips in the cold state by stretching beyond their limit with a maximum extension deformation of 3%. A sketch indicates stand, drive for untwisting strip, power cylinder, slide block, and front clamping head, among other aspects of the machine. The technical characteristics are specified and the process is explained. The economy affected on introducing the straightening-stretching machine is indicated.

Deryabin, G.N.; Kuz'min, G.G.

1983-07-01

446

Edheads: Simple Machines  

NSDL National Science Digital Library

This interactive Flash activity invites kids to learn about simple and compound machines by investigating common household objects found in the kitchen and tool shed. The animated activities help them understand how the machines work and how to differentiate the various types of simple machine. Additionally the site provides a glossary of important terms, lesson plans and a teacher's guide. This page is part of a larger collection of game-like animations developed to teach children about science.

2007-08-16

447

Machining: An Introduction  

NSDL National Science Digital Library

Basic machining processes are introduced on this site that is devoted to engineering fundamentals. Descriptions and illustrations of drilling, turning, grinding, and other common processes are provided for people with little to no prior machining knowledge. A waterjet is a non-traditional machining technology that uses high pressure streams of water with abrasive additives rather than solid cutting instruments to slice through metal and other materials.

2008-04-23

448

47 CFR 25.144 - Licensing provisions for the 2.3 GHz satellite digital audio radio service.  

Code of Federal Regulations, 2013 CFR

...Licensing provisions for the 2.3 GHz satellite digital audio radio service. 25...CONTINUED) COMMON CARRIER SERVICES SATELLITE COMMUNICATIONS Applications and Licenses...Licensing provisions for the 2.3 GHz satellite digital audio radio service....

2013-10-01

449

47 CFR 25.144 - Licensing provisions for the 2.3 GHz satellite digital audio radio service.  

Code of Federal Regulations, 2010 CFR

...Licensing provisions for the 2.3 GHz satellite digital audio radio service. 25...CONTINUED) COMMON CARRIER SERVICES SATELLITE COMMUNICATIONS Applications and Licenses...Licensing provisions for the 2.3 GHz satellite digital audio radio service....

2009-10-01

450

Audio-based queries for video retrieval over Java enabled mobile devices  

NASA Astrophysics Data System (ADS)

In this paper we propose a generic framework for efficient retrieval of audiovisual media based on its audio content. This framework is implemented in a client-server architecture where the client application is developed in Java to be platform independent whereas the server application is implemented for the PC platform. The client application adapts to the characteristics of the mobile device where it runs such as screen size and commands. The entire framework is designed to take advantage of the high-level segmentation and classification of audio content to improve speed and accuracy of audio-based media retrieval. Therefore, the primary objective of this framework is to provide an adaptive basis for performing efficient video retrieval operations based on the audio content and types (i.e. speech, music, fuzzy and silence). Experimental results approve that such an audio based video retrieval scheme can be used from mobile devices to search and retrieve video clips efficiently over wireless networks.

Ahmad, Iftikhar; Cheikh, Faouzi Alaya; Kiranyaz, Serkan; Gabbouj, Moncef

2006-02-01

451

Technologies de l'Information et Creation d'Emplois: L'Industrie Audio-Visuelle. Series FAST No. 12 (Information Technologies and the Creation of Employment in the Audio-Visual Industry. FAST Series No. 12).  

National Technical Information Service (NTIS)

The current market for audio-visual programs is examined and, taking into account the evolution of audio-visual technology, the potential for creating employment in this industry is assessed. It is estimated that the production of audio-visual leisure and...

W. Riblier J. P. Barbier

1983-01-01

452

A high quality re-quantization\\/quantization method for MP3 and MPEG4 AAC audio coding  

Microsoft Academic Search

MP3 is the popular audio coding standard. But now, a new higher quality audio coding standard Advanced Audio Coding (AAC) is proposed and widely used. The quantization\\/re-quantization is essential in both MP3 and AAC. This paper proposes a new high accuracy estimation algorithm for MP3 and MEPG-4 AAC audio coding. The algorithm can be applied not only for re-quantization process

Tsung-Han Tsai; Chuh-Chu Yen

2002-01-01

453

Effect of Audio Coaching on Correlation of Abdominal Displacement With Lung Tumor Motion  

SciTech Connect

Purpose: To assess the effect of audio coaching on the time-dependent behavior of the correlation between abdominal motion and lung tumor motion and the corresponding lung tumor position mismatches. Methods and Materials: Six patients who had a lung tumor with a motion range >8 mm were enrolled in the present study. Breathing-synchronized fluoroscopy was performed initially without audio coaching, followed by fluoroscopy with recorded audio coaching for multiple days. Two different measurements, anteroposterior abdominal displacement using the real-time positioning management system and superoinferior (SI) lung tumor motion by X-ray fluoroscopy, were performed simultaneously. Their sequential images were recorded using one display system. The lung tumor position was automatically detected with a template matching technique. The relationship between the abdominal and lung tumor motion was analyzed with and without audio coaching. Results: The mean SI tumor displacement was 10.4 mm without audio coaching and increased to 23.0 mm with audio coaching (p < .01). The correlation coefficients ranged from 0.89 to 0.97 with free breathing. Applying audio coaching, the correlation coefficients improved significantly (range, 0.93-0.99; p < .01), and the SI lung tumor position mismatches became larger in 75% of all sessions. Conclusion: Audio coaching served to increase the degree of correlation and make it more reproducible. In addition, the phase shifts between tumor motion and abdominal displacement were improved; however, all patients breathed more deeply, and the SI lung tumor position mismatches became slightly larger with audio coaching than without audio coaching.

Nakamura, Mitsuhiro [Department of Radiation Oncology and Image-applied Therapy, Kyoto University Graduate School of Medicine, Kyoto (Japan)], E-mail: m_nkmr@kuhp.kyoto-u.ac.jp; Narita, Yuichiro; Matsuo, Yukinori; Narabayashi, Masaru [Department of Radiation Oncology and Image-applied Therapy, Kyoto University Graduate School of Medicine, Kyoto (Japan); Nakata, Manabu [Clinical Radiology Service Division, Kyoto University Hospital, Kyoto (Japan); Sawada, Akira; Mizowaki, Takashi [Department of Radiation Oncology and Image-applied Therapy, Kyoto University Graduate School of Medicine, Kyoto (Japan); Nagata, Yasushi [Division of Radiation Oncology, Hiroshima University Hospital, Hiroshima (Japan); Hiraoka, Masahiro [Department of Radiation Oncology and Image-applied Therapy, Kyoto University Graduate School of Medicine, Kyoto (Japan)

2009-10-01

454

Combined video and audio watermarking: embedding content information in multimedia data  

NASA Astrophysics Data System (ADS)

Audio and video watermarking enable the copyright protection with owner or customer authentication and the detection of media manipulations. The available watermarking technology concentrates on single media like audio or video. But the typical multimedia stream consists of both video and audio data. Our goal is to provide a solution with robust and fragile aspects to guarantee authentication and integrity by using watermarks in combination with content information. We show two solutions for the protection of audio and video data with a combined robust and fragile watermarking approach. The first solution is to insert a time code into the data: We embed a signal as a watermark to detect gaps or changes in the flow of time. The basic idea uses numbers increasing by one. If in the verification process the next number is smaller than the last one or the step is greater than one, the time flow has been changed. This is realized without the combination of video and audio data. But we can synchronize the two data streams. A time signal is only valid if the combination of audio and video signals satisfy a certain attribute. To keep the basic example: if we embed an increasing a number in the audio and a decreasing number in the video, we could test if the combination of both always equals zero. The second solution is more complex: We use watermarks to embed information in each media about the content of the other media. With the help of speech recognition technology it is possible to embed the spoken text, the content, of an audio file in the video. With an algorithm previously developed in [1] we extract video content representation which is embedded in the audio stream. In our paper we present the problem of copyright protection and integrity checks for combined video and audio data. We show our two solutions and discuss our results.

Dittmann, Jana; Steinebach, Martin; Rimac, Ivica; Fischer, Stephan; Steinmetz, Ralf

2000-05-01

455

15 CFR 700.31 - Metalworking machines.  

Code of Federal Regulations, 2013 CFR

...machines Miscellaneous machine tools Miscellaneous secondary metal forming and cutting machines Planers and shapers Polishing, lapping, boring, and finishing machines Punching and shearing machines Riveting machines Saws and filing...

2013-01-01

456

76 FR 57923 - Establishment of Rules and Policies for the Satellite Digital Audio Radio Service in the 2310...  

Federal Register 2010, 2011, 2012, 2013

...Establishment of Rules and Policies for the Satellite Digital Audio Radio Service in the 2310-2360...collection requirements contained in the Satellite Digital Audio Radio Service (SDARS...effective date of these rule sections. See Satellite Digital Audio Radio Service...

2011-09-19

457

78 FR 44029 - Establishment of Rules and Policies for the Digital Audio Radio Satellite Service in the 2310...  

Federal Register 2010, 2011, 2012, 2013

...Policies for the Digital Audio Radio Satellite Service in the 2310-2360 MHz Frequency...the revised information collections for Satellite Digital Audio Radio Service (SDARS...Policies for the Digital Audio Radio Satellite Service in the 2310-2360 MHz...

2013-07-23

458

A Multi-Class Audio Classification Method With Respect To Violent Content In Movies Using Bayesian Networks  

Microsoft Academic Search

In this work, we present a multi-class classification algorithm for audio segments recorded from movies, focusing on the detection of violent content, for protecting sensitive social groups (e.g. children). Towards this end, we have used twelve audio features stemming from the nature of the signals under study. In order to classify the audio segments into six classes (three of them

Theodoros Giannakopoulos; Aggelos Pikrakis; Sergios Theodoridis

2007-01-01

459

Stone Picking Machines.  

National Technical Information Service (NTIS)

Stone picking machines are compared on the basis of their capabilities. Machines are classified as having a cyclic action which removes stones or a direct action which continuously loosens and sieves a layer of soil. The direct acting pickers have better ...

K. I. Preobrazhenskii V. S. Liflyandskii

1970-01-01

460

Fril++ for Machine Learning  

Microsoft Academic Search

Machine learning is one of the successful application areas of fuzzy set theory and fuzzy logic, which provide soft, and thus tolerant, way of partitioning attribute domains. Theoretical results have shown that there is no (fuzzy) machine learning algorithm that is the best for all tasks. Therefore, for a particular task, it is very useful to have a tool to

T. H. Cao; J. M. Rossiter

461

Semantics via Machine Translation  

ERIC Educational Resources Information Center

Recent experiments in machine translation have given the semantic elements of collocation in Russian more objective criteria. Soviet linguists in search of semantic relationships have attempted to devise a semantic synthesis for construction of a basic language for machine translation. One such effort is summarized. (CHK)

Culhane, P. T.

1977-01-01

462

Multimodal interactive machine translation  

Microsoft Academic Search

Interactive machine translation (IMT) [1] is an alternative approach to machine translation, integrating human expertise into the automatic translation process. In this framework, a human iteratively interacts with a system until the output desired by the human is completely generated. Traditionally, interaction has been performed using a keyboard and a mouse. However, the use of touchscreens has been popularised recently.

Vicent Alabau; Daniel Ortiz-Martínez; Alberto Sanchis; Francisco Casacuberta

2010-01-01

463

Stirling machine operating experience  

SciTech Connect

Numerous Stirling machines have been built and operated, but the operating experience of these machines is not well known. It is important to examine this operating experience in detail, because it largely substantiates the claim that stirling machines are capable of reliable and lengthy operating lives. The amount of data that exists is impressive, considering that many of the machines that have been built are developmental machines intended to show proof of concept, and are not expected to operate for lengthy periods of time. Some Stirling machines (typically free-piston machines) achieve long life through non-contact bearings, while other Stirling machines (typically kinematic) have achieved long operating lives through regular seal and bearing replacements. In addition to engine and system testing, life testing of critical components is also considered. The record in this paper is not complete, due to the reluctance of some organizations to release operational data and because several organizations were not contacted. The authors intend to repeat this assessment in three years, hoping for even greater participation.

Ross, B. [Stirling Technology Co., Richland, WA (United States); Dudenhoefer, J.E. [Lewis Research Center, Cleveland, OH (United States)

1994-09-01

464

The Hooey Machine.  

ERIC Educational Resources Information Center

Describes how students can make and use Hooey Machines to learn how mechanical energy can be transferred from one object to another within a system. The Hooey Machine is made using a pencil, eight thumbtacks, one pushpin, tape, scissors, graph paper, and a plastic lid. (PR)

Scarnati, James T.; Tice, Craig J.

1992-01-01

465

Simple Machine Junk Cars  

ERIC Educational Resources Information Center

During the month of May, the author's eighth-grade physical science students study the six simple machines through hands-on activities, reading assignments, videos, and notes. At the end of the month, they can easily identify the six types of simple machine: inclined plane, wheel and axle, pulley, screw, wedge, and lever. To conclude this unit,…

Herald, Christine

2010-01-01

466

Relational Algebra Machine GRACE  

Microsoft Academic Search

Most of the data base machines proposed so far which adopts a filter processor as their basic unit show poor performance for the heavy load operation such as join and projection etc., while they can process the light load operations such as selection and update for which a full scan of a file suffices. These machines which incorporates n processors

Masaru Kitsuregawa; Hidehiko Tanaka; Tohru Moto-oka

1982-01-01

467

Automatic soldering machine  

NASA Technical Reports Server (NTRS)

Fully-automatic tube-joint soldering machine can be used to make leakproof joints in aluminum tubes of 3/16 to 2 in. in diameter. Machine consists of temperature-control unit, heater transformer and heater head, vibrator, and associated circuitry controls, and indicators.

Stein, J. A.

1974-01-01

468

BRITISH MOLDING MACHINE, PBQ AUTOMATIC COPE AND DRAG MOLDING MACHINE ...  

Library of Congress Historic Buildings Survey, Historic Engineering Record, Historic Landscapes Survey

BRITISH MOLDING MACHINE, PBQ AUTOMATIC COPE AND DRAG MOLDING MACHINE MAKES BOTH MOLD HALVES INDIVIDUALLY WHICH ARE LATER ROTATED, ASSEMBLED, AND LOWERED TO POURING CONVEYORS BY ASSISTING MACHINES. - Southern Ductile Casting Company, Casting, 2217 Carolina Avenue, Bessemer, Jefferson County, AL

469

Cooperating reduction machines  

SciTech Connect

This paper presents a concept and a system architecture for the concurrent execution of program expressions of a concrete reduction language based on lamda-expressions. If formulated appropriately, these expressions are well-suited for concurrent execution, following a demand-driven model of computation. In particular, recursive program expressions with nonlinear expansion may, at run time, recursively be partitioned into a hierarchy of independent subexpressions which can be reduced by a corresponding hierarchy of virtual reduction machines. This hierarchy unfolds and collapses dynamically, with virtual machines recursively assuming the role of masters that create and eventually terminate, or synchronize with, slaves. The paper also proposes a nonhierarchically organized system of reduction machines, each featuring a stack architecture, that effectively supports the allocation of virtual machines to the real machines of the system in compliance with their hierarchical order of creation and termination. 25 references.

Kluge, W.E.

1983-11-01

470

Detection of vibrations in the audio range using photorefractive polymers  

NASA Astrophysics Data System (ADS)

We report on the use of a photorefractive polymer composite as the active material for a planar photo- EMF detector suitable for the adaptive detection of optical phase modulated signals in the audio range (10Hz-10KHz). The composite is based on a conjugated triphenyldiamine- phenylenevinylene polymer (TPD-PPV) and is sensitized with a highly soluble fullerene derivative (PCBM). We demonstrate experimentally that the responsitivity of such polymer based detectors can be remarkably enhanced if the polymer sample is biased by an external dc field. This effect is theoretically explained by the strong dependence of the charge carrier generation rate on the external dc field, which is an inherent property of organic photoconductors.

Mansurova, S.; Espinosa, M.; Rodriguez, P.; Gather, M.; Meerholz, K.

2006-09-01

471

Audio-visual speech perception: a developmental ERP investigation  

PubMed Central

Being able to see a talking face confers a considerable advantage for speech perception in adulthood. However, behavioural data currently suggest that children fail to make full use of these available visual speech cues until age 8 or 9. This is particularly surprising given the potential utility of multiple informational cues during language learning. We therefore explored this at the neural level. The event-related potential (ERP) technique has been used to assess the mechanisms of audio-visual speech perception in adults, with visual cues reliably modulating auditory ERP responses to speech. Previous work has shown congruence-dependent shortening of auditory N1/P2 latency and congruence-independent attenuation of amplitude in the presence of auditory and visual speech signals, compared to auditory alone. The aim of this study was to chart the development of these well-established modulatory effects over mid-to-late childhood. Experiment 1 employed an adult sample to validate a child-friendly stimulus set and paradigm by replicating previously observed effects of N1/P2 amplitude and latency modulation by visual speech cues; it also revealed greater attenuation of component amplitude given incongruent audio-visual stimuli, pointing to a new interpretation of the amplitude modulation effect. Experiment 2 used the same paradigm to map cross-sectional developmental change in these ERP responses between 6 and 11 years of age. The effect of amplitude modulation by visual cues emerged over development, while the effect of latency modulation was stable over the child sample. These data suggest that auditory ERP modulation by visual speech represents separable underlying cognitive processes, some of which show earlier maturation than others over the course of development.

Knowland, Victoria CP; Mercure, Evelyne; Karmiloff-Smith, Annette; Dick, Fred; Thomas, Michael SC

2014-01-01

472

RDM: A Relational Database Machine.  

National Technical Information Service (NTIS)

There is felt a need in the data processing industry for computing machines which are designed specifically to handle information retrieval and storage. One concept of such a database machine, called a Relational Database Machine (RDM), is presented. The ...

H. S. Ames

1977-01-01

473

Integrated Machine Translation System PIVOT,  

National Technical Information Service (NTIS)

The Integrated Machine Translation System PIVOT is a machine translation system using a knowledge base that accumulates knowledge of what is to be translated. The use of the epochal PIVOT system for machine translation system (inter mediate expression by ...

K. Muraki S. Ichiyama Y. Okazaki Y. Nagao Y. Aoki

1988-01-01

474

Single-Machine Bicriteria Scheduling.  

National Technical Information Service (NTIS)

Contents: Machine scheduling and multicriteria optimization: an introduction; Complexity of single-machine bicriteria scheduling: a survey; Minimizing maximum promptness and maximum lateness on a single machine; Polynomial-time algorithms for single-machi...

H. Hoogeveen

1992-01-01

475

Error-resilient design of high-fidelity scalable audio coding  

NASA Astrophysics Data System (ADS)

Current high quality audio coding techniques mainly focus on coding efficiency, which makes them extremely sensitive to channel noise, especially in high error rate wireless channels. In our previous work, we developed a progressive high quality audio codec, which was shown to outperform MPEG-4 version 2's scalable audio codec. In this work, we extend the error-free progressive audio codec to an error-resilient scalable audio codec by re-organizing the bitstream and modifying the noiseless coding module. A dynamic segmentation scheme is used to divide an audio bitstream into several segments. Each segment contains independently decodable data so that errors will not propagate across segment boundaries. An unequal error protection scheme is then adopted to improve error resilience of the final bitstream. The performance of the proposed algorithm is tested under different error patterns of WCDMA channels with several test audio materials. Our experimental results show that the proposed approach achieves excellent error resilience at a regular user bit rate of 64 kb/s.

Yang, Dai; Ai, Hongmei; Kyriakakis, Christos; Kuo, C.-C. Jay

2002-06-01

476

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

NASA Technical Reports Server (NTRS)

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; Mckinley, Richard; Kibbe, Marion; Francis, Daniel

1994-01-01

477

TECHNICAL NOTE: Portable audio electronics for impedance-based measurements in microfluidics  

NASA Astrophysics Data System (ADS)

We demonstrate the use of audio electronics-based signals to perform on-chip electrochemical measurements. Cell phones and portable music players are examples of consumer electronics that are easily operated and are ubiquitous worldwide. Audio output (play) and input (record) signals are voltage based and contain frequency and amplitude information. A cell phone, laptop soundcard and two compact audio players are compared with respect to frequency response; the laptop soundcard provides the most uniform frequency response, while the cell phone performance is found to be insufficient. The audio signals in the common portable music players and laptop soundcard operate in the range of 20 Hz to 20 kHz and are found to be applicable, as voltage input and output signals, to impedance-based electrochemical measurements in microfluidic systems. Validated impedance-based measurements of concentration (0.1-50 mM), flow rate (2-120 µL min-1) and particle detection (32 µm diameter) are demonstrated. The prevailing, lossless, wave audio file format is found to be suitable for data transmission to and from external sources, such as a centralized lab, and the cost of all hardware (in addition to audio devices) is ~10 USD. The utility demonstrated here, in combination with the ubiquitous nature of portable audio electronics, presents new opportunities for impedance-based measurements in portable microfluidic systems.

Wood, Paul; Sinton, David

2010-08-01

478

Interactive MPEG-4 low-bit-rate speech/audio transmission over the Internet  

NASA Astrophysics Data System (ADS)

The recently developed MPEG-4 technology enables the coding and transmission of natural and synthetic audio-visual data in the form of objects. In an effort to extend the object-based functionality of MPEG-4 to real-time Internet applications, architectural prototypes of multiplex layer and transport layer tailored for transmission of MPEG-4 data over IP are under debate among Internet Engineering Task Force (IETF), and MPEG-4 systems Ad Hoc group. In this paper, we present an architecture for interactive MPEG-4 speech/audio transmission system over the Internet. It utilities a framework of Real Time Streaming Protocol (RTSP) over Real-time Transport Protocol (RTP) to provide controlled, on-demand delivery of real time speech/audio data. Based on a client-server model, a couple of low bit-rate bit streams (real-time speech/audio, pre- encoded speech/audio) are multiplexed and transmitted via a single RTP channel to the receiver. The MPEG-4 Scene Description (SD) and Object Descriptor (OD) bit streams are securely sent through the RTSP control channel. Upon receiving, an initial MPEG-4 audio- visual scene is constructed after de-multiplexing, decoding of bit streams, and scene composition. A receiver is allowed to manipulate the initial audio-visual scene presentation locally, or interactively arrange scene changes by sending requests to the server. A server may also choose to update the client with new streams and list of contents for user selection.

Liu, Fang; Kim, JongWon; Kuo, C.-C. Jay

1999-11-01

479

The Basic Anaesthesia Machine  

PubMed Central

After WTG Morton's first public demonstration in 1846 of use of ether as an anaesthetic agent, for many years anaesthesiologists did not require a machine to deliver anaesthesia to the patients. After the introduction of oxygen and nitrous oxide in the form of compressed gases in cylinders, there was a necessity for mounting these cylinders on a metal frame. This stimulated many people to attempt to construct the anaesthesia machine. HEG Boyle in the year 1917 modified the Gwathmey's machine and this became popular as Boyle anaesthesia machine. Though a lot of changes have been made for the original Boyle machine still the basic structure remains the same. All the subsequent changes which have been brought are mainly to improve the safety of the patients. Knowing the details of the basic machine will make the trainee to understand the additional improvements. It is also important for every practicing anaesthesiologist to have a thorough knowledge of the basic anaesthesia machine for safe conduct of anaesthesia.

Gurudatt, CL

2013-01-01

480

Machine learning and radiology.  

PubMed

In this paper, we give a short introduction to machine learning and survey its applications in radiology. We focused on six categories of applications in radiology: medical image segmentation, registration, computer aided detection and diagnosis, brain function or activity analysis and neurological disease diagnosis from fMR images, content-based image retrieval systems for CT or MRI images, and text analysis of radiology reports using natural language processing (NLP) and natural language understanding (NLU). This survey shows that machine learning plays a key role in many radiology applications. Machine learning identifies complex patterns automatically and helps radiologists make intelligent decisions on radiology data such as conventional radiographs, CT, MRI, and PET images and radiology reports. In many applications, the performance of machine learning-based automatic detection and diagnosis systems has shown to be comparable to that of a well-trained and experienced radiologist. Technology development in machine learning and radiology will benefit from each other in the long run. Key contributions and common characteristics of machine learning techniques in radiology are discussed. We also discuss the problem of translating machine learning applications to the radiology clinical setting, including advantages and potential barriers. PMID:22465077

Wang, Shijun; Summers, Ronald M

2012-07-01

481

The basic anaesthesia machine.  

PubMed

After WTG Morton's first public demonstration in 1846 of use of ether as an anaesthetic agent, for many years anaesthesiologists did not require a machine to deliver anaesthesia to the patients. After the introduction of oxygen and nitrous oxide in the form of compressed gases in cylinders, there was a necessity for mounting these cylinders on a metal frame. This stimulated many people to attempt to construct the anaesthesia machine. HEG Boyle in the year 1917 modified the Gwathmey's machine and this became popular as Boyle anaesthesia machine. Though a lot of changes have been made for the original Boyle machine still the basic structure remains the same. All the subsequent changes which have been brought are mainly to improve the safety of the patients. Knowing the details of the basic machine will make the trainee to understand the additional improvements. It is also important for every practicing anaesthesiologist to have a thorough knowledge of the basic anaesthesia machine for safe conduct of anaesthesia. PMID:24249876

Gurudatt, Cl

2013-09-01

482

Machine Learning and Radiology  

PubMed Central

In this paper, we give a short introduction to machine learning and survey its applications in radiology. We focused on six categories of applications in radiology: medical image segmentation, registration, computer aided detection and diagnosis, brain function or activity analysis and neurological disease diagnosis from fMR images, content-based image retrieval systems for CT or MRI images, and text analysis of radiology reports using natural language processing (NLP) and natural language understanding (NLU). This survey shows that machine learning plays a key role in many radiology applications. Machine learning identifies complex patterns automatically and helps radiologists make intelligent decisions on radiology data such as conventional radiographs, CT, MRI, and PET images and radiology reports. In many applications, the performance of machine learning-based automatic detection and diagnosis systems has shown to be comparable to that of a well-trained and experienced radiologist. Technology development in machine learning and radiology will benefit from each other in the long run. Key contributions and common characteristics of machine learning techniques in radiology are discussed. We also discuss the problem of translating machine learning applications to the radiology clinical setting, including advantages and potential barriers.

Wang, Shijun; Summers, Ronald M.

2012-01-01

483

Verticality in hydroelectric machines  

SciTech Connect

Everyone who erects a vertical hydroelectric machine makes an effort to put the machine together so that its elements are vertical. A plumb line (or its optical equivalent) is a practical starting point for aligning the stationary parts of the machine - the generator stator, turbine case, and bearings. This does not mean, though, that the machine parts must be in near-perfect vertical orientation for the machine to perform well. Verticality is sometimes over-emphasized when procedures are undertaken to achieve good machine alignment. If the generator rotor, the connecting shaft, and the turbine runner are centered in stationary parts that are well-aligned along the same axis angle, this angle can depart from true vertical by a significant amount without ill effect. Mechanical balance does not often play a large part in determining how well a generator rotor is centered in the air gap. Magnetic forces are much more important. This is why it is desirable to maintain air gap variations around the machine to less than 5 percent from the average. However, this is sometimes difficult, especially if bearing clearances are large.

O'Kelly, F.

1991-12-01

484

Audio-visual onset differences are used to determine syllable identity for ambiguous audio-visual stimulus pairs  

PubMed Central

Content and temporal cues have been shown to interact during audio-visual (AV) speech identification. Typically, the most reliable unimodal cue is used more strongly to identify specific speech features; however, visual cues are only used if the AV stimuli are presented within a certain temporal window of integration (TWI). This suggests that temporal cues denote whether unimodal stimuli belong together, that is, whether they should be integrated. It is not known whether temporal cues also provide information about the identity of a syllable. Since spoken syllables have naturally varying AV onset asynchronies, we hypothesize that for suboptimal AV cues presented within the TWI, information about the natural AV onset differences can aid in speech identification. To test this, we presented low-intensity auditory syllables concurrently with visual speech signals, and varied the stimulus onset asynchronies (SOA) of the AV pair, while participants were instructed to identify the auditory syllables. We revealed that specific speech features (e.g., voicing) were identified by relying primarily on one modality (e.g., auditory). Additionally, we showed a wide window in which visual information influenced auditory perception, that seemed even wider for congruent stimulus pairs. Finally, we found a specific response pattern across the SOA range for syllables that were not reliably identified by the unimodal cues, which we explained as the result of the use of natural onset differences between AV speech signals. This indicates that temporal cues not only provide information about the temporal integration of AV stimuli, but additionally convey information about the identity of AV pairs. These results provide a detailed behavioral basis for further neuro-imaging and stimulation studies to unravel the neurofunctional mechanisms of the audio-visual-temporal interplay within speech perception.

ten Oever, Sanne; Sack, Alexander T.; Wheat, Katherine L.; Bien, Nina; van Atteveldt, Nienke

2013-01-01

485

An audio-visual corpus for speech perception and automatic speech recognition.  

PubMed

An audio-visual corpus has been collected to support the use of common material in speech perception and automatic speech recognition studies. The corpus consists of high-quality audio and video recordings of 1000 sentences spoken by each of 34 talkers. Sentences are simple, syntactically identical phrases such as "place green at B 4 now". Intelligibility tests using the audio signals suggest that the material is easily identifiable in quiet and low levels of stationary noise. The annotated corpus is available on the web for research use. PMID:17139705

Cooke, Martin; Barker, Jon; Cunningham, Stuart; Shao, Xu

2006-11-01

486

Engineering: Simple Machines  

NSDL National Science Digital Library

Simple machines are devices with few or no moving parts that make work easier. Students are introduced to the six types of simple machines â the wedge, wheel and axle, lever, inclined plane, screw, and pulley â in the context of the construction of a pyramid, gaining high-level insights into tools that have been used since ancient times and are still in use today. In two hands-on activities, students begin their own pyramid design by performing materials calculations, and evaluating and selecting a construction site. The six simple machines are examined in more depth in subsequent lessons in this unit.

Integrated Teaching And Learning Program

487

Machinability of Titanium Alloys  

NASA Astrophysics Data System (ADS)

Titanium and its alloys find wide application in many industries because of their excellent and unique combination of high strength-to-weight ratio and high resistance to corrosion. The machinability of titanium and its alloys is impaired by its high chemical reactivity, low modulus of elasticity and low thermal conductivity. A number of literatures on machining of titanium alloys with conventional tools and advanced cutting tool materials is reviewed. The results obtained from the study on high speed machining of Ti-6Al-4V alloys with cubic boron nitride (CBN), binderless cubic boron nitride (BCBN) and polycrystalline diamond (PCD) are also summarized.

Rahman, Mustafizur; Wong, Yoke San; Zareena, A. Rahmath

488

Virtual Turing Machine 2  

NSDL National Science Digital Library

A Turing machine is theoretical computer consisting of a finite set of internal states, a finite alphabet that includes a blank symbol, and a finite set of instructions. It has a physical head and a physical infinitely long tape, which is divided into cells. The cell values consist of the alphabet. The tape has a finite number of non-blank cells. The head can read and write to the cells and move the tape one cell to the left and one cell to the right. The Virtual Turing Machine lets you input tape values and an instruction set to see the output of a turing machine.

Ming, Paul R.

489

Mark Making Machines  

NSDL National Science Digital Library

In this activity, learners create a Mark Making Machine, a motorized writing tool that seems to walk about using an off-balanced DC motor, marker, and other materials. The rotational vibrations cause the machine to move and make marks. Mark Making Machines can take multiple forms and be made out of a variety of recycled materials. After an initial investment, materials can be re-used almost indefinitely. This activity guide contains suggested materials, building instructions, and other resources. A great activity for a wide age range.

Houston, Children'S M.

2009-01-01

490

Electrical Discharge Machining (EDM).  

National Technical Information Service (NTIS)

The objectives of this program were to increase the efficiency of the electrical discharge machining process, decrease manufacturing costs, increase reliability and structural integrity of production parts, and extend the utilization of EDM manufacturing ...

D. J. Moracz

1973-01-01

491

Positive Linear Function Machine  

NSDL National Science Digital Library

Students investigate linear functions with positive slopes by trying to guess the slope and intercept from inputs and outputs. Positive Linear Function Machine is one of the Interactivate assessment explorers.

492

Machine Building: Chapter IV.  

National Technical Information Service (NTIS)

This report on machine building is devoted to basic information on the topic of sheet forming. Sub-headings include: (1) relief forming; (2) flanging of holes and external contour; (3) high power forming; (4) stretching (hollow upsetting); (5) squeezing; ...

V. P. Romanovskii

1967-01-01

493

Concrete Casting Machine.  

National Technical Information Service (NTIS)

The machine is for casting, on a production line basis, articulated concrete revetment mats employed to protect riverbanks and flood control levees from hydraulic erosion. The apparatus includes a moving conveyor system which carries flat pallets to a cas...

J. I. Boswell T. Burks G. F. Dixon G. S. Lee

1964-01-01

494

Metalworking and Machining Fluids.  

National Technical Information Service (NTIS)

Improved boron-based metal working and machining fluids. Boric acid and boron-based additives that, when mixed with certain carrier fluids, such as water, cellulose and/or cellulose derivatives, polyhydric alcohol, polyalkylene glycol, polyvinyl alcohol, ...

A. Erdemir F. Sykora M. Dorbeck

2003-01-01

495

Meso-Machining Capabilities.  

National Technical Information Service (NTIS)

Meso-scale manufacturing processes are bridging the gap between silicon-based MEMS processes and conventional miniature machining. These processes can fabricate two and three-dimensional parts having micron size features in traditional materials such as s...

Benavides Adams Yang

2001-01-01

496

Monitoring of Machines.  

National Technical Information Service (NTIS)

The application of classical vibration analysis to the surveillance of the mechanical condition of machines is described. Automated procedures handled by specialized computers, using advanced signal processing techniques, are sought. Vibration measurement...

M. Gaillochet

1981-01-01

497

A Function Machine  

ERIC Educational Resources Information Center

In this article, the author describes a lesson he observed involving a function machine. This function machine was a box with a slot at the top of one side and a large cut-out hole at the bottom of the opposite side. A card with a number written on it (the input) was pushed into the slot and the teacher put their hand through the hole of the other…

Hewitt, Dave

2008-01-01

498

Weka Machine Learning Project  

NSDL National Science Digital Library

If you are inspired to try the process, the Weka Machine Learning Project from Waikato University offers open source software that can be used for data mining tasks. Visitors can also find the projects 1993 to 2006 publications, many of which are available for free in as PDFs. The "related" section offers a number of links to further information on topics such as artificial intelligence and machine learning.

2008-01-10

499

Metalworking and machining fluids  

SciTech Connect

Improved boron-based metal working and machining fluids. Boric acid and boron-based additives that, when mixed with certain carrier fluids, such as water, cellulose and/or cellulose derivatives, polyhydric alcohol, polyalkylene glycol, polyvinyl alcohol, starch, dextrin, in solid and/or solvated forms result in improved metalworking and machining of metallic work pieces. Fluids manufactured with boric acid or boron-based additives effectively reduce friction, prevent galling and severe wear problems on cutting and forming tools.

Erdemir, Ali (Naperville, IL); Sykora, Frank (Caledon, ON, CA); Dorbeck, Mark (Brighton, MI)

2010-10-12

500

Sealing intersecting vane machines  

DOEpatents

The invention provides a toroidal intersecting vane machine incorporating intersecting rotors to form primary and secondary chambers whose porting configurations minimize friction and maximize efficiency. Specifically, it is an object of the invention to provide a toroidal intersecting vane machine that greatly reduces the frictional losses through intersecting surfaces without the need for external gearing by modifying the width of one or both tracks at the point of intermeshing. The inventions described herein relate to these improvements.

Martin, Jedd N. (Providence, RI); Chomyszak, Stephen M. (Attleboro, MA)

2007-06-05